00:00.17 | Tuplink | is theyr a free one |
00:00.22 | Tuplink | maybe not alison? |
00:00.23 | tzanger | harryvv: sure |
00:00.28 | bkw_ | har ha rhar |
00:00.33 | Hogie | when you are flying and you just happen to have an ATC that's female, and being an asshole, when you are leaving her airspace, you say See You Next Time |
00:00.41 | harryvv | okay and in most cases that is what the channel bank is used for then. |
00:00.43 | tzanger | Tuplink: use your own voice or convince a woman with a sexy voice to do it for you |
00:00.46 | MikeJ[Jayden] | a free person to record prompts for you, sure, a soft phone, and you! |
00:01.25 | tzanger | persoanlly I find alison's recordings have FAR too much bass |
00:01.25 | MikeJ[Jayden] | but beware, you sound like shit |
00:01.25 | MikeJ[Jayden] | :) |
00:01.45 | Hogie | any idea why my tdm03b wants to dial out of a channel that is unplugged, when my x100 didn't do that? |
00:01.50 | facek_ | anyone use linksys PAP2 adapter? |
00:01.52 | Tuplink | MikeJ i agree |
00:02.04 | shido6 | facek_, whats up? |
00:02.24 | tzanger | Hogie: I thought we discussed this already |
00:02.27 | facek_ | shido6 I have problem with quality and some times with log |
00:02.35 | tzanger | the wctdm driver does not detect line voltage, the wcfxo driver does |
00:02.37 | Hogie | tzanger: if you replied earlier, i didn't see it |
00:02.40 | harryvv | BTW, Transport canada commerical affliate navair I think it was called had there ATC controler all equipment moved 1/2 away from Vancouver into our city. no one ever knew thay had a atc sitting behind cosco store until it was in the news paper. |
00:02.51 | facek_ | shido6 and maybe you know how to set time (for digit wait) .. because i dont want press # at end of number |
00:02.54 | harryvv | 1/2 hour away |
00:02.57 | Hogie | so I need to have wcfxo loaded, and not tdm? |
00:03.12 | tzanger | Hogie: no |
00:03.21 | tzanger | the driver for the fxo modules on the tdm400p does not do what you want |
00:03.26 | tzanger | the driver for the fxo card does |
00:03.32 | tzanger | so fix the wctdm driver |
00:03.51 | MichaelCat | Does anyone want to help me try to fix caller ID inbound to my X100P clone |
00:03.58 | harryvv | tzanger, do you recomend the rhino channel bank |
00:04.01 | bkw_ | clones don't work right with callerid |
00:04.02 | bkw_ | NEXT!!! |
00:04.03 | facek_ | shido6 ? |
00:04.06 | bkw_ | thats one thing you'll see |
00:04.08 | Hogie | I thought it had already been done... There was a bug made in Sep about it... |
00:04.15 | tzanger | harryvv: never used it |
00:04.18 | harryvv | k |
00:04.19 | tzanger | I used the adit600 |
00:04.29 | tzanger | and the access bank i/ii for FXS ONLY |
00:04.35 | harryvv | I see |
00:05.08 | MichaelCat | bkw_, so how can I be sure when I buy one on ebay to get a real one? |
00:05.09 | Hogie | http://bugs.digium.com/bug_view_page.php?bug_id=0002359 |
00:05.17 | Hogie | that's what's happening to me... |
00:06.00 | MikeJ[Jayden] | MichaelCat, digium does not make them anymore |
00:06.45 | MichaelCat | <MikeJ[Jayden]>, I know but neet to get something working |
00:07.00 | Hogie | and as I see on there, mark said he fixed it in cvs back then, so it should be in stable now, right? |
00:07.30 | MikeJ[Jayden] | MichaelCat, you may be able to sniff one out, but the tdm cards will do what you need |
00:07.35 | MichaelCat | I do have a Dialogic D/4PCI but have no idea how to get it to work with Asterisk if it is even possible |
00:08.24 | MichaelCat | All I need is one inbound FXO which supports Caller ID in the US |
00:09.45 | MikeJ[Jayden] | MichaelCat, http://www.voip-info.org/wiki-Asterisk+Hardware, there is a note on those cards, you will have to see if it is compatible or not |
00:11.11 | MichaelCat | <MikeJ[Jayden]>, actually the card I listed is different than the one on that web site, unless I am missing something |
00:12.11 | MikeJ[Jayden] | the question is, is it full duplex or not... and you will need to deal with digium for drivers |
00:13.11 | MichaelCat | It is cuposed to be full duplex but I do not know were to find the drivers |
00:15.44 | MikeJ[Jayden] | did you read that page |
00:17.20 | MichaelCat | So it is not possible to get caller ID working on the X100P clone (Ambient chip on it)? |
00:19.27 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
00:23.11 | MichaelCat | <MikeJ[Jayden]>, thanks, I will just call Digiom and order the TDM11B bundle |
00:23.15 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
00:25.15 | shmaltz | anybody here knows how much a Televantage or Avaya, or Toshiba system suporting 300 users in 3 different locaions, each having quad span T1 capabilities, would run (ball park figure) with the phones and all the equipment? |
00:28.34 | shido6 | in the asterisk world |
00:28.35 | shido6 | ? |
00:28.44 | MikeJ[Jayden] | 300 bucks a port? |
00:28.56 | MikeJ[Jayden] | including phones??? |
00:28.58 | MikeJ[Jayden] | that |
00:29.02 | *** join/#asterisk jf_ (~feulghulc@modemcable077.187-80-70.mc.videotron.ca) |
00:29.04 | MikeJ[Jayden] | 's a giess |
00:29.09 | MikeJ[Jayden] | guess... |
00:29.11 | shmaltz | shido6, no in avaya and toshiba |
00:29.30 | MikeJ[Jayden] | excuse me, I have to go teach myself to type...bbiab |
00:29.32 | shmaltz | MikeJ, including the phones? |
00:29.39 | *** join/#asterisk Mentat (~mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
00:29.41 | MikeJ[Jayden] | wag |
00:29.49 | MikeJ[Jayden] | 400. |
00:29.58 | shmaltz | I'm asking with the phones, I think its much more than 300 |
00:30.01 | MikeJ[Jayden] | 500 |
00:30.10 | MikeJ[Jayden] | used or new? |
00:30.20 | shmaltz | new |
00:30.44 | Qwell | shido6: got a second for an odd question? Maybe you can help me figure out why/where its happening |
00:30.55 | *** join/#asterisk jf_ (~feulghulc@modemcable077.187-80-70.mc.videotron.ca) |
00:30.57 | shido6 | k |
00:31.00 | shmaltz | more than 500 |
00:31.03 | shmaltz | I think |
00:31.18 | jf_ | anyone have an idea why each time i reboot, i have to recompile zaptel to use it |
00:31.29 | tzanger | jf_: build the modules correctly |
00:31.29 | jf_ | i use kernel 2.6 |
00:31.33 | jf_ | i did |
00:31.35 | tzanger | sounds like you have a fucked up distro |
00:31.59 | Qwell | from my nufone account, if I call a certain 800 number, with my CIDNum set as my 800 DID...I get fast busy. If I change the "area code", it works fine. I'm also able to call to the direct lines of people, npanxx1234, 800 CID or not |
00:32.12 | tzanger | Qwell: that is not nufone's problem |
00:32.23 | jf_ | tzanger: or maybe i need to emerge another package |
00:32.23 | tzanger | the 800 # you are calling does not have 800 as part of their acceptable NPAs |
00:32.29 | tzanger | ugh gentoo |
00:32.35 | jf_ | ya |
00:32.55 | Qwell | tzanger: Didn't say it was a nufone problem. I was certain it wasn't actually. |
00:33.04 | Qwell | tzanger: any idea if thats fixable on their end? |
00:33.11 | Qwell | they being the remote party |
00:33.16 | tzanger | jf_: find out specifically what seems to be missing (I'm gonna hazard a guess that the modules aren't present in an initrd) and fix it |
00:33.19 | tzanger | Qwell: yeah |
00:33.22 | MikeJ[Jayden] | shmaltz, ok, you guess.. I was using used numbers, I havn't gotten new in quite a while, toshiba phones run around 350-400 |
00:33.34 | jf_ | initrd |
00:33.36 | jf_ | ok |
00:33.40 | jf_ | let's see |
00:33.42 | tzanger | Qwell: don't set your CID to an 800# if it's calling an 80)# |
00:33.46 | tzanger | that is what I did to fix it |
00:33.48 | shmaltz | Mike per port with the system, or the phones itself? |
00:33.55 | Qwell | tzanger: I don't have any other DIDs. heh |
00:33.57 | jf_ | u mean u want me to do a rc-update |
00:34.06 | tzanger | Qwell: you don't have a non-800# to set it to? Then set it to empty |
00:34.17 | tzanger | jf_: I don't run gentoo, it was just a guess |
00:34.30 | Qwell | I'm just gonna fake it. 939-555-0113 |
00:34.34 | tzanger | Qwell: that'sll work |
00:34.35 | jf_ | gentoo is no good u mean |
00:34.36 | Qwell | bonus points if anyone knows what that number is |
00:34.46 | tzanger | jf_: I personally dislike that distro |
00:34.52 | tzanger | www.funroll-loops.org |
00:35.08 | jf_ | which one should i use then |
00:35.16 | tzanger | jf_: whatever you want to |
00:35.28 | tzanger | jf_: if you're comfortable with gentoo, then figure out what it doesn't like and fix it |
00:35.32 | Qwell | tzanger: Is that generally something at the provider or PBX level? |
00:35.46 | tzanger | Qwell: it's what the company with the 800# wanted |
00:35.52 | jf_ | k |
00:36.16 | Qwell | tzanger: I tried asking our telecom lady about it, but shes been ignoring me for the last 2 days. heh |
00:36.26 | tzanger | yeah |
00:36.28 | tzanger | she won't fix it |
00:36.29 | tzanger | she can't |
00:36.48 | Qwell | so, something the provider has to "fix"? |
00:36.53 | tzanger | the person with the 800# either did not realize or cared not to allow the toll-free NPAs to access their WAITS line |
00:36.57 | tzanger | Qwell: NO! |
00:37.02 | tzanger | dammit listen to me |
00:37.10 | Qwell | trying...not quite following |
00:37.16 | tzanger | when I buy an 800# I can decide who gets to and who doesn't get to call |
00:37.35 | tzanger | someone either specifically said "no toll-free NPAs" or they didn't realize they should include them |
00:37.45 | tzanger | you can't fix it and your telco can't fix it |
00:37.51 | Qwell | hmm |
00:38.02 | tzanger | you can work around it by setting your outgoing CID to a NON-800# number when calling 800#s |
00:38.38 | Qwell | workarounds due to lack of vision suck, heh |
00:38.42 | file[laptop] | it's the way it is. |
00:38.48 | tzanger | Qwell: welcome to the real world, baby |
00:38.49 | tzanger | :-) |
00:38.53 | Qwell | tzanger: indeed... |
00:40.04 | Qwell | So, to avoid something like that, one should specifically state that they DO want toll-free NPAs to be able to call, when ordering service? |
00:41.35 | tzanger | something like that, yea |
00:44.31 | Tuplink | how do i test my MOH? |
00:44.58 | tzanger | ... |
00:45.08 | tzanger | Tuplink: use your cell, call your house, answer and put yourself on hold |
00:45.10 | tzanger | jesus |
00:45.14 | Tuplink | well... Started music on hold, class 'default', on SIP/20001-c100 |
00:45.20 | Tuplink | but i heer nothing |
00:46.09 | Tuplink | default => mp3:/var/lib/asterisk/mohmp3 |
00:46.23 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
00:46.54 | Tuplink | shouldnt i heed somthing? |
00:47.34 | Tuplink | hear* |
00:47.57 | Qwell | Do you have the right version of mpg123? |
00:48.47 | Tuplink | its instaled but done see a version |
00:49.09 | Qwell | type `mpg123` |
00:49.12 | Tuplink | FreeBSD dosnt come with it so i must have ported it |
00:49.37 | Tuplink | .59r |
00:49.43 | Qwell | thats a first |
00:49.48 | Sedorox | ahah |
00:49.55 | Tuplink | what ver i need |
00:50.07 | Sedorox | that one... |
00:50.12 | Tuplink | hehe... |
00:50.17 | Tuplink | well it dont work |
00:50.19 | Tuplink | ;) |
00:50.26 | Sedorox | apparently so... |
00:50.31 | Sedorox | what problem are you having? |
00:50.37 | Tuplink | no MOH |
00:50.42 | Qwell | no sound |
00:50.43 | Tuplink | there we go.. |
00:50.49 | Qwell | huge difference |
00:50.49 | Sedorox | ? |
00:50.53 | Tuplink | it took for ever to come on... like 5 min |
00:51.12 | Sedorox | I know the defaults are kinda quiet to start... |
00:51.12 | Tuplink | *CLI> Warning, flexibel rate not heavily tested! |
00:51.12 | Tuplink | Apr 14 20:50:51 NOTICE[3956]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! |
00:51.13 | Qwell | Did you put a song with 5 minutes of silence in there? |
00:51.14 | Sedorox | you could always change the level from quiet to loud... |
00:51.27 | Tuplink | default |
00:51.48 | Tuplink | it takes forever for MOH to start i think |
00:52.00 | Tuplink | cause i took me off of hold and put me back and still |
00:52.03 | Sedorox | I know when I first did it.. it was just quiet |
00:53.14 | Tuplink | can i put any random MP3 in there? or need a specific birthrate? |
00:53.37 | Qwell | birthrate...thats also a first |
00:53.38 | Sedorox | I did... worked for me.. but I don't think its recommended... |
00:56.11 | *** join/#asterisk zilas (~1@adsl-065-015-074-044.sip.asm.bellsouth.net) |
00:56.47 | Tuplink | i think i got it figured out |
00:56.54 | Tuplink | ps -aux > root 4328 91.7 1.1 3628 2868 p0 R+ 8:50PM 0:57.48 mpg123 -q -s --mono -r 8000 -b 2048 -f 40 |
00:57.05 | Tuplink | 91% CPU usage |
00:57.55 | Sedorox | hehe.. ours was doing that.. dunno if it still is... |
00:58.07 | zilas | does anybody have any good example to setup asterisk callback feature after voicemail is left please? The one at wiki doesnt work good.. |
00:58.08 | Tuplink | how do i fix it? |
00:58.23 | Sedorox | <PROTECTED> |
00:58.25 | Sedorox | actually no.... |
00:58.48 | Tuplink | what CPU you have? |
00:59.10 | Sedorox | that is from a Celery 333 |
00:59.11 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
00:59.33 | Sedorox | we also run a dual Athlon MP machine... |
01:00.01 | Sedorox | and I forget what the last machine is |
01:03.25 | Tuplink | i killed the process and then the music came up |
01:03.27 | Tuplink | ;) |
01:03.43 | Sedorox | hmm |
01:03.57 | Sedorox | mpg123 is a PITA anyway.. |
01:04.51 | Tuplink | PITA? |
01:04.55 | Sedorox | pain in the ass |
01:05.11 | Tuplink | but as soon as i put me back on hold it fucks me |
01:06.13 | mgth | tuplink: Is it good at fucking? |
01:06.26 | Qwell | hmm, I know I can do ${EXTEN:1} to remove the last char, but is it possible to remove the first char(s)? |
01:07.40 | Sedorox | ${exten:1} is the first charactderf |
01:07.44 | Sedorox | (sp) |
01:07.46 | Tuplink | dosnt that remove the 1st |
01:07.54 | Qwell | erm, right...other way around |
01:08.03 | Sedorox | yes... |
01:08.04 | Sedorox | Ummm |
01:08.07 | Tuplink | maybe ${1:EXTEN} |
01:08.11 | Sedorox | ${EXTEN:-1} |
01:08.12 | Sedorox | I think... |
01:08.17 | Qwell | hmm |
01:08.20 | Sedorox | will remove the last digit... |
01:08.26 | Tuplink | kool |
01:08.29 | Sedorox | its in the wiki.. just came across it the other day |
01:08.37 | Tuplink | so... how do you remove the 1st and last? |
01:08.48 | Qwell | and, if I need to remove the first, and the last, I'd need something hackish like... ${{EXTEN:-1}:1}? |
01:08.51 | Sedorox | ummm |
01:08.56 | MajestiK | Does anyone have the Sipura Provisioning document that they could send my way? |
01:08.57 | Sedorox | ${EXTEN:1:1} |
01:09.00 | Tuplink | kool |
01:09.03 | Qwell | Sedorox: oh, sweet |
01:09.04 | Sedorox | hold on.. let me find the wiki |
01:09.41 | Tuplink | anyone have FWD? |
01:09.42 | JunK-Y | ${EXTEN:-1} is suppose to be valid. |
01:09.46 | Qwell | Tuplink: yeah |
01:09.53 | Sedorox | Tuplink: yes |
01:09.54 | Qwell | Tuplink: I'm sure a bunch of people do |
01:10.10 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
01:10.15 | shmaltz | Qwell this should work: |
01:10.17 | shmaltz | ${EXTEN:1:${LEN(${EXTEN})}-2} |
01:10.26 | tzanger | shmaltz: jesus |
01:10.29 | Sedorox | there is a easier way |
01:10.30 | JunK-Y | huH? |
01:10.33 | tzanger | step away from the computer |
01:10.36 | tzanger | and put DOWN the crack pipe |
01:10.38 | Sedorox | just have to find the damn page... |
01:10.39 | shmaltz | why :) |
01:10.45 | JunK-Y | tzanger: mouhahaa |
01:10.47 | harryvv | btw, is it possible to tie a asterisk box into the skype network? I did not relize how big and how fast thay are growing. |
01:10.48 | tzanger | we will be by shortly to confiscate your asterisk install |
01:10.57 | JunK-Y | ${EXTEN:-1} |
01:10.59 | tzanger | harryvv: short answer: no |
01:11.03 | shmaltz | why ??????????????????????????????? |
01:11.30 | JunK-Y | shmaltz: u passing by tokyo to go in europa. |
01:11.47 | shmaltz | why? JunK-Y |
01:12.01 | harryvv | mmm thats a bummer. So far thay have what 23 million users thats really large. |
01:12.01 | Sedorox | http://www.voip-info.org/wiki-Asterisk+variables |
01:12.05 | Sedorox | Look under Substrings |
01:12.09 | *** join/#asterisk ManxPower (~eric@adsl-35-236-60.msy.bellsouth.net) |
01:12.10 | Sedorox | and replace the number with EXTEN |
01:12.30 | Sedorox | <PROTECTED> |
01:12.31 | Micc | I do believe asterisk kicks ass. I just got asterisk setup with my broadvoice line. |
01:12.40 | harryvv | so what is skype a peer to peer network no centralized servers? |
01:13.07 | harryvv | micc well yes it does then there are other issues :) |
01:14.10 | *** join/#asterisk jbAU (~johnblade@61.8.110.41) |
01:15.45 | JunK-Y | harryvv: there's a lot of info related to skype |
01:16.20 | harryvv | yea downloading the rpm |
01:16.30 | *** join/#asterisk asteriskn00b (asteriskn0@adsl-68-91-7-226.dsl.tulsok.swbell.net) |
01:17.52 | JunK-Y | off topic: i wonder if i should get a Canon A75, whatcha think? |
01:19.02 | shmaltz | OK, well my original didn't work but this works: |
01:19.04 | shmaltz | exten => 12345678,1,Noop(${EXTEN:1:${LEN(${EXTEN:2})}}) |
01:19.05 | shmaltz | exten => 12345678,2,Hangup |
01:19.07 | shmaltz | it returns 234567 |
01:19.32 | JunK-Y | what ya want to return exactly? |
01:20.44 | shmaltz | or this: |
01:20.45 | *** join/#asterisk iq (~iq@70-59-161-91.omah.qwest.net) |
01:20.46 | shmaltz | exten => 12345678,1,Noop(${EXTEN:1:$[${LEN(${EXTEN})} - 2]}) |
01:20.47 | shmaltz | exten => 12345678,2,Hangup |
01:20.52 | shmaltz | that it should strip the first and last digit |
01:21.08 | Tuplink | dose exten => _2.,1,Dial(${EXTEN}) look rite to dial an extention from a VIR |
01:22.00 | shmaltz | Tuplink, nope |
01:22.24 | JunK-Y | shmaltz: u just want to take off the 1st and the last digits of ur exten right? |
01:22.26 | Tuplink | no...? |
01:22.28 | shmaltz | you could do exten => _2.,1,Dial(Local/${EXTEN}@contextname |
01:22.39 | shmaltz | yep JunK-Y |
01:22.48 | shmaltz | tzanger, I think I'm ok |
01:22.56 | shmaltz | what wrong with what I did? |
01:23.14 | Tuplink | i have all or my terminal ext under [localext] |
01:23.24 | shmaltz | Tuplink, goto has much better results |
01:23.41 | Tuplink | hum.... |
01:23.50 | JunK-Y | wait, phone |
01:24.01 | Tuplink | i jsut want hte user to be able to enter it at any time |
01:24.32 | shmaltz | if it's all in the same context, then you don't need anything |
01:24.37 | Tuplink | im new to this just started 3 days ago |
01:24.45 | shmaltz | or you could do an include if it's in a different context |
01:25.00 | shmaltz | Tuplink, np, we were all new at one point |
01:25.09 | Tuplink | i put an include => localext in the [mainmenu] |
01:25.16 | shmaltz | yep |
01:25.27 | Sedorox | shmaltz: maybe you.. but I had it loaded into my brain.... |
01:25.29 | Tuplink | and that did the trick |
01:25.30 | Sedorox | :-p |
01:25.41 | shmaltz | but if you did any AbsoluteTimeout in your IVR then you will have some trouble with dissconnected phone calls |
01:25.58 | Tuplink | nope |
01:26.09 | shmaltz | Sedorox, what did you have in ur brain? |
01:26.47 | Sedorox | its the matrix man... one button and asterisk is loaded into my brain |
01:27.02 | *** join/#asterisk hypa7ia (~leigh@modemcable176.166-203-24.mc.videotron.ca) |
01:27.18 | shmaltz | well, I don't need asterisk to tell me that if you take off the first and last digit of 12345678 its 234567 |
01:27.22 | shmaltz | heh |
01:27.25 | *** join/#asterisk ethzer0 (~ethzer0@d141-238-51.home.cgocable.net) |
01:27.32 | ethzer0 | hoi hoi |
01:28.19 | shmaltz | tzanger, you tried that? |
01:28.32 | shmaltz | you have another solution? |
01:30.27 | JunK-Y | ${EXTEN:-1:1} should be an interesting option, no? |
01:30.33 | JunK-Y | or 1:-1 |
01:30.44 | shmaltz | lets see |
01:31.00 | shmaltz | why? (I didn't test it yet) |
01:31.07 | JunK-Y | isnt working, but a patch should be make |
01:31.23 | shmaltz | the first one tells * strip the first digit, negated it tells it strip the last |
01:31.40 | JunK-Y | yea, whatcha think? |
01:31.43 | shmaltz | then the second number (after the :) |
01:32.01 | shmaltz | tells * for the length of 1 |
01:32.25 | shmaltz | this should return just the first digit |
01:32.35 | shmaltz | or maybe just the second to last |
01:32.44 | shmaltz | am I wrong? |
01:34.26 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
01:34.34 | Mazda-MX5 | Hi,all~ |
01:35.35 | JunK-Y | i said isnt make yet, that should be an option (a patch) to make, for like :1:-1 takes off the 1st and last digits |
01:36.01 | shmaltz | oh, that I agree with |
01:36.12 | JunK-Y | just for negative value |
01:36.25 | JunK-Y | for positive, that should take it as a length |
01:36.37 | JunK-Y | so :1:3 will stay the same as is it now. |
01:37.08 | JunK-Y | that makes sense? |
01:38.16 | shmaltz | nope |
01:38.21 | shmaltz | b/c right now |
01:38.55 | shmaltz | exten => 1234,1,Noop(${EXTEN:-1:2}) |
01:38.57 | shmaltz | returns 34 |
01:39.20 | shmaltz | so ${EXTEN:-1:1} should return |
01:39.22 | shmaltz | 4 |
01:39.26 | JunK-Y | im taking about the 2nd option |
01:39.40 | JunK-Y | 1:-1 |
01:39.46 | shmaltz | but this maybe |
01:40.02 | JunK-Y | its start:length |
01:40.02 | shmaltz | I was thinking -1:-1 |
01:40.12 | JunK-Y | no, im talking about 1:-1 |
01:40.24 | shmaltz | yep you are right |
01:40.33 | JunK-Y | 1:-2 would take off the 1st digit and the 2 last one. |
01:40.37 | shmaltz | 1:-1 should be the one |
01:40.38 | Qwell | there...8xx toll-free DID, CIDNum hack |
01:41.13 | JunK-Y | u know where all that kind of parse in done in which file exactly? |
01:42.22 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr) |
01:42.42 | shmaltz | don't have a clue |
01:42.55 | shmaltz | since I never (almost never) touch the source |
01:43.47 | file[mac] | okay, who stole res_gemini? IT WAS YOU JUNKY WASN'T IT?!? |
01:43.55 | JunK-Y | i think i found it |
01:44.08 | JunK-Y | file[laptop]: what the hell is res_gemini? |
01:44.15 | Mazda-MX5 | Hi, I have question , in my sip.conf can I write "bindaddr=192.168.0.0" to only allow 192.168.*.* phone connect ??? |
01:44.17 | file[mac] | my billing system |
01:44.37 | file[mac] | I can't find my specific build for it |
01:44.44 | JunK-Y | never heard about it. |
01:44.47 | timecop | er |
01:44.55 | file[mac] | AHA |
01:44.56 | file[mac] | found it |
01:44.56 | timecop | h323 requires some old crusty pwlib + oh323 |
01:45.04 | timecop | is it gonna break if I use more recent shit? |
01:45.11 | timecop | I cant even find 1.8.1 pwlib |
01:45.14 | file[mac] | timecop: probably |
01:45.17 | timecop | sf only has 1.8.0 and 1.8.3 |
01:45.25 | JunK-Y | shmaltz: file[laptop]: scroll up and tell me whatcha think about it. |
01:45.37 | timecop | http://sourceforge.net/project/showfiles.php?group_id=80674&package_id=89974 |
01:45.39 | timecop | :( |
01:45.44 | file[mac] | your idea? I think it's nifty enough |
01:45.58 | shmaltz | JunK-Y you mean the :1:-1 ? |
01:46.17 | file[mac] | I don't exactly take digits off from the end though, but meh - it may have it's application somewhere |
01:46.18 | JunK-Y | shmaltz: yep. |
01:46.37 | shmaltz | I thinks it should be done |
01:46.39 | shmaltz | meaning if the second number is negative then it just means how many numbers to cut off from the end |
01:46.39 | file[mac] | and yes I realize the ability to do what I just said is already there |
01:47.05 | JunK-Y | shmaltz: if u want to take a look on it, it's in pbx_retrieve_variable in pbx.c |
01:47.06 | timecop | ugh |
01:47.11 | timecop | wehre teh fuck do I get these old oh232 and pwlib |
01:47.16 | timecop | im just gonna compile 1.8.3 |
01:47.17 | timecop | and if it fails |
01:47.18 | *** join/#asterisk tessier (~treed@203.210.216.187) |
01:47.21 | timecop | blame on jerjer |
01:47.22 | shmaltz | which means that |
01:47.24 | shmaltz | :2:-4 on 123456789 will return 345 |
01:47.28 | Micc | what are the issues with multiple broadvoice lines and asterisk? |
01:47.29 | JunK-Y | file[laptop]: to go at cluecon, u pass by montreal? |
01:47.42 | file[mac] | JunK-Y: yes |
01:47.49 | JunK-Y | file[laptop]: trudeau? |
01:47.55 | file[mac] | lemme check |
01:48.06 | JunK-Y | probably, mirabel is almost dead. |
01:48.13 | *** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net) |
01:48.25 | JunK-Y | shmaltz: correct. |
01:49.18 | JunK-Y | when are ya passing by montreal? maybe we'll can go together from montreal to chicago. |
01:49.43 | file[mac] | it just says YUL |
01:50.08 | JunK-Y | YUL? |
01:50.14 | file[mac] | airport code |
01:50.23 | blitzrage | blah |
01:50.47 | JunK-Y | YUL is trudeau |
01:50.53 | file[mac] | there you go |
01:50.54 | JunK-Y | with aircanada right? |
01:51.00 | file[mac] | Tuesday August 2nd, 12:49PM get in |
01:51.02 | file[mac] | yes - Air Canada |
01:51.08 | file[mac] | depart 1:40PM |
01:51.22 | file[mac] | flight 519 |
01:51.24 | JunK-Y | so i should take my depart at 1:40pm too? |
01:51.27 | JunK-Y | let me check |
01:51.32 | file[mac] | well we'll see, I have yet to buy this ticket :p |
01:51.55 | file[mac] | as it's still a ways away and bkw hasn't given me all 'da info |
01:52.27 | JunK-Y | its chicago ORD i think. |
01:52.34 | shmaltz | JunK-Y, ok one will have to add code that if the second one is negative it should do a backward strip instead of offset |
01:52.42 | file[mac] | yes |
01:52.45 | file[mac] | that's O'Hare |
01:52.52 | shmaltz | I mean offset from end insead of length |
01:53.08 | JunK-Y | shmaltz: i'll try to patch it |
01:53.33 | JunK-Y | AC519 |
01:53.45 | file[mac] | yup. |
01:53.50 | JunK-Y | at 214$ right? |
01:53.51 | file[mac] | and then for the return flight, AC518 |
01:53.52 | Mazda-MX5 | Hi, I have question , in my sip.conf can I write "bindaddr=192.168.0.0" to only allow 192.168.*.* phone connect ??? |
01:54.03 | file[mac] | I use expedia myself |
01:54.07 | JunK-Y | file: which day? |
01:54.13 | file[mac] | Tuesday, the 2nd |
01:54.17 | shmaltz | Thanks but I'm ok with either: |
01:54.19 | shmaltz | ${EXTEN:1:${LEN(${EXTEN:2})}} |
01:54.21 | shmaltz | or: |
01:54.22 | JunK-Y | the return! |
01:54.23 | shmaltz | ${EXTEN:1:$[${LEN(${EXTEN})} - 2]} |
01:54.26 | JunK-Y | not the go. |
01:54.32 | file[mac] | ah 6th |
01:54.45 | shmaltz | ;) |
01:54.50 | file[mac] | maybe the aircanada site is cheaper, let's try my |
01:54.52 | file[mac] | flight... |
01:55.05 | JunK-Y | 6th at 13:25 right? |
01:55.32 | JunK-Y | go is at 214$, return is at 225$. |
01:55.48 | file[mac] | not too bad |
01:55.59 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.158) |
01:56.01 | JunK-Y | for a total of 563.01 $ |
01:56.12 | JunK-Y | tell me if u find something cheaper. |
01:56.13 | file[mac] | mine is $606.75 grand total |
01:56.22 | JunK-Y | but i'll call aircanada next week |
01:56.32 | JunK-Y | to know if they can offer me something cheaper. |
01:56.41 | file[mac] | but I don't know if I could stand being on the same flight as you *G* |
01:57.04 | JunK-Y | shut up, i'll kick ya out of the plane :) |
01:57.09 | file[mac] | haha |
01:57.12 | Qwell | Whats in Chicago? |
01:57.22 | file[mac] | developer conference |
01:57.23 | JunK-Y | Qwell: my grand-mother lives there. |
01:57.23 | JunK-Y | :) |
01:57.47 | Qwell | are non devs invited? heh |
01:57.48 | *** join/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
01:58.11 | blitzrage | well.. its a dev conf... not much point in going if you're not a dev :) |
01:58.18 | blitzrage | actually, I think they are having beginner tracks |
01:58.24 | blitzrage | so I guess there could be a point in going :) |
01:58.25 | file[mac] | yeah, we'll talk dev to eachother and you'll be scared! |
01:58.29 | PBXtech | what do SLIP's show up as in CLI? |
01:58.37 | Qwell | file[mac]: sweet |
01:59.03 | blitzrage | damn you! |
01:59.08 | *** join/#asterisk FryGuy- (~FryGuy@24.10.47.136) |
01:59.13 | file[mac] | blitzrage: muahahaha |
01:59.20 | blitzrage | hrmmmm... I just programmed something, and then realized I don't know what its doing :) |
01:59.33 | JunK-Y | file[laptop] : u reserved ur hotel? |
01:59.34 | file[mac] | oh right, I was going to work on this |
01:59.36 | file[mac] | silly me |
01:59.37 | fugitivo | blitzrage: skynet! |
02:00.07 | file[mac] | JunK-Y: I'm just paying the amount to anthm and bkw... includes some junk |
02:00.13 | file[mac] | and I get in for free since I'm a speaker, so yay |
02:00.24 | JunK-Y | where ya gonna lives? |
02:00.31 | JunK-Y | at anthm's house? |
02:00.32 | Qwell | file[laptop]: get me a speaker pass. I'll come up with something good. :p |
02:00.41 | file[mac] | hotel, he has some deal setup |
02:00.43 | blitzrage | JunK-Y: the 1:-1 doesn't work? |
02:00.45 | Qwell | like...I'll read chan_sip as though it were poety...or something. heh |
02:00.45 | timecop | so anyone running h323 channel with non-recommended pwlib/h323 bersions? |
02:00.51 | Qwell | poetry* |
02:00.54 | file[mac] | Qwell: sing a song, do a dance! |
02:00.56 | blitzrage | JunK-Y: I thought the whole x:x stuff got fixed... |
02:01.00 | file[mac] | make chan_sip totally RFC compliant |
02:01.01 | file[mac] | HAHAHAHAHA |
02:01.01 | JunK-Y | blitzrage: yues |
02:01.11 | file[mac] | sorry, I like to amuse myself sometimes |
02:01.19 | blitzrage | file[mac]: I don't think the SIP RFC is SIP RFC complient |
02:01.35 | file[mac] | blitzrage: I like that statement, it's very very nice |
02:01.40 | blitzrage | :D |
02:02.09 | file[mac] | I tell you baby all my dreams come true |
02:02.11 | blitzrage | shouldn't Astricon be posted in the topic? it's before ClueCOn |
02:02.13 | file[mac] | when I'm laying next to you |
02:02.23 | Qwell | another astricon already? |
02:02.29 | file[mac] | the Europe one. |
02:02.30 | blitzrage | Qwell: already? :) |
02:02.37 | file[mac] | which I will not be attending, so there's no reason to go *G* |
02:02.42 | Qwell | wasn't there one like...2 weeks ago? :p |
02:02.50 | blitzrage | file[mac]: drumkilla is going, so tons of reason to go :) |
02:02.51 | JunK-Y | blitzrage: no, :1-1 is like the :1 |
02:02.52 | file[mac] | oh oh oh I need to talk to oej |
02:02.57 | shmaltz | Qwell, you got the it done? |
02:03.03 | Qwell | shmaltz: the it? |
02:03.09 | shmaltz | the cutting the first and the last digit |
02:03.21 | Qwell | yeah, finished my toll-free DID CIDNum hack |
02:03.30 | shmaltz | how? |
02:03.34 | Qwell | lemme pastebin |
02:03.54 | malbech | I search a softswitch for a good price but it's very diificult to find one ... no ??? |
02:04.57 | JunK-Y | blitzrage: astricon europe yea |
02:05.05 | blitzrage | JunK-Y: you going? |
02:05.07 | timecop | i heard h323 compile will take half a day |
02:05.08 | Qwell | shmaltz: http://pastebin.ca/9579 |
02:05.08 | JunK-Y | ive no idea when the next ATL. |
02:05.14 | Qwell | I could have probably done it cleaner, but...meh |
02:05.18 | file[mac] | it's Thursday |
02:05.19 | JunK-Y | blitzrage: only if the boss of boss wants to pay |
02:05.19 | file[mac] | new stargate |
02:05.20 | file[mac] | I'M GONE |
02:05.21 | JunK-Y | which i doubt. |
02:05.23 | blitzrage | JunK-Y: ATL is around October probably |
02:05.33 | blitzrage | SG1 sucks |
02:05.33 | JunK-Y | yes, around that. |
02:05.44 | file[laptop] | blitzrage: shutup you |
02:05.47 | blitzrage | hehehe |
02:06.01 | JunK-Y | blitzrage: i'll need ppl to share hotel room |
02:06.12 | JunK-Y | want to share a room? |
02:06.21 | Qwell | JunK-Y: Stay with your grammy :p |
02:06.25 | JunK-Y | and dont take that for sexual advance! |
02:06.28 | *** join/#asterisk TheEmperor (~mattn@203.114.48.47) |
02:06.28 | JunK-Y | :P |
02:06.36 | JunK-Y | Qwell: hhehe |
02:06.38 | file[laptop] | blitzrage is straight! OMG |
02:06.42 | blitzrage | OMG! |
02:06.58 | file[laptop] | blitzrage: really though, you are *hot* |
02:07.00 | blitzrage | JunK-Y: hahaha, what are we talking about? Hotel for which conf? :) |
02:07.03 | shmaltz | Qwell, I thought you wanted to strip the last and first digit of a variable |
02:07.07 | file[laptop] | anyway |
02:07.11 | file[laptop] | stargate! |
02:07.17 | JunK-Y | if we can share a huge room, that would be cheaper |
02:07.19 | JunK-Y | cluecon |
02:07.21 | Qwell | shmaltz: well, no, not technically first and last. More like first, and last 4 |
02:07.21 | blitzrage | file[laptop]: well, thats what everyone says anyways... doesn't seem to help me pick up chicks though :) |
02:07.25 | JunK-Y | file[laptop]: star ya! |
02:07.27 | bjohnson | Qwell: why wouldn't you just do pattern matches instead of all those gotoifs? |
02:07.28 | PBXtech | what do SLIP's show up as in CLI? |
02:07.29 | file[laptop] | blitzrage: bah |
02:07.42 | JunK-Y | SLIP show up? |
02:07.45 | blitzrage | JunK-Y: agreed. Sure, if I can figure out a way to get there, I'm in. |
02:07.47 | PBXtech | dont they |
02:07.49 | Qwell | bjohnson: Because 808 is a valid areacode |
02:07.59 | bjohnson | I don't understand |
02:08.06 | Qwell | no, you are...I'm not |
02:08.06 | shmaltz | Qwelland, if I need to remove the first, and the last, I'd need something hackish like... ${{EXTEN:-1}:1}? |
02:08.07 | Qwell | explain? |
02:08.10 | PBXtech | dont SLIP error show up in the CLI? |
02:08.11 | JunK-Y | blitzrage: talk to that to others, i want a room. |
02:08.26 | Qwell | shmaltz: first and last was easier to understand then first and last 4 |
02:08.29 | blitzrage | JunK-Y: I don't even care how big of a room. We had a room at Astricon with 2 double beds, and 4 of us in a room. I slept on my own cot though. |
02:08.31 | JunK-Y | PBXtech: ive no idea of what u think. |
02:08.45 | PBXtech | t1 slips |
02:08.46 | bjohnson | Qwell: _1800NXXXXXX |
02:08.49 | JunK-Y | then, reserve me a place. |
02:08.53 | blitzrage | JunK-Y: yah for sure. We'll figure something out. I'm sure it'll be easy to get a bunch of people to share, no one wants to spend lots :) |
02:09.03 | file[laptop] | I'll share |
02:09.04 | Qwell | because I'm lazy, and I would still need a goto at the end of that match |
02:09.11 | blitzrage | file[laptop]: I thought you were gone :) |
02:09.13 | file[laptop] | any way I can cut costs I'm down for |
02:09.18 | file[laptop] | I have a laptop, I might as well use it :p |
02:09.22 | blitzrage | lol, fair enough |
02:09.24 | Qwell | or, I'd have to repeat the entire extension |
02:09.26 | bjohnson | Qwell: why? |
02:09.28 | blitzrage | JunK-Y: there you go... 3 people already |
02:09.35 | shmaltz | so here is how: |
02:09.36 | JunK-Y | blitzrage: why not going in a bar, find girls and go at their apt? :) |
02:09.37 | shmaltz | ${EXTEN:1:$[${LEN(${EXTEN})} - 5]} |
02:09.39 | shmaltz | this will do |
02:09.40 | Qwell | bjohnson: I need to get back into the "normal" flow anyhow |
02:09.48 | blitzrage | JunK-Y: its not that easy outside of Quebec :) |
02:09.50 | bjohnson | superdial baby |
02:09.50 | Qwell | shmaltz: So will ${EXTEN:1:3} |
02:09.56 | shmaltz | I know |
02:10.01 | blitzrage | ...well, sometimes it is... but not often :) |
02:10.03 | JunK-Y | blitzrage: damn, then move cluecon to montreal! |
02:10.06 | JunK-Y | hehehe |
02:10.08 | blitzrage | JunK-Y: no shit |
02:10.10 | shmaltz | but what if you don't know the length? |
02:10.20 | Qwell | bjohnson: back to the lazy part. :p |
02:10.26 | Qwell | shmaltz: then I have no business doing a hack like that, heh |
02:10.28 | shmaltz | like in sip channels |
02:10.44 | shmaltz | ${CHANNEL} holds the current channel |
02:11.17 | shmaltz | which always consists of SIP/sipaccountinsip.conf-16bithexnumber |
02:11.41 | bjohnson | Qwell: http://pastebin.ca/9580 |
02:11.46 | shmaltz | and I want to know how long the sipaccount part is |
02:11.53 | shmaltz | so this is what I do |
02:12.18 | *** join/#asterisk docelmo (~me@116-39.202-68.tampabay.res.rr.com) |
02:12.36 | docelmo | whadup? |
02:13.14 | Qwell | hmm |
02:13.29 | Qwell | if I do a Goto(abc), and have exten => abc,1,blah |
02:13.33 | Qwell | Does ${EXTEN} become abc? |
02:13.38 | bjohnson | and then I do lond distance in a separate context |
02:13.47 | tzanger | Qwell: why not try it? |
02:13.55 | tzanger | abc,1,noop(${EXTEN}) |
02:13.59 | docelmo | Qwell, yes it should.. |
02:14.01 | Qwell | tzanger: lazy mostly |
02:14.08 | shmaltz | {LEN(${CHANNEL:4:${LEN({CHANNEL:9})})} |
02:14.08 | bjohnson | and include them both in the main context so that I can control pattern match order (match toll free before it gets to the long distance pattern match) |
02:14.09 | shmaltz | or: |
02:14.11 | shmaltz | {LEN(${CHANNEL:4:$[${LEN({CHANNEL})} - 9])} |
02:14.12 | JunK-Y | Qwell: then we are too :) |
02:14.15 | docelmo | But I dont know about alpha extensions |
02:14.29 | JunK-Y | tzanger: whatcha think about the 1:-1 ? |
02:14.37 | tzanger | docelmo: This_is_a_valid_extension_name |
02:14.41 | bjohnson | I don't use the same least cost routing for toll free and long distance |
02:14.50 | docelmo | tz, then yes Q it will work |
02:15.30 | bjohnson | Qwell: yes |
02:15.37 | Qwell | damn |
02:15.45 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.245) |
02:15.49 | Qwell | I guess I could just setvar it before my goto |
02:16.01 | JunK-Y | yes u can. |
02:16.10 | Luhiwu | hi all |
02:16.20 | PBXtech | getting IRQ misses damn USB |
02:16.22 | Luhiwu | anyone is using cdr_addon_mysql? |
02:16.26 | Qwell | bjohnson: yeah yeah :p |
02:16.31 | shmaltz | gtg guys |
02:16.38 | shmaltz | gn |
02:16.46 | Qwell | bjohnson: I guess I'll go find a good one |
02:17.20 | Qwell | there are a bunch though, aren't there? |
02:17.34 | bjohnson | there's only one superdial baby |
02:17.49 | Qwell | oh |
02:18.26 | bjohnson | you could edit it .. but it does what people want to do about 99% of the time |
02:18.52 | docelmo | Luh yes |
02:19.27 | Luhiwu | docelmo: are you using accountcode set by sip.conf? it doesn't get inserted in my database |
02:19.53 | docelmo | yes its broken Im using the setaccount something in the dialplan |
02:20.24 | asteriskn00b | anyone have a opinion on the Aastra 480i ip phone? |
02:20.30 | Luhiwu | ok, thanks for the information, i didn't know it was broken |
02:20.55 | docelmo | I dont know for sure if its broke.. But thats the only way I can get it to work for me |
02:21.12 | *** join/#asterisk JerJer (~JerJer@DSL-224.210-rt-bras.che.centurytel.net) |
02:22.40 | blitzrage | hey, whats an easy way of searching for a term in a bunch of text files in Linux, but outputing the line number and the filename when it matches the term |
02:23.02 | Qwell | grep can do that, can't it? |
02:23.25 | Qwell | -H (which should be on by default) and -n |
02:24.13 | *** join/#asterisk brimstone (me@146.229.188.198) |
02:24.43 | JunK-Y | i love grep -R (recursive) |
02:25.07 | JunK-Y | it doesnt print the line number, but u can do a regex in that file. |
02:25.08 | blitzrage | Qwell: thanks, that works |
02:25.15 | Qwell | blitzrage: man grep :P |
02:25.18 | brimstone | is there an app that will pause xmms if it's playing then resume it after the call ends if it was playing? |
02:25.29 | blitzrage | Qwell: yep, thats what I'm doing now :) |
02:25.31 | JunK-Y | thx for info Qwell too. |
02:31.04 | blitzrage | ok, thats it for me, time to go relax in the living room with a pipe, then head to bed and do this all over again tomorrow. |
02:38.50 | Sedorox | hmm.. I thought you said pie... |
02:38.52 | shepherd | it's grep -r |
02:40.57 | libpcp | i would like to ask if anyone has already implemented an asterisk box running with 300 concurrent calls pstn or ip to ip ? |
02:42.06 | |Vulture| | 300... thats some # |
02:42.16 | |Vulture| | load balance to multiple boxes... |
02:42.32 | remmo | ip to ip should not be a problem |
02:42.36 | |Vulture| | pass internal to eachother and keep them apart from eachother |
02:42.48 | remmo | especially if asterisk is NOT in the media path |
02:46.03 | *** join/#asterisk urkle (~urkle@12-203-212-230.client.insightBB.com) |
02:47.18 | urkle | I'm trying to setup asterisk to connect to sipphone.com. I have outgoing calls working perfectly fine. but incoming calls get dumped to sipphone.com's voice mail, and packet sniffing on my end I see that asterisk is sending a 407 "proxy auth required" packet back to sipphone.com |
02:48.04 | ManxPower | ~docs |
02:48.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
02:48.05 | bugbot | docs is assigned nothing and reported nothing. |
02:48.06 | ManxPower | ~mailinglist |
02:48.07 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
02:48.07 | bugbot | mailinglist is assigned nothing and reported nothing. |
02:48.12 | *** join/#asterisk Rick_Hunter (~rhunter@01-098.008.popsite.net) |
02:48.14 | *** part/#asterisk brimstone (me@146.229.188.198) |
02:49.25 | urkle | ManxPower: I have looked through the docs for weeks.. and found at least 10 different docs, mailing list, forum posts on the subject.. and the examples provided that "claim" to work do not on my server. |
02:49.30 | *** join/#asterisk kks (~kks@203.115.208.140) |
02:51.57 | TheEmperor | hi guys, can anyone tell me what Asterisk died with code 1.Aborting. means? |
02:52.05 | JerJer | seg fault |
02:52.19 | denon | TheEmperor: tail /var/log/asterisk/messages |
02:52.29 | TheEmperor | how to rectify? |
02:52.31 | denon | TheEmperor: tail /var/log/asterisk/messages |
02:52.43 | TheEmperor | denon:ok, will check |
02:53.03 | *** join/#asterisk Moc_ (~Moc@modemcable165.109-70-69.mc.videotron.ca) |
03:00.28 | Yellow_Fuzzy | hi |
03:00.44 | Yellow_Fuzzy | anyone got experience with Oztell? |
03:01.07 | TheEmperor | denon: got these error messages |
03:01.20 | TheEmperor | denon: Apr 15 11:02:03 ERROR[2930]: Unable to register channel '1-4' |
03:01.20 | TheEmperor | Apr 15 11:02:03 WARNING[2930]: chan_zap.so: load_module failed, returning -1 |
03:01.20 | TheEmperor | Apr 15 11:02:03 WARNING[2930]: Loading module chan_zap.so failed! |
03:02.21 | *** join/#asterisk jeffik (jefik@69.158.30.24) |
03:03.42 | jeffik | I need to add ability to access voice mail from outside asteisk using * during mailbox greeting |
03:06.35 | timecop | watching openh323 compile is SO exciting |
03:06.38 | JunK-Y | ~seen paulc |
03:06.52 | jbot | paulc <~paulc@S010600062586a0b4.vc.shawcable.net> was last seen on IRC in channel #asterisk, 7d 8h 13m 12s ago, saying: 'Vancouver BC, it just hit C$1.00/liter'. |
03:06.52 | bugbot | seen paulc is assigned nothing and reported nothing. |
03:06.52 | timecop | ~seen timecop |
03:06.53 | jbot | timecop is currently on #asterisk (15d 20h 21s). Has said a total of 185 messages. Is idling for 1s |
03:06.53 | bugbot | seen timecop is assigned nothing and reported nothing. |
03:06.53 | timecop | bullshit :( |
03:06.57 | JunK-Y | what bugbot does here? |
03:08.00 | drumkilla | he's hanging out with us |
03:08.13 | *** join/#asterisk doughecka_ (~Doug@doughecka.user) |
03:08.16 | JunK-Y | ~junky |
03:08.17 | bugbot | junky is assigned M3026, M3085, M3732, M3674, M3725, M3679 and reported M2922, M2593, M2951, M3878, M2643, M2627, M2776, M4023, M3949, M2869, M2635, M2868, M2923, M3212, M3257, M3679, M3145, M4032, M3661, M2968 et al. |
03:08.22 | JunK-Y | nice. |
03:08.42 | drumkilla | ~drumkilla |
03:08.43 | jbot | well, drumkilla is the Asterisk v1.0-stable maintainer. ph33r him. |
03:08.43 | bugbot | drumkilla is assigned M2338, M3154, M3758, M3857, M3320, M3012, M2140, M2790, M2983, M3979, M3989, M1595, M3733, M2968, M3977, M2755, M3150, M2662, M3188, M2669 et al. and reported M2814, M4000, M3746, M3046, M3842, M3254, M3124, M3858, M3838, M3864, M3280, M3130, M3083, M3749, M3997, M3990, M3876, M3934, M3989. |
03:08.51 | drumkilla | ha |
03:08.53 | timecop | ~timecop |
03:08.54 | bugbot | timecop is assigned nothing and reported M1178, M1426, M25, M1475. |
03:08.59 | doughecka_ | ~doughecka |
03:09.00 | jbot | donations are accepted tad@heckaman.com |
03:09.00 | bugbot | doughecka is assigned nothing and reported nothing. |
03:09.03 | timecop | LOL SOMEONE FIX MY BUGS |
03:09.06 | doughecka_ | lol |
03:09.07 | timecop | PLZ |
03:09.09 | Corydon76-home | ~corydon76 |
03:09.11 | bugbot | corydon76 is assigned nothing and reported nothing. |
03:09.20 | drumkilla | denied |
03:09.27 | doughecka_ | does anyone know the default password on a cisco phone? |
03:09.33 | drumkilla | M4000 |
03:09.35 | bugbot | M4000 is a feature bug that is new (unassigned): [patch] WaitExten option for Music on Hold. It was filed by drumkilla and was last updated on 04-13-05. http://bugs.digium.com/bug_view_page.php?bug_id=4000 |
03:09.37 | doughecka_ | to unlock it so I can set the tftp server? |
03:09.52 | Corydon76-home | M2278 |
03:09.52 | Hogie | doughecka: why not set the tftp server in dhcp? |
03:09.52 | bugbot | M2278 is a feature bug that is closed (markster): [patch] Allow functions to be set. It was filed by Corydon76 and was last updated on 03-30-05. http://bugs.digium.com/bug_view_page.php?bug_id=2278 |
03:09.55 | JunK-Y | when new bugs gonna be reported, it will be past here too (as in #asterisk-bugs) |
03:09.56 | JunK-Y | ? |
03:10.10 | doughecka_ | Hogie: dhcp server doesnt allow it |
03:10.18 | drumkilla | not sure ... |
03:10.19 | doughecka_ | and I cant figure out how to configure linux dhcp to do it |
03:10.26 | Hogie | that's easy |
03:10.31 | timecop | M1475 |
03:10.32 | bugbot | M1475 is a minor bug that is closed (unassigned): SIP registration fails with Bad Request 400 until "reload" is executed. It was filed by timecop and was last updated on 05-05-04. http://bugs.digium.com/bug_view_page.php?bug_id=1475 |
03:10.36 | Hogie | option tftp-server-name "ipaddress"; |
03:10.37 | riksta | Hogie: you can, i'm sure |
03:10.40 | timecop | M1426 |
03:10.41 | bugbot | M1426 is a tweak bug that is closed (bkw918): [patch] Incoming SIP calls from SIP provider get wack channel names. It was filed by timecop and was last updated on 09-25-04. http://bugs.digium.com/bug_view_page.php?bug_id=1426 |
03:10.43 | Hogie | that's on rhel4 |
03:10.44 | timecop | ah this one |
03:10.50 | timecop | its stll broken |
03:10.55 | timecop | and nobody cares to fix it :( |
03:11.00 | timecop | i even had a patch, but it got ignored |
03:11.06 | TheEmperor | WARNING[2974]: chan_zap.c:771 zt_open: Unable to specify channel 2: No such device |
03:11.07 | Hogie | doughecka: try cisco |
03:11.09 | doughecka_ | in /etc/blah.conf? |
03:11.11 | Damin | M3660 |
03:11.11 | bugbot | M3660 is a feature bug that is closed (markster): [PATCH] don't do codec matching until we know who the caller is. It was filed by KP "beeps" Fleming and was last updated on 04-13-05. http://bugs.digium.com/bug_view_page.php?bug_id=3660 |
03:11.12 | TheEmperor | does this mean my fxo port is busted? |
03:11.12 | drumkilla | was it determined that it was a bug with your provider? |
03:11.26 | Damin | M3630 |
03:11.27 | bugbot | M3630 is a tweak bug that is closed (markster): [patch] allows 0 second retry intervals. It was filed by cmaj and was last updated on 02-20-05. http://bugs.digium.com/bug_view_page.php?bug_id=3630 |
03:11.28 | timecop | TheEmperor: more like you probably didnt configure it |
03:11.32 | timecop | drumkilla: what, me? |
03:11.46 | drumkilla | yes |
03:11.54 | timecop | drumkilla: its not a bug wiht a provider. |
03:11.57 | Hogie | doughecka: i'd paste my dhcpd.conf, but the rhel4 box that has it is not reachable because its offnet atm... my other dhcpds are windows, lol |
03:12.09 | timecop | incoming calls from FWD -> asterisk, the SIP/??? name is SIP/yourfwd#-random |
03:12.14 | TheEmperor | timecop: but all configured in zaptel.conf and zapata.conf... |
03:12.15 | drumkilla | timecop: ok, I see the bug now |
03:12.18 | timecop | which is highly pointless if one wants to know hte remote side |
03:12.22 | timecop | TheEmperor: did you run ztcfg? |
03:12.23 | timecop | or wahtever? |
03:12.34 | TheEmperor | timecop:yes |
03:12.38 | timecop | then you might be screwed. |
03:12.41 | drumkilla | timecop: you can reopen it ... *shrugs* |
03:12.48 | timecop | drumkilla: but it wont get fixed |
03:12.52 | TheEmperor | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
03:12.52 | TheEmperor | Channel 02: FXS Kewlstart (Default) (Slaves: 02) |
03:12.52 | TheEmperor | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
03:12.52 | TheEmperor | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
03:12.55 | timecop | because people probably think its dumb. |
03:13.10 | TheEmperor | timecop:would channel 2 be busted? |
03:13.18 | Hogie | Emperor did it ever work? |
03:13.32 | timecop | you shoul probably read |
03:13.33 | TheEmperor | Hogie: first installation now |
03:13.48 | urkle | ok.. outgoing calls work fine to sipphone.com and incoming calls can dial in and they can here me but I can not hear them.. and it looks (according to fiewall logs) that all the RTP traffic is going to port 5004 instead of the RTP range of 10000-10500.. |
03:14.23 | urkle | which is the RTP port of the internal sip adapter I'm using.. |
03:14.38 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
03:14.45 | TheEmperor | any ideas? |
03:15.50 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
03:15.52 | timecop | heh, stop wasting time wiht sip behind nat. |
03:15.56 | timecop | jsut get a global IP |
03:16.15 | Hogie | TheEmperor: tried swapping module 2 with another module? |
03:16.43 | TheEmperor | Hogie: the card I have is a 4 port fxo card... |
03:16.57 | Hogie | yeah.. try swapping module 2 and 3 around |
03:17.02 | TheEmperor | oh.. |
03:17.04 | Hogie | and see if it follows to 3 |
03:17.07 | TheEmperor | you mean take the card out and swap? |
03:17.11 | TheEmperor | take the module out from the card |
03:17.14 | Hogie | yes |
03:17.21 | TheEmperor | ? ok, i can try that |
03:17.22 | Hogie | might give you an idea if the module is bad |
03:17.47 | Hogie | that's what I would do anyway |
03:17.53 | TheEmperor | ok, i can give that a try |
03:18.17 | TheEmperor | Hogie:cause i mean modprobe, ztcfg and all is working well |
03:18.51 | Hogie | that happened with my 3 fxo |
03:18.57 | TheEmperor | hmm |
03:19.02 | Hogie | until I moved fxo from port 3 to 1 |
03:19.06 | Hogie | and then it was fine |
03:20.01 | malbech | I search a softswitch for a good price but it's very diificult to find one ... any idea ? |
03:20.19 | remmo | asterisk $0 + pc |
03:20.22 | JerJer | asterisk -- GPL |
03:22.49 | pigpen | hey..how would I set the outgoing caller id for lets say..fax numbers in * ? in the sip.conf? |
03:23.00 | Hogie | JerJer: is it possible to split my 1 account on nufone into 2, and have them pull from the same money poll, or should I just add a 3rd box on high bandwidth to direct 800's to the right offices? |
03:23.57 | *** join/#asterisk file[laptop] (~file@mctn1-6079.nb.aliant.net) |
03:24.27 | timecop | PWLib version is 1.8.3... BAD |
03:24.27 | timecop | Please read README for further details! |
03:24.27 | timecop | Makenshi: *** [checkversion] Error 1 |
03:24.29 | timecop | heh |
03:24.57 | timecop | lorf. |
03:24.57 | file[laptop] | you're mad! totally bad |
03:24.57 | file[laptop] | er mad |
03:24.57 | file[laptop] | and bad too |
03:25.00 | drumkilla | then don't you dare report bugs :p |
03:25.13 | timecop | i cant evne find required versions |
03:25.34 | timecop | they are NOT on sf.net's oh323 page. |
03:26.37 | timecop | ha it compiled. |
03:30.02 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
03:31.56 | Hogie | bleh blah |
03:32.16 | WGFreewill | Anybody every connect * via H.323 to Nortel carrier gear? |
03:32.29 | WGFreewill | (succession) |
03:32.40 | timecop | im about to connect it via H323 to some chinese voip provider |
03:32.43 | timecop | no idea what they use though. |
03:32.58 | WGFreewill | I have been passing 50k minutes a month |
03:33.00 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
03:33.05 | *** join/#asterisk jdg (~jdg@CA03F857.adsl.mana.pf) |
03:33.10 | WGFreewill | using JerJer chan_h323 |
03:33.20 | WGFreewill | to Cisco |
03:34.17 | WGFreewill | very reliable with this one small patch from PCadach |
03:34.23 | WGFreewill | from the bugtracker |
03:34.49 | PCadach | Which one you mean? WGFreewill? |
03:34.49 | WGFreewill | GW - Gw |
03:34.51 | timecop | good to know, too bad h323 is a pig |
03:35.03 | WGFreewill | deadlock H323 |
03:35.24 | WGFreewill | 10 tries and still no kill |
03:35.38 | PCadach | There still have an issues, especially when H.245 isn't embedded into signalling (H.225). |
03:35.43 | WGFreewill | #3643 and #3848 |
03:35.58 | WGFreewill | I got it to crash again |
03:36.02 | WGFreewill | during provisioning |
03:36.06 | WGFreewill | with Nortel CS2k |
03:36.14 | WGFreewill | you talk to a nortel box as GK |
03:36.22 | WGFreewill | and there are a farm of Gateway controllers |
03:36.24 | WGFreewill | that pass the RTP |
03:36.41 | WGFreewill | http://products.nortel.com/go/product_content.jsp?parId=0&segId=0&catId=-9274&prod_id=37501&locale=en-US |
03:37.04 | WGFreewill | this is some heavy duty voip gear the AS5850s were like flies |
03:37.19 | PCadach | I have crashes on callgen323 when H.245 goes through additional connection. It's not Asterisk-related but OpenH323... |
03:37.49 | WGFreewill | thats the noH245Tunneling = no |
03:37.53 | WGFreewill | right |
03:38.14 | *** join/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca) |
03:38.19 | *** join/#asterisk Moc[NX] (~mochouina@64.235.196.24) |
03:38.34 | WGFreewill | Nortel Problems so far |
03:38.37 | WGFreewill | no DTMF |
03:38.43 | WGFreewill | and g.711 ulaw fails |
03:38.49 | WGFreewill | g.729 works with no DTMF |
03:39.06 | Moc[NX] | Hail |
03:39.11 | WGFreewill | I am just not finding where it dies in the debugs |
03:39.21 | WGFreewill | I took packet sniff |
03:39.31 | WGFreewill | asterisk sends a release complete |
03:39.39 | WGFreewill | after the facilitycapability |
03:39.52 | DaLion | moc can u pm me ? |
03:40.33 | WGFreewill | Noafter terminalCapabilitySet |
03:40.41 | WGFreewill | * ReleaseComplete s |
03:40.45 | *** join/#asterisk Rick_Hunter (~rhunter@01-098.008.popsite.net) |
03:41.06 | *** join/#asterisk file[laptop] (~file@mctn1-6079.nb.aliant.net) |
03:41.08 | blitzrage | file[laptop]: ! |
03:41.20 | file[laptop] | what's happening?!? |
03:41.46 | Hogie | gah, flight school sucks sometimes |
03:41.52 | PCadach | Also, I'd not tested outgoing calls. When I makes next: callgen323 -> (H323) -> Asterisk-> (H323) -> callgen323 I have Asterisk's crash after about 5-10 successful calls. Probably there is the same issue with outgoing calls when H.245 isn't tunneled. |
03:43.30 | *** part/#asterisk urkle (~urkle@12-203-212-230.client.insightBB.com) |
03:44.15 | blitzrage | file[laptop]: not too much. Just heading to bed |
03:44.26 | file[laptop] | bah bed |
03:47.52 | pigpen | Is there any way to override the outgoing callerid info if a user has a did assigned to them? |
03:47.59 | *** part/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca) |
03:48.14 | file[laptop] | pigpen: yes, it's called dialplan logic - learn it |
03:48.34 | file[laptop] | or you can set the default callerid in their entry in the respective configuration file usually |
03:48.35 | pigpen | gee....thanks...you are so helpful. |
03:48.36 | file[laptop] | such as sip.conf |
03:48.50 | pgpkeys | pigpen: it'll take some time. I've been working on systems for over a decade and asterisk gives me the shits :) |
03:48.57 | pgpkeys | so expect some frustration :) |
03:49.14 | file[laptop] | but yes, learn about dialplan logic because you can do tons of stuff in it |
03:49.14 | pigpen | oh...I don't mind...but I have just been reading for 3 hours... |
03:49.34 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
03:49.34 | pigpen | yeah...I have been working "feaverishly" on it... |
03:49.46 | pgpkeys | pigpen: hehe I spent the night reading the manual. asterisk does a LOT. it's not a fire and forget like rogerwilco or teamspeak or ventrilo |
03:49.51 | remmo | anyone using those wuchuan ip phones? from aredfox.com |
03:50.05 | pigpen | but in the sip.conf I see only examples of: callerid=John Doe <1234> |
03:50.09 | pgpkeys | not me, I'm still using softsip |
03:50.29 | pigpen | so If I want to override the outgoing phone number just replace the 1234 part...not the actual name... |
03:50.45 | file[laptop] | exactly... |
03:50.45 | *** join/#asterisk iq (~IQ@70-59-161-91.omah.qwest.net) |
03:50.53 | pigpen | great...I was thinking that... |
03:51.02 | pigpen | but couldn't test at the moment. |
03:52.05 | *** join/#asterisk Rick_Hunter (~rhunter@04-156.008.popsite.net) |
03:53.14 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
03:54.21 | iq | ~int 13 |
03:54.22 | bugbot | int 13 is assigned nothing and reported nothing. |
03:54.28 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
03:54.34 | iq | ~iq |
03:54.35 | jbot | rumour has it, iq is that apts IQ is lower than 1 |
03:54.35 | bugbot | iq is assigned nothing and reported nothing. |
03:55.13 | iq | jbot got a girlfriend |
03:55.44 | |Vulture| | hmm is wiki down? |
03:56.27 | PTG1234 | file[mac]: you alive? |
03:56.31 | *** join/#asterisk Prowler1 (Prowler@d142-59-42-93.abhsia.telus.net) |
03:56.34 | file[laptop] | PTG1234: yes |
03:56.54 | PTG1234 | didn't see your email :) |
03:56.56 | PTG1234 | did you send it? |
03:57.03 | file[laptop] | yes |
03:57.08 | file[laptop] | maybe my SMTP server is freaking out |
03:57.28 | file[laptop] | lemme hop on gmail |
03:57.33 | PTG1234 | what would the from name show? I may have deleted it as spam, or the subject |
03:57.42 | file[laptop] | Joshua Colp |
03:58.24 | PTG1234 | k one sec let me search deleted |
03:58.24 | PTG1234 | hey you have a pda right |
03:58.24 | file[laptop] | I've got a Pocket PC GSM Phone |
03:58.37 | PTG1234 | found you |
03:58.44 | PTG1234 | moved it to inbox, so for now on you won't go in spam folder :P) |
03:58.49 | PTG1234 | you use imap mail with it? |
03:59.03 | file[laptop] | it hasn't arrived yet :) |
03:59.09 | file[laptop] | but I hope so... |
03:59.57 | PTG1234 | a bunch of that stuff i would be willing to send you but for stuff like the ups esp you wan tto pay the shipping and boxing charges? :) |
04:00.06 | PTG1234 | oh well i can't get it to pull mail from folders other then inbox |
04:00.09 | PTG1234 | thats why i was asking :) |
04:00.26 | file[laptop] | send back a list with what you can send :) |
04:00.28 | Qwell | PTG1234: gonna be around in like an hour? |
04:00.39 | PTG1234 | yah man i meant to talk to you hit me up then qwell |
04:00.52 | Qwell | alright, cool |
04:01.20 | PTG1234 | well like i probably have alot of keyboards, mice upses, bunch of scsi stuff, etc |
04:01.36 | PTG1234 | and i have 3 boxes of firbre channel drives, perfect for your storage array |
04:02.06 | file[laptop] | ooh |
04:02.21 | PTG1234 | probably 24 fibre channel drives |
04:02.22 | PTG1234 | etc |
04:02.24 | file[laptop] | like, ooooooooh |
04:02.46 | PTG1234 | but the freight is on your shoulders if you want me to send that shit :) |
04:02.46 | Micc | how do I record gsm files? |
04:02.56 | file[laptop] | PTG1234: yes yes of course |
04:03.00 | Micc | Is there an audio program for linux that will record gsm files? |
04:03.05 | PTG1234 | record a wav, use sox to convert,.. search wiki |
04:03.10 | PTG1234 | or just call your voicemail and leave a message |
04:03.14 | file[laptop] | PTG1234: just e-mail back with a yes/no beside each thing, estimated weight, and I'll gather the cost and see what I can do |
04:03.37 | PTG1234 | file[laptop]: its gonna require another trip up to my dads to figure out the weight and etc.. so it will be a few days |
04:03.46 | PTG1234 | he stores all my stuff in his garage |
04:03.52 | file[laptop] | PTG1234: not a problem |
04:03.55 | PTG1234 | anyone know anyone who would want a 7206VXR cheap? |
04:04.29 | Sedorox | whats that? |
04:04.29 | file[laptop] | I just want an e-mail back so I know exactly what you think you've got |
04:04.32 | pgpkeys | don';t even know what that is. |
04:04.35 | PTG1234 | cisco router |
04:04.44 | PTG1234 | has an OC3 interface in it |
04:04.46 | Sedorox | how much? |
04:04.53 | file[laptop] | PTG1234: am I like, cleaning you out? |
04:04.55 | pgpkeys | Sedorox: book |
04:04.59 | pgpkeys | err book |
04:05.01 | pgpkeys | damn that K |
04:05.04 | PTG1234 | file: i can only hope so :) |
04:05.08 | Sedorox | ahaha |
04:05.29 | PTG1234 | Sedorox: i don't know just want someone to offer me for it.. its a serious piece of machinery can handle an OC12 |
04:05.37 | file[laptop] | PTG1234: what was the size of those drives btw |
04:05.42 | Sedorox | $12 |
04:05.43 | Sedorox | :-p |
04:05.44 | Sedorox | j/k |
04:06.00 | PTG1234 | hah |
04:06.12 | PTG1234 | file[mac]: no idea probably 16gigs i am betting.. maybe 32gigs |
04:06.20 | file[laptop] | PTG1234: cool |
04:06.23 | PTG1234 | hey did someone see the original voice broadcasting email this guy replied to today? |
04:06.41 | PTG1234 | on -biz |
04:07.03 | file[laptop] | I try to ignore the lists |
04:11.18 | denon | PTG1234: You have a tracking number for me? <G> |
04:11.40 | file[laptop] | my internet is dying, noooooooo |
04:11.49 | Sedorox | everyone's is |
04:12.10 | pgpkeys | NOOOO! not my internet! please! i haven't surfed to the end of the internet yet! |
04:12.18 | PTG1234 | denon: its going out tommorow, i'll have it for you in morning :) |
04:12.28 | denon | ah ok |
04:12.44 | PTG1234 | sorry man :) |
04:12.50 | Sedorox | anyone remember those DSL commericals where the guy gets a popup.. "you've reached the end of the internet" |
04:12.50 | PTG1234 | just not set up to easilys hip, i am not an ebay junky |
04:12.51 | Sedorox | ? |
04:13.11 | pgpkeys | Sedorox: yeah |
04:13.12 | file[laptop] | and with all he's shipping me... ha ha ha |
04:13.18 | Sedorox | hehe |
04:13.18 | pgpkeys | that's what i was playing off of :) |
04:13.24 | Sedorox | :-p |
04:19.41 | PTG1234 | haha |
04:19.55 | PTG1234 | its ok file laptop may be answering alot of stupid * questions for me come the future :) |
04:20.01 | file[laptop] | yeah |
04:20.04 | PTG1234 | file: you know anything about shitty app_queue? :) |
04:20.14 | file[laptop] | PTG1234: unfortunately yes |
04:20.56 | PTG1234 | oh now your gonna earn your keep :) |
04:21.43 | file[laptop] | I also know that chan_sip slowly steals your sanity |
04:22.24 | PTG1234 | yah thats what i am working on now :) complete rewrite.. but was hoping to avoid that with app_queue |
04:22.34 | PTG1234 | i need to make it more virtualpbx friendly.. and iface with my new user stuff |
04:24.44 | sivana | ~sivana |
04:24.45 | jbot | i heard sivana is a putz |
04:24.45 | bugbot | sivana is assigned nothing and reported M2515. |
04:24.56 | sivana | M2515 |
04:24.56 | bugbot | M2515 is a tweak bug that is closed (markster): [patch] cleaned up cdr_mysql.c. It was filed by sivana and was last updated on 01-10-05. http://bugs.digium.com/bug_view_page.php?bug_id=2515 |
04:25.09 | sivana | M2515 open |
04:25.09 | bugbot | M2515 is a tweak bug that is closed (markster): [patch] cleaned up cdr_mysql.c. It was filed by sivana and was last updated on 01-10-05. http://bugs.digium.com/bug_view_page.php?bug_id=2515 |
04:25.13 | sivana | heh |
04:25.37 | sivana | ~bugbot |
04:25.38 | jbot | well, bugbot is a bot that gives bug statuses. You can /msg bugbot help for info or visit him on #asterisk-bugs. |
04:25.42 | bugbot | bugbot is assigned nothing and reported nothing. |
04:32.34 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
04:36.56 | timecop | heh mysql |
04:37.10 | *** join/#asterisk walnuck (~James@modemcable106.82-200-24.mc.videotron.ca) |
04:37.11 | walnuck | hi |
04:37.12 | timecop | with a high performance opensores database like mysql, you're better off logging to flat files. |
04:37.25 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
04:37.41 | dec | 'opensores'? |
04:38.58 | walnuck | dec, /s/wh |
04:39.14 | walnuck | *lol |
04:39.26 | walnuck | goddamit wtf are the channels dead? |
04:39.31 | dec | heh |
04:40.00 | walnuck | dec, know any simple routting table tinker? |
04:40.31 | dec | sorry, what do you mean by that? do I know how to setup a basic route table? |
04:40.49 | walnuck | dec, i have two nics on the card, just trying to add my laptop.. |
04:41.10 | dec | okay |
04:41.28 | dec | so you have two machines, one has two nics and one is a laptop? |
04:41.33 | walnuck | dec, the desktop has eth0 and eth1 already hwaddress mac set, nicely seen, all that's left is my router table. |
04:42.09 | walnuck | dec, the laptop has eth0 and has been successful checked with dhcp, currently i'm on the desktop |
04:42.55 | walnuck | dec, i have no firewall rules to simplify the setup, can I show my table here? |
04:43.03 | *** join/#asterisk U-238 (~U-238@CPE-138-217-33-205.vic.bigpond.net.au) |
04:43.20 | dec | in here? i don't know what the channel rules are... you can PM me with it if you wish. |
04:43.39 | dec | what's the problem you're having though? the desktop and laptop can't communicate ? |
04:44.02 | walnuck | 169.254.0.0 * 255.255.0.0 U 0 0 0 eth0 |
04:44.02 | walnuck | default 192.168.1.1 0.0.0.0 UG 0 0 0 eth0 |
04:51.17 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:59.23 | file[laptop] | goodnight all |
04:59.31 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-1-164.d4.club-internet.fr) |
04:59.42 | *** part/#asterisk U-238 (~U-238@CPE-138-217-33-205.vic.bigpond.net.au) |
05:01.40 | brc_ | hey hey |
05:01.43 | brc_ | file[laptop], |
05:02.19 | Qwell | PTG1234: still around? |
05:02.36 | PTG1234 | yah |
05:02.37 | PTG1234 | i am :) |
05:02.42 | Qwell | aim? |
05:02.44 | PTG1234 | just message me i don't look in the channel much |
05:02.45 | PTG1234 | sure |
05:03.55 | *** join/#asterisk dec (~tom@203.87.91.78) |
05:04.22 | dec | i'm back walnuck |
05:05.23 | walnuck | dec, if I want internet for my laptop and I were a guru, do I need anything else other than a proper router table and not any special progie to install? |
05:06.31 | Qwell | walnuck: Should look at the advanced networking howto on tldp.org |
05:07.03 | PTG1234 | well where are you? :) |
05:07.08 | Qwell | I IMd you. :p |
05:07.20 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
05:07.36 | FuriousGeorge | whats that bash command to list interrupts |
05:07.45 | FuriousGeorge | im trying to find out what pci bay has a free irq |
05:07.49 | drumkilla | cat /proc/interrupts |
05:09.00 | FuriousGeorge | grunkilla: thanks, i see that 4 and 6 are free, how can i find out which bay uses which interrupt, that would be in the bios, no |
05:10.10 | *** join/#asterisk michael_t (~michael_t@c-24-20-234-51.hsd1.or.comcast.net) |
05:11.22 | niZon | anyone use the cisco 7905/12 with *? |
05:15.55 | *** part/#asterisk walnuck (~James@modemcable106.82-200-24.mc.videotron.ca) |
05:17.15 | niZon | dead in here.. |
05:17.59 | remmo | no thats just your brain ;) |
05:18.56 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
05:20.43 | *** join/#asterisk danalien (~danalien@danalien.user) |
05:20.58 | michael_t | has anyone else seen this error in the asterisk console while playing a .wav? ast_waitstream_full: Wait failed (No such file or directory) |
05:21.49 | Micc | ok, so I took a wav file and ran it through sox but when I use it in asterisk it plays at a really slow speed. |
05:22.05 | Micc | Call (425)278-0757 to see what it sounds like |
05:22.54 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
05:23.20 | drumkilla | michael_t: don't specify the extension of the filename |
05:23.29 | drumkilla | if it's foo.wav ... then Playback(foo) |
05:23.29 | michael_t | i didn't |
05:24.24 | michael_t | i'm using /path/filename (w/o extension) |
05:26.05 | Micc | does asterisk play wav files too? |
05:26.18 | Micc | So I don't have to convert to gsm? |
05:27.46 | Micc | ok I get unexpected frequency now |
05:28.33 | Micc | So what format should the wav file be in? |
05:29.06 | remmo | anyone here using OSP? |
05:35.53 | *** join/#asterisk jbAU (~johnblade@c210-49-42-214.rochd2.qld.optusnet.com.au) |
05:37.30 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
05:44.52 | remmo | sigh |
05:53.29 | *** join/#asterisk dansoftware (~dansoftwa@tdata.ru) |
05:54.40 | dansoftware | Hi guys |
05:54.46 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
05:57.14 | *** join/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca) |
05:57.34 | DaLion | anyone know where script that dumps mysql to flat file configs ? or anyone used it ? |
05:57.36 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
06:03.40 | DaLion | wiki down ;( |
06:04.07 | Qwell | DaLion: mysqldump? |
06:04.21 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
06:04.24 | ellvis | hi people |
06:10.22 | remmo | lllalla |
06:10.57 | ellvis | sounds like moh :) |
06:12.46 | JerJer | do you have ASCAP and BMI rights to play that MOH file? |
06:15.24 | remmo | lol |
06:15.29 | |Vulture| | is there a way to change the pager email in voicemail.conf? |
06:15.43 | ellvis | :) |
06:16.09 | remmo | i'm trying to get OSP up and i have unresolved symbols |
06:16.54 | JerJer | |Vulture|: um vi |
06:16.57 | JerJer | insert mode |
06:17.01 | JerJer | scroll to appropriate line |
06:17.05 | JerJer | change email |
06:17.14 | JerJer | esc shift zz |
06:17.43 | remmo | i prefer vim |
06:17.49 | |Vulture| | JerJer: but its in voicemail.conf? is it the same as emailbody? |
06:17.54 | |Vulture| | I like pico :P |
06:18.14 | remmo | ewww i went pico - joe - vim - joe |
06:18.27 | *** join/#asterisk brettnem (~mive29@user-0ccsr10.cable.mindspring.com) |
06:18.29 | |Vulture| | Ive just been using pico for so long |
06:18.38 | brettnem | EK pico?! |
06:18.44 | brettnem | good morning all |
06:18.46 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
06:19.05 | *** join/#asterisk qkit (~qkit@60.48.11.74) |
06:19.16 | yxa | you shouldnt need to type 4 letters to get to an editor :) |
06:19.16 | brettnem | ~seen bkw_ |
06:19.18 | jbot | bkw_ <~brian@bkw.developer.and.friend.of.asterisk> was last seen on IRC in channel #asterisk, 6h 15m 12s ago, saying: 'thats one thing you'll see'. |
06:19.26 | bugbot | seen bkw_ is assigned nothing and reported nothing. |
06:19.35 | brettnem | yxa: heh, like "emacs" ? |
06:20.04 | JerJer | word |
06:20.27 | brettnem | ha |
06:20.28 | yxa | heh |
06:20.49 | qkit | hey guys? i wonder what can be difference in using h323,sip or iax protocol? i have read the readme in the source but still can really understand it......what can be the difference of it in using in asterisk???? |
06:20.58 | qkit | can=cant |
06:21.24 | brettnem | support and easy of use |
06:21.29 | brettnem | nat traversability |
06:21.38 | brettnem | maturity |
06:22.12 | The_Ape | Have anyone tried the presense support in eyebeam with asterisk? i cant get it to work, everyone is constantly online(even when they are not). |
06:22.14 | qkit | but will it have any problem with other system? as it used its own protocol and not the standard protocol |
06:22.35 | brettnem | of course it will have problems talking to devices that don't support IAX |
06:22.39 | qkit | what can be the disadvantace of the sip or h323 protocol |
06:22.40 | brettnem | don't use it then |
06:23.05 | brettnem | h323 support really sucks on asterisk (I think that's due to problems in the actual stack which isn't part of asterisk at all) |
06:23.25 | brettnem | SIP dosen't support real line side signalling.. like flashhooks |
06:23.35 | remmo | hey h323 is not that bad |
06:23.46 | brettnem | heh.. I knew I'd offend someone! :) |
06:24.04 | brettnem | have you seen zoa's performance charts for h323?? |
06:24.10 | remmo | not i, just had to bust my chops to get it working with my provider |
06:24.13 | remmo | no |
06:24.20 | brettnem | pretty nasty stuff! |
06:24.24 | remmo | url? |
06:24.43 | brettnem | wish I knew.. maybe on the astricon site.. hey, actually he has some astertest site I think.. |
06:24.58 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
06:25.09 | brettnem | check out www.astertest.com |
06:25.14 | *** join/#asterisk znoG (gs@200.115.216.109) |
06:29.08 | |Vulture| | anyone know of the top of their head the reboot keys for the polycom ip500.. its like voicemail + mute, then the 2 volume keys? |
06:29.38 | brettnem | |Vulture|: try vol down, vol up, hold and... um.. messages? |
06:29.45 | |Vulture| | yea thats it |
06:29.46 | |Vulture| | thanx |
06:30.01 | |Vulture| | I am writing a letter to someone to do it and I don't have one in front of me |
06:30.08 | remmo | `hmmm |
06:30.43 | brettnem | hmm? |
06:33.01 | remmo | hmmm how to open powerpoint in fbsd |
06:33.06 | remmo | openoffice |
06:33.54 | brettnem | heh |
06:34.11 | brettnem | it's a windows world we live in |
06:34.58 | brettnem | anyone in here used SEMS? |
06:35.21 | brettnem | lets bring on the competition |
06:35.32 | remmo | WINDOWS SUX |
06:35.58 | brettnem | yeah, well |
06:36.20 | tessier | remmo: You are preaching to the choir, my friend |
06:38.25 | brettnem | so no one has used SEMS? |
06:38.53 | brettnem | how is it that a room with 289 members is so quiet? |
06:38.55 | remmo | tessier: well hey what can i say |
06:39.05 | JerJer | friday night |
06:39.13 | remmo | they are sleeping |
06:39.55 | JerJer | Add warning for _. match (bug #4032) <--- THANK YOU |
06:40.05 | brettnem | isn't it friday morning? |
06:40.13 | brettnem | oh, nice! |
06:41.00 | PTG1234 | anyone in here do any pocketpc development? |
06:41.11 | brettnem | cross compiling asterisk? |
06:41.18 | PTG1234 | no for something else :) |
06:41.32 | remmo | c# |
06:41.58 | PTG1234 | that a language they use alot for pocketpc? |
06:42.16 | *** join/#asterisk gres (~serg@81.222.48.242) |
06:42.28 | ellvis | pocketpc? ehm, palm rulez, not pocketpc :) |
06:42.35 | remmo | yup |
06:42.41 | PTG1234 | any idea what compiler? |
06:42.52 | remmo | msdn.microsoft.com |
06:43.07 | remmo | they have a free ide called visual studio express beta |
06:43.11 | PTG1234 | m$ compiler |
06:43.16 | remmo | free |
06:43.32 | remmo | but its missing dotNet stuff |
06:43.35 | PTG1234 | i probably have visual studio but its a pain |
06:43.42 | brettnem | oh.. make it free to stamp out all the little guys making a business selling compilers |
06:43.49 | remmo | i have been using it and its not that bad |
06:44.00 | brettnem | soon they'll be giving away clothes and shoes for free and put Walmart out of business |
06:44.01 | PTG1234 | whats does C# add to the c language? |
06:44.09 | remmo | not much |
06:44.14 | ellvis | PTG1234: java mess:) |
06:44.20 | remmo | more like c++ , java, php |
06:44.28 | PTG1234 | ugh :) |
06:44.36 | brettnem | java.. barf |
06:44.37 | remmo | its actually really good when you get your head around it |
06:44.38 | PTG1234 | are you sure that compiler will work on pocketpc as well? |
06:45.13 | remmo | pretty sure. if its ppc 2003 should be right |
06:45.34 | remmo | i have only been doing console stuff with it, but does forms and all |
06:45.48 | brettnem | ok.. I'm going to give sems a shot for voicemail.. wahoo |
06:45.52 | remmo | and the XML stuff, well lets just say hats off to m$ |
06:45.57 | PTG1234 | b/c i thought it was usually an addon |
06:46.30 | remmo | b/c? |
06:46.31 | PTG1234 | i need like the simpliest program for pocketpc, but i am not sure what would be the easiest route to do it |
06:46.37 | PTG1234 | how is java's support on pocketpc |
06:46.45 | PTG1234 | then it could work on palm as well :) |
06:46.59 | ellvis | java suxx :) |
06:47.04 | remmo | ewww java. good luck on ppc |
06:47.11 | brettnem | ellvis: agreed!!! |
06:47.13 | remmo | c# for simple |
06:47.26 | brettnem | quickbasic, all the way |
06:47.31 | remmo | never thought i would be a m$ advocate |
06:49.17 | drumkilla | must be bored :p |
06:49.21 | elric | does ${DIALSTATUS} exist in CVS head only or does stable release have it as well? |
06:49.38 | brettnem | it should be in stable.. it's been around for a while |
06:49.46 | drumkilla | yeah, it's in stable |
06:49.50 | elric | alright, thanks |
06:50.13 | drumkilla | show application dial ;) |
06:53.21 | elric | will Dial(${EXTEN}|60|gM(detect)^DIALSTATUS) work? |
06:54.06 | elric | it doesnt seem to pass DIALSTATUS to the macro |
06:54.54 | *** join/#asterisk TheEmperor (~user@203.121.47.165) |
06:56.15 | elric | s-${DIALSTATUS} shows up as s- on the CLI |
06:56.47 | elric | oh |
06:56.57 | elric | i forgot to read upon completion |
06:56.59 | elric | :| |
06:58.50 | *** join/#asterisk luke-jr_ (~luke-jr@207.192.221.172) |
07:12.32 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
07:14.11 | *** join/#asterisk WorkTooMuch (~work@82.148.188.1) |
07:14.39 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
07:14.40 | RoyK | anyone that knows what it'll take to support CallingPres on SIP? |
07:15.48 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
07:16.00 | clive- | mwhois atacom |
07:17.58 | dec | grr |
07:18.02 | dec | silly SIP |
07:18.04 | dec | not working :( |
07:18.09 | *** join/#asterisk pbxjunkie (~nkatzakis@videocomputer.gr) |
07:18.15 | pbxjunkie | howdy hey:D |
07:19.12 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
07:20.10 | PTG1234 | does gmail support imap? |
07:20.45 | elric | ah the wiki is down |
07:21.04 | elric | its a good thing in a way, now I can take a break |
07:21.57 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-033-191.arcor-ip.net) |
07:23.10 | brettnem | royk.. callingpres=rpid |
07:24.06 | RoyK | brettnem: er. what is rpid? I thought chan_sip didn't support callingpres. |
07:24.36 | RoyK | brettnem: or is this in HEAD? |
07:24.47 | brettnem | oh, aybe asterisk dosen't but sip does |
07:24.54 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
07:25.04 | RoyK | Martohtar: morgen |
07:25.30 | *** join/#asterisk gres (~serg@81.222.48.242) |
07:29.25 | Qwell | PTG1234: just pop3 I think |
07:30.46 | facek_ | i am looking for two sound files.. beep (for good) and beep (for wrong) |
07:30.49 | facek_ | do anybody have? |
07:30.57 | wildgoose | My music on hold is blaring loud and distorted. Setup says to use quietmp3. Any thoughts on why? |
07:31.40 | pbxjunkie | wildgoose: there is an issue of compatibility between versions of mpg123 |
07:31.55 | pbxjunkie | to my knowledge the LATEST version of mpg123 causes problems with asterisk |
07:32.13 | *** join/#asterisk emitrax (~emitrax@ingnatdyn33.unime.it) |
07:32.24 | Qwell | 0.59r is the latest stable, which is the only version thats supported |
07:32.34 | pbxjunkie | yea , that's the one |
07:32.50 | *** join/#asterisk iamcool (Omega11@69-165-65-219.sbtnvt.adelphia.net) |
07:33.11 | wildgoose | ok thanks |
07:33.14 | *** part/#asterisk iamcool (Omega11@69-165-65-219.sbtnvt.adelphia.net) |
07:33.23 | wildgoose | doesn't that have some security issues thouhg? |
07:33.42 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
07:34.19 | |Vulture| | 0.59r is what is grabbed with make mpg123 |
07:34.29 | |Vulture| | "make mpg123" is the safest way |
07:34.59 | wildgoose | is mpg321 supported instead? |
07:35.09 | RoyK | no.... |
07:35.18 | foobos | only mpg123 r59 is supported, no other version or product |
07:35.24 | RoyK | but there's native mp3 support in -addons |
07:35.57 | |Vulture| | anyone know if there is a patch to be able to view the status of a Zap card through the manager interface... thinking about incorperating it in to Nagios |
07:36.00 | The_Ape | on my system the symlink to mpg321 wasn't overwriten when i did make mpg123. Might be an idea to remove that manually first. |
07:36.07 | |Vulture| | unless there is already a solution |
07:36.28 | |Vulture| | so if say a T1 drops... you will be able to get a notice |
07:36.51 | *** part/#asterisk mozrat (~mozrat@80.68.89.215) |
07:37.05 | RoyK | |Vulture|: I guess you can start off with my nagios plugin and do a pro show span 1 or something |
07:37.44 | |Vulture| | RoyK: oh thats your plugin? the IAX one? it works great! |
07:37.53 | |Vulture| | yea I was thinking about that... |
07:38.18 | RoyK | it does iax and manager |
07:38.22 | |Vulture| | wasn't it JunkY who coded the show span utility? |
07:38.31 | RoyK | then use the manager interface to check the pris |
07:38.51 | |Vulture| | I haven't tried the manager part just hitting IAX to check up status of * |
07:39.28 | RoyK | http://karlsbakk.net/asterisk/ |
07:39.34 | RoyK | get the plugin from there |
07:40.10 | RoyK | hit. is the wiki down again? |
07:40.33 | |Vulture| | yea |
07:40.38 | |Vulture| | went down about 2 hrs ago |
07:41.21 | RoyK | shit |
07:41.38 | RoyK | someone ought to replace that |
07:41.45 | RoyK | or move it, I mean |
07:41.58 | drumkilla | I wish we had one that used MediaWiki |
07:42.02 | drumkilla | it looks so much better :) |
07:42.06 | *** join/#asterisk heison (~heison@p85.n-lapop06.stsn.com) |
07:42.33 | RoyK | well. tikiwiki does the job |
07:42.40 | |Vulture| | when its up :P |
07:42.52 | RoyK | only you need to use google to search for something. the tiki search sucks big tim |
07:42.53 | RoyK | e |
07:42.55 | |Vulture| | its usually up... just seems like when it goes down.... it dies |
07:43.11 | |Vulture| | RoyK: agreed... searching for a .conf never brings it up |
07:43.25 | |Vulture| | Ive just been using google now and using cache from the wiki since its down |
07:44.13 | drumkilla | you people and your attachment to that damn wiki :p |
07:44.33 | RoyK | it's a nice docs db |
07:45.05 | drumkilla | yeah, its unfortunately the most complete thing out there, heh |
07:47.50 | wildgoose | seems that the latest version of mpg123 WILL work, but not the -mmx version. Change it to the 486 or generic version in the symlink and it's playing fine here |
07:48.07 | wildgoose | My guess is that the rescale flag is broken in the latest mmx version |
07:49.16 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
07:54.09 | fenlander | Hello, Is there a problem with the asterisk mailing lists? I've not had any mail from them in the last 17 hours |
07:55.41 | drumkilla | i'm getting them |
07:56.51 | *** join/#asterisk nitram (nitram@superblob.com) |
07:57.14 | fenlander | Hmm. I seem to be getting mail from other lists. What have I broken this time. |
07:59.00 | emitrax | I can't get my 7940 registered with asterisk. On the status messages of the phone I get: E640 REG msg unsupported |
07:59.23 | emitrax | I m useing SIP 7.0 as firmware |
07:59.55 | newl | I thought SIP was only up to 2.0. :) |
08:00.24 | fenlander | That explains the mail problem. Google has started marking them all as spam. |
08:01.11 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
08:04.27 | PTG1234 | i am missing emails from mailing list |
08:04.28 | PTG1234 | i have no idea |
08:04.53 | elric | has anyone compiled the iax library on FreeBSD |
08:05.24 | fenlander | Gmail is marking all my mail from asterisk-users, dev and biz as spam |
08:06.31 | drumkilla | no cvs?! |
08:09.16 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
08:09.29 | PTG1234 | who needs cvs when you got stable :) |
08:09.53 | drumkilla | you can track stable on the cvs mailing list, too :) |
08:10.18 | PTG1234 | heh |
08:10.35 | PTG1234 | i am now wiring my chan_sip for stable.... i got sick of cvs bugs |
08:10.52 | PTG1234 | you know is an ipod mini the way to go |
08:10.58 | PTG1234 | anyone have any mp3 player opinions |
08:10.59 | PTG1234 | ? :) |
08:11.05 | drumkilla | i got an ipod photo |
08:11.12 | PTG1234 | i had an ipod |
08:11.16 | PTG1234 | it broke |
08:11.18 | PTG1234 | those minis look slick |
08:11.49 | *** join/#asterisk jwitte (~jwitte@port-212-202-101-206.static.qsc.de) |
08:11.53 | drumkilla | but, but ... ONLY 4 GB! |
08:12.14 | PTG1234 | 6gb :) |
08:12.22 | PTG1234 | do i really need more then that :) |
08:12.43 | PTG1234 | 18 hours of battery life |
08:12.45 | PTG1234 | thats pretty strong |
08:14.27 | *** join/#asterisk pino (~z@host241-115.pool80116.interbusiness.it) |
08:15.53 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
08:15.58 | rvhi | hi |
08:16.25 | rvhi | anyone has paging working by sending sip call with "auto answer"? |
08:16.51 | rvhi | i am going to post a bounty on this. |
08:16.58 | rvhi | $200 |
08:17.10 | DaLion | yeah winamp rocks |
08:17.25 | DaLion | this #mp3playersforsale now ? |
08:17.28 | PTG1234 | yah i will just throw my laptop in my pocket |
08:17.28 | DaLion | ;) |
08:18.05 | *** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-65337.adsl.ttnet.net.tr) |
08:18.21 | pino | rvhi: what client are you using? |
08:18.25 | *** part/#asterisk glLoadIdentity (~tuyan@dsl81-214-65337.adsl.ttnet.net.tr) |
08:18.27 | rvhi | polycom |
08:18.43 | DaLion | anyone tried the testsuite ? |
08:18.46 | DaLion | for sip |
08:18.47 | *** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-65337.adsl.ttnet.net.tr) |
08:18.48 | DaLion | and iax |
08:19.01 | DaLion | btw got my mysql cluster up |
08:19.07 | DaLion | was pretty fun |
08:19.07 | pino | rvhi: i see that there's already something on the wiki, but i can't load any page from it right now :( |
08:19.19 | DaLion | yeah wiki is down/slow/down tonight |
08:20.13 | rvhi | i saw some basic code in wiki |
08:20.41 | rvhi | i want the ability to do zone page |
08:21.05 | rvhi | web front to add/remove zones, and add/remove users from a zone |
08:21.43 | rvhi | so it records a message, then does paging |
08:25.09 | pino | i think you need the first configuration files for the polycom, first. |
08:25.18 | pino | e.g. http://www.kriscompanies.com/modules.php?name=Downloads&d_op=viewdownload&cid=1 |
08:25.33 | pino | s/first/right/ |
08:25.38 | *** join/#asterisk jonathh (~asd@217.46.145.65) |
08:26.00 | jonathh | morning gents |
08:26.06 | rvhi | polycom can auto answer, it is not a problem |
08:26.13 | rvhi | i got it figured out |
08:26.24 | rvhi | the * part is my issue |
08:27.03 | pino | so the polycom already gets you on the speaker? |
08:27.31 | rvhi | y |
08:27.51 | pino | you just need to connect the same channel to multiple phones? |
08:28.33 | rvhi | i don't know how to do it in * |
08:28.58 | jonathh | Can someone tell me.. if the wilcard x100p is being discontinued. what is it being replaced with? |
08:29.53 | drumkilla | it has been discontinued for a while |
08:29.57 | drumkilla | the TDM400P is its replacement |
08:30.38 | jonathh | the price difference from what i can see is HUGE |
08:30.59 | jonathh | all i need is a PCI FXO (i think) pc to phone line.. to play about with |
08:31.26 | jonathh | the X100p i have seen for $6~ |
08:31.44 | drumkilla | but you're not getting a Digium card |
08:32.02 | jonathh | if it works with asterisk? does it matter at this stage? |
08:32.07 | pino | rvhi: i'd try to pick a MeetMe room and connect each extension in the zone in monitor mode ('m' option). |
08:32.35 | jonathh | once i have a working model and am making millons i'll plouge some back... but i cant afford to just yet. |
08:32.51 | |Vulture| | jonas: if you just playing with it, nothing more than testing.. its fine, but if you plan on using it for anything production wise.. don't count on it |
08:33.15 | jonathh | i am just playing with it currently |
08:33.15 | |Vulture| | urg jonathh |
08:33.18 | jonathh | on my home line.. |
08:33.26 | |Vulture| | yea then by all means... |
08:33.57 | |Vulture| | just its funny to see people trying to throw 6 X100P clones into a box |
08:34.07 | jonathh | the TDM10B: which has 1 FXS port.. like the Zaptel x100p does(did)... |
08:34.14 | jonathh | but the price is huge.. |
08:34.17 | jonathh | what happens then? |
08:34.39 | |Vulture| | jonathh: with 6 clones... well I don't think more than 3 work |
08:34.47 | jonathh | oh right.. |
08:34.49 | jonathh | why is that? |
08:34.53 | jonathh | addressing? |
08:34.54 | |Vulture| | not a clue |
08:34.59 | jonathh | oh :) |
08:35.00 | |Vulture| | possibly |
08:35.05 | |Vulture| | never tried it myself |
08:35.09 | drumkilla | jonathh: the x100p is FXO |
08:35.20 | |Vulture| | only have had 2 x100p digiums in a box at once |
08:35.24 | jonathh | that is where getting more FXO modules for the TDM card owrks |
08:35.25 | |Vulture| | that was just testing |
08:35.34 | jonathh | yeah sorry drumkilla |
08:35.46 | jonathh | i mean the port that goes from the PC to the phone line..... |
08:35.52 | |Vulture| | I am about to have a buncha FXO modules for sale.. moving most of my offices to TE110P cards |
08:35.53 | drumkilla | no prob, just don't want you confused |
08:35.55 | jonathh | did it work? |
08:36.11 | jonathh | modules for the TDM? |
08:36.31 | |Vulture| | yea |
08:36.54 | jonathh | if your floggin'um on ebay.. make sure you notify the room! |
08:36.55 | jonathh | :) |
08:37.01 | |Vulture| | 21 modules to be exact |
08:37.06 | |Vulture| | will do |
08:37.06 | jonathh | where in the world are you? |
08:37.17 | |Vulture| | Orlando, FL |
08:37.23 | jonathh | ahh |
08:37.30 | jonathh | Brighton, UK :) |
08:37.50 | |Vulture| | ah... yes our neighbors across the pond ;) |
08:37.56 | jonathh | :) indeedy |
08:38.01 | jonathh | what i need to do |
08:38.17 | jonathh | is talk my employers into letting me disconnct our PBX |
08:38.31 | jonathh | so i can play about with an interface to a digital line |
08:38.37 | jonathh | (which i know nothing about) |
08:38.43 | |Vulture| | where * just blows away people is in new installs |
08:38.45 | jonathh | .. i dont think they will let me :)_ |
08:39.00 | jonathh | i am amazed by the potential it has |
08:39.02 | jonathh | what do you do? |
08:39.06 | |Vulture| | you have a E1 going in there? |
08:39.06 | *** join/#asterisk pranav (pranav@221.128.181.21) |
08:39.21 | drumkilla | be careful, you will become addicted to asterisk |
08:39.25 | drumkilla | it will consume your life |
08:39.28 | |Vulture| | I use Polycom IP500s and Dell servers |
08:39.30 | jonathh | im not sure of the terminology.. ISDN something or other |
08:39.36 | drumkilla | and you will stay up all night working on it |
08:39.37 | jonathh | 15lines in.. |
08:39.39 | jonathh | 5 out |
08:39.44 | |Vulture| | drumkilla: I just redesigned my dialplan for the 3 time since I started today |
08:39.45 | jonathh | doning that already! |
08:39.53 | pranav | hello everuone |
08:39.55 | |Vulture| | drumkilla: but thats the 3rd time in a year... not too bad |
08:39.57 | jonathh | fasinating stuff |
08:39.58 | drumkilla | I went on a coding frenzy this week |
08:40.22 | |Vulture| | its amazing you learn a little then look back at your code like... dear lord what was I thinking? |
08:40.26 | jonathh | just trying to decide what actual hardware to buy.. for the first way of playing with actual hardware |
08:40.28 | pranav | tell me the programing that we use in the extensions.conf is written in which language |
08:40.45 | drumkilla | its not a language |
08:40.46 | DaLion | in asterisk |
08:40.47 | DaLion | ;) |
08:40.52 | DaLion | they call it CLI |
08:40.52 | |Vulture| | jonathh: cheap PC and a x100p clone... |
08:40.55 | DaLion | ;) j/k |
08:40.58 | drumkilla | the AEL ... asterisk extensions language |
08:41.02 | drumkilla | :) |
08:41.03 | DaLion | lol |
08:41.07 | pranav | ok |
08:41.17 | pino | ADDL, asterisk dialplan definition language |
08:41.17 | DaLion | i love the wtf is this language when it crashes for nothing |
08:41.23 | jonathh | yeah.. so far i got a shit box pc.. x100p clone... 1 cheapy sip handset.. 1 ata (probably the grandsteam) |
08:41.27 | pino | maybe we can come up with something even better ... ;) |
08:41.29 | |Vulture| | jonathh: you mine as well get a nice IP phone though |
08:41.30 | pranav | so is there any prior requirement sbefor learning this language |
08:41.47 | DaLion | bah .. even on dual xeons 2 gigers u get problems |
08:41.47 | drumkilla | no, it is fairly simple |
08:41.49 | jonathh | so far as in that is what we are gonna buy |
08:41.58 | pranav | ok thanks |
08:41.59 | DaLion | a simple %%%#$@!# shit in diaplan can cause crashes |
08:42.08 | smiley- | is there any way to update fields in SQL (real time stuff) from extensions.conf ? like realtime update in the CLI? |
08:42.20 | |Vulture| | jonathh: okay Id recommend looking at IP500 or Cisco 7940 phones when you go to do a demo |
08:42.23 | DaLion | yes |
08:42.23 | DaLion | odbc something |
08:42.25 | pino | DaLion: and if you can't get it, there's app_segfault! :D |
08:42.28 | riksta | DaLion: you really shouldn't shit in your dial plan |
08:42.33 | jonathh | yeah the cisco look nicccce. |
08:42.36 | |Vulture| | I pitch the IP500s cause they are so cheap for what you get |
08:42.39 | drumkilla | smiley-: realtime doesn't have a way to do updates like that |
08:42.48 | jonathh | lemme make some notes :) |
08:42.50 | DaLion | hehe |
08:42.54 | riksta | jonathh: they are, i know :) |
08:43.05 | |Vulture| | jonathh: Cisco has the name, but Polycom IP500 is where it is at! just ask the channel |
08:43.14 | jonathh | i'll take a look |
08:43.32 | DaLion | x: realtime update sipfriends name bobsphone port 4343 |
08:43.33 | jonathh | they are abit pricey for me just now tho.. £200 compared to £70 for a shitty grandstream |
08:43.36 | |Vulture| | me saying all this and I have a 7960 in front of me... cause I haven't sprung for a IP600 yet |
08:43.40 | DaLion | that would work |
08:43.53 | DaLion | smiley- ? x: realtime update sipfriends name bobsphone port 4343 |
08:43.55 | jonathh | so who make the IP500? |
08:43.58 | smiley- | drumkilla: ok.. since it's possible from the CLI and voicemailmain.. oh well.. I guess I have to do a small .c-application then.. |
08:43.59 | riksta | i have a 7940, but i think it looks nicer than the IP500 |
08:44.00 | |Vulture| | jonathh: true but they are very nice phones... Polycom |
08:44.05 | riksta | jonathh: polycom |
08:44.06 | jonathh | ok |
08:44.09 | jonathh | lemme google i |
08:44.11 | smiley- | DaLion: from extensions.conf ? |
08:44.13 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
08:44.23 | DaLion | no in CLI |
08:44.29 | DaLion | asterick -vvvvvvvgrc |
08:44.33 | |Vulture| | the new firmware for the IP500s suck... anyone else try it? |
08:44.35 | smiley- | DaLion: yeah.. I know that ;) |
08:44.39 | DaLion | CLI> realtime update sipfriends name bobsphone port 4343 |
08:44.47 | smiley- | DaLion: but I wanted to do that from extensions.conf |
08:44.55 | |Vulture| | I think its like 1.5 bootrom and 1.4.1 SIP... |
08:44.56 | drumkilla | that could be converted into an app easily |
08:45.03 | DaLion | oh |
08:45.03 | DaLion | yeah |
08:45.11 | DaLion | odbc set or something check the F. docs |
08:45.12 | DaLion | ;) |
08:45.28 | DaLion | try odbcget |
08:45.32 | DaLion | or something |
08:45.41 | smiley- | hehe.. ok ;) |
08:45.46 | |Vulture| | I just went through my DP today and make it much easier on CDR... so now its easy to track extensions |
08:46.07 | jonathh | forgive me if this is a obviously discovered answer. .but it has just struck me. can asterisk store the voicemail in a DB? |
08:46.24 | |Vulture| | files or users? |
08:46.44 | jonathh | the actual voicemail files. |
08:46.50 | |Vulture| | users yes, files... not sure but don't think so |
08:46.54 | drumkilla | in cvs head, yes |
08:46.54 | jonathh | i am aware you can store extention stuff in a DB |
08:47.07 | jonathh | oh version is the head at? |
08:47.13 | |Vulture| | *mutters* everything is in head |
08:47.14 | jonathh | ^what version |
08:47.31 | drumkilla | cvs head and the 1.0 branch are surely very different |
08:47.31 | jonathh | but presumably the head isn't all the stable..... |
08:47.41 | elric | does ${DIALSTATUS} get set after the call is completed? |
08:48.01 | |Vulture| | yea I run v1-0 |
08:48.07 | DaLion | smiley- |
08:48.07 | DaLion | ODBCput(family/key=value): |
08:48.19 | DaLion | Stores the given value in the Asterisk database. Always returns 0. |
08:48.20 | jonathh | in what ways the 1.0 branch and the hdea different? |
08:48.27 | *** join/#asterisk basta (~kqj@62-101-126-233.fastres.net) |
08:48.38 | smiley- | DaLion: ah.. that might be a way |
08:49.11 | smiley- | except for that I don't use ODBC for the real time stuff ;) |
08:49.26 | basta | anyone using cisco 7912/7960 ? I've a problem with music on hold ... |
08:50.01 | *** part/#asterisk pranav (pranav@221.128.181.21) |
08:50.03 | |Vulture| | 7960 here |
08:50.06 | jonathh | what version is the head upto? |
08:50.11 | DaLion | exten => 111,1,ODBCput(sipfriends/password=1234): |
08:50.12 | DaLion | hehe |
08:50.17 | DaLion | not sure anout that one |
08:50.21 | drumkilla | cvs head isn't versioned |
08:50.28 | jonathh | oh right |
08:50.36 | drumkilla | at this point, anyway |
08:50.41 | jonathh | ok |
08:51.02 | drumkilla | though it is commonly referred to as the branch for the future 1.2 ... |
08:51.04 | jonathh | is there a world of difference between 1.0.7 and the head? |
08:51.10 | basta | vulture, what version o * ? i think moh stopped working after an upgrade to 1.0.6 |
08:51.11 | drumkilla | or maybe the 1.1 dev branch |
08:51.17 | drumkilla | but most people just call it cvs head |
08:51.38 | drumkilla | basta: upgrade to cvs head |
08:51.44 | drumkilla | basta: no |
08:51.48 | drumkilla | basta: i meant 1.0.7 |
08:52.06 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
08:52.30 | basta | k, already compiled it, I'll try it (now i can't restart the servers) |
08:52.39 | pino | is anyone using *+spandsp to send faxes with a decent interface? |
08:52.43 | jonathh | i know this isn't really relivant.. but what distro are peeps using asterisk with? |
08:52.48 | drumkilla | there was a moh problem in 1.0.6, 1.0.7 should fix it for you |
08:52.57 | drumkilla | jonathh: whatever you want |
08:53.25 | basta | hope it will, everyone hanging up calls, thinking the line is down ! |
08:53.28 | DaLion | http://www.planetwayne.com/forums/viewtopic.php?t=211&sid=f308ab865deff3c83638ab9b15c40e2b |
08:53.33 | DaLion | this guys an idiot |
08:53.37 | jonathh | wondered if there were any comments to be made about what distro to use if it is client side.. and only running asterosk |
08:53.45 | DaLion | garage doror openein based on asterisk ? |
08:53.47 | jonathh | current i have a red-hat, gentoo, and feudora |
08:53.50 | DaLion | hmm... |
08:53.58 | DaLion | great way to get robbed |
08:54.00 | |Vulture| | okay bedtime |
08:54.01 | |Vulture| | night guys |
08:54.04 | jonathh | it needs to be lean |
08:54.06 | jonathh | night d00d |
08:55.03 | *** join/#asterisk cinzas (~ashes@83.240.144.145) |
08:55.07 | cinzas | g'morning |
08:55.16 | jonathh | morning |
08:55.36 | cinzas | I need help ... hehe |
08:55.41 | jonathh | dont we all :) |
08:55.43 | cinzas | ast_readaudio_callback: Failed to write frame |
08:55.50 | cinzas | I'm getting this error |
08:55.52 | jonathh | i get that.. |
08:55.57 | drumkilla | what channel driver |
08:55.58 | jonathh | dunno what it means :) |
08:56.08 | cinzas | drumkilla: sip |
08:56.27 | cinzas | and strange thins are happening ... |
08:56.39 | jonathh | when i get it.. the sound goes all shoppy |
08:57.10 | cinzas | I'm running * with 3 callcenters and voicemail |
08:57.35 | cinzas | Average of 500-600 minutes by day |
08:57.47 | cinzas | With a trunk SIP to Cisco Call Manager |
08:58.06 | cinzas | Yesterday i got strange errors. |
08:58.37 | cinzas | Sometime when a calls goes to a queue, it starts ringing a member, when he pickups the call is dropped and starts ringing in other queue member |
08:59.05 | cinzas | ANd if he pickup, the call jump to ohter member. The ppl here are getting crazy with calls jumping phone by phone |
09:02.05 | cinzas | anyone ? |
09:02.34 | *** join/#asterisk Newbie___ (me@60.48.55.141) |
09:02.56 | jonathh | sorry dude. |
09:03.37 | cinzas | mail time ;) |
09:04.07 | jonathh | i rekon your gonna be busy this weekend though :) |
09:04.32 | cinzas | brbrbrbr |
09:04.37 | cinzas | Please dont |
09:04.39 | cinzas | lol |
09:05.03 | Newbie___ | hi, my X101P is successfully recognized by *, but i can dial out, * gives me " Unable to create channel of type 'Zap'" |
09:05.09 | Newbie___ | please help |
09:05.19 | Newbie___ | i mean can't dial out |
09:07.13 | elric | is there a way to check if a Dial()'ed call has been answered? |
09:07.43 | jonathh | ask the person your calling? :) |
09:08.03 | elric | well i wish extensions.conf could do that and then execute the macro |
09:08.10 | elric | :) |
09:08.22 | drumkilla | elric: there is an option to Dial to do that |
09:08.39 | elric | ah ok drumkilla, i just need to read more on it then |
09:08.42 | drumkilla | might be only cvs head, though ... |
09:08.48 | drumkilla | show application dial ... look for it in there |
09:09.45 | elric | because Dial(${EXTEN}|60|M(macro)) executes as soon as the call connects. anyway i will read up on it |
09:09.50 | elric | and get cvs head |
09:10.02 | drumkilla | oh, just kidding |
09:10.05 | drumkilla | that's what I was talking about ... |
09:10.46 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
09:12.07 | elric | hrm coz M() doesnt wait till the call is actaully answered |
09:12.27 | elric | althought there is an option r |
09:12.43 | elric | ringing but that stops all other callprogress info |
09:18.41 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
09:19.52 | facek_ | i have this extensions |
09:19.52 | facek_ | exten => _0XXXXXXXXX,1,Dial(SIP/202&SIP/203) |
09:19.53 | facek_ | exten => _0XXXXXXXXX,2,Hangup |
09:20.08 | facek_ | and why when SIP/202 answer the call.. the caller still have beep beep |
09:20.09 | facek_ | ? |
09:21.00 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
09:28.22 | emitrax | does anyone have cisco 7940 here? Im having in trouble with the registration process |
09:28.39 | emitrax | Im using 7.0 as firmware version |
09:30.07 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
09:33.06 | riksta | emitrax: yeah |
09:33.42 | riksta | emitrax: pastebin your sip.conf part and the cisco .cnf |
09:36.18 | emitrax | I dont have a cisco.cnf |
09:36.21 | tessier | With RealTime why aren't the contexts stored in the db also? Why do we have to still have [context] lines and then a switch statement in sip.conf and extensions.cnf/ |
09:36.23 | tessier | ? |
09:36.51 | tessier | And since each sip device has its own context in sip.conf do I still need a [sipphone] and a switch for each phone in sip.conf? |
09:38.54 | facek_ | Why asterisk couldn't bridhe incoming call on ZAP. when SIP peer answered? |
09:39.42 | TheEmperor | if i've got 4 fxo ports, how do i make it so that in the extensions.conf file asterisk will use the free zap channel to dial out? |
09:39.57 | TheEmperor | dial zap/2/3/4/5? |
09:40.08 | drumkilla | dial zap/g1 |
09:40.13 | drumkilla | where all the channels are in group=1 |
09:40.27 | TheEmperor | considering it's a 4 port fxo card? |
09:40.43 | drumkilla | yeah, that should work |
09:40.54 | TheEmperor | g1 means any available channel? |
09:41.07 | drumkilla | yeah, any channel in group 1 |
09:41.13 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
09:41.25 | TheEmperor | do i need to define that in zapata and zaptel? |
09:43.08 | drumkilla | just zapata |
09:44.36 | TheEmperor | thanks drumkilla |
09:45.30 | TheEmperor | for signalling=fxo_ls for outgoing calls? and then channel=>1-4 is that correct? |
09:45.41 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
09:45.42 | drumkilla | signalling=fxs_ls |
09:45.59 | TheEmperor | ah.. |
09:46.01 | drumkilla | fxo uses fxs signalling and vise-versa |
09:46.27 | TheEmperor | signalling=fxs_ls; group=1; channel=>1-4 |
09:46.30 | TheEmperor | is that right? |
09:46.34 | drumkilla | looks good |
09:47.03 | TheEmperor | sweet. thank you :) |
09:47.09 | drumkilla | noooo problem |
09:47.14 | drumkilla | you can thank my insomnia |
09:47.54 | TheEmperor | should i put group=1 on top instead? then signalling=fxs_ls ; channel =>1-4? |
09:48.13 | drumkilla | as long as its before channel |
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09:48.55 | TheEmperor | so in that order just now is ok? |
09:49.18 | pbxjunkie | anybody got any experience with zaptel and asterisk?:) |
09:49.24 | drumkilla | yup |
09:49.29 | cypromis | pbxjunkie: nobody |
09:49.31 | pbxjunkie | :D |
09:49.55 | pbxjunkie | my boss is SO going to fire me if I don't get it working soon :D |
09:50.37 | tessier | Anyone know how context includes are handled when using realtime? |
09:50.44 | drumkilla | pbxjunkie: support@digium.com |
09:50.48 | tessier | Do you still have to put the include => line in the extensions.conf file? |
09:51.02 | Newbie___ | FXO uses fxs_ks or fks_ls signalling ? |
09:51.17 | drumkilla | you probably want fxs_ks |
09:51.18 | tessier | Depends on your fxo line. I suggest trying _ks |
09:51.20 | cypromis | btw, /w 25 |
09:51.22 | cypromis | sorry |
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09:51.53 | Newbie___ | ok, i am using _ks, but earlier TheEmperor said fxs_ls so was a bit confusing |
09:51.58 | RoyK | hmm |
09:52.06 | RoyK | poor cypromis |
09:52.14 | riksta | cypromis: you have to wait to make coffee? |
09:52.18 | riksta | sounds like a shit job :P |
09:52.28 | RoyK | anyone that've used sip with call waiting, three-way calling etc? |
09:52.30 | elric | according to app_dial.c the M() actually should execute if status is set to ANSWER, does this mean if Zap/3 is the outgoing line and the caller is using Zap/1 it regards the bridging of Zap/3 and Zap/1 as ANSWER? |
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10:12.06 | pbxjunkie | hmm... when asterisk first starts, I get (on the cli) Asterisk Ready. |
10:12.06 | pbxjunkie | *CLI> == Primary D-Channel on span 1 up |
10:13.42 | pbxjunkie | then when I first try to make a call I get : http://pastebin.ca/9592 |
10:14.08 | pbxjunkie | but ONLY the first time, every time after that, I get "unable to create channel zap" |
10:18.27 | jonathh | does anyone in here have any passion either for or against slackware? as a nice minimal.. distro for using asterisk with? |
10:19.06 | riksta | i used it when it was slackware 6, i bet it's changed since then.....back then the package management was terrible |
10:20.03 | jonathh | i dont use packages really... i mean i do with gentoo.. but i am a no thrills distro.. minimal so there is less to go wrong |
10:20.16 | olivier_ | <pbxjunkie> try to better understand pb : CLI>pri debug span 1 |
10:20.24 | riksta | gentoo shud be fine for you then |
10:20.34 | pbxjunkie | when I used channel 2 it worked |
10:20.44 | pbxjunkie | I only get this on channel 1 |
10:20.55 | pbxjunkie | the parameter after the first slash, is group or channel? Zap/x <--- |
10:21.08 | olivier_ | u can use both |
10:21.35 | olivier_ | but if u use group for a call, and after an channel number, it can be busy |
10:22.40 | pbxjunkie | oh so.. since I don't care which channel to use I should use groups.. i see |
10:23.18 | olivier_ | yep :) |
10:26.03 | RoyK | anyone here using a predictive dialler? |
10:27.47 | pbxjunkie | predictive dialer? what's that? |
10:29.08 | RoyK | used for annoying telemarketers |
10:29.10 | RoyK | or by |
10:30.33 | pbxjunkie | and what does it do exactly? |
10:30.55 | RoyK | http://www.voip-info.org/wiki-Predictive+dialer |
10:31.54 | pbxjunkie | evil |
10:32.02 | RoyK | :) |
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10:43.58 | RoyK | if used from agi, what is checkgroup supposed to return? |
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10:46.39 | elric | i am coding a predictive dialer as we speak |
10:46.47 | elric | or well trying to |
10:47.26 | RoyK | elric: nice. I can possibly help |
10:47.31 | RoyK | boss wants one |
10:48.43 | elric | RoyK, cool i am doing one in perl right now but first I want to sort answering machine detection out |
10:49.03 | elric | then do nifty shit like make a self learning GA predictive dilaer |
10:49.14 | RoyK | GA? |
10:49.35 | elric | genetic algorithms |
10:49.41 | RoyK | heh |
10:49.48 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
10:49.50 | elric | evolution |
10:49.54 | elric | :) |
10:49.55 | RoyK | sounds reasonable |
10:49.55 | RoyK | I know what it is... |
10:50.18 | elric | cool use postgres as the backend, i dont like mysql |
10:50.43 | RoyK | although it might be getter to just use a fuzzy function or even nn, or what do you think? |
10:50.59 | RoyK | s/nn/ANN/ |
10:51.07 | elric | yeah those are good options. |
10:51.25 | elric | but first need to get answering machine detection sorted |
10:51.44 | RoyK | I'd start fuzzy. that should be good enough for a start, and a lot simpler than the two others |
10:52.25 | RoyK | have you done some of the PD yet? |
10:52.48 | elric | nah just learning * yet |
10:53.00 | elric | well i am getting alright at it |
10:53.16 | elric | we are starting in May. |
10:56.32 | pbxjunkie | i find it incredible that people actually get paid to work towards making spamming easier |
10:56.40 | elric | lol |
10:56.51 | pbxjunkie | either via phone.. or the web ..building address-collecting searchbots |
10:56.51 | elric | err wrong window |
10:56.54 | pbxjunkie | :D |
10:56.54 | elric | :| |
10:58.10 | RoyK | pbxjunkie: boss tells me "we need a predictive dialer for our TM team" |
10:58.12 | RoyK | so I just do it |
10:58.25 | pbxjunkie | RoyK: fair enough |
10:59.18 | elric | pbxjunkie, its not only spamming, say our client is a company that needs to call people about pending bills/ overdue accounts |
11:01.09 | elric | its annoying but since its a job and it pays the bills |
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11:02.04 | elric | RoyK, may i message you? |
11:02.20 | RoyK | yep |
11:02.26 | elric | cool |
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11:18.59 | *** join/#asterisk FLeiXiuS (~Nick@pcp0011094024pcs.essex01.md.comcast.net) |
11:19.44 | FLeiXiuS | Is it possible to use asterisk without a provider? I just want to run this locally throughout my house then have a certain extention ring a specific IP phone. |
11:20.33 | pbxjunkie | FLeiXiuS: everything is possible |
11:21.07 | FLeiXiuS | pbxjunkie: :-P, would this feature already be built into Asterisk? |
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11:23.44 | foobos | FLeiXiuS, providers are optional. no need to connect it to the outside world at all |
11:24.07 | foobos | you can even run asterisk with two soundcards and have two extensions that way |
11:25.03 | FLeiXiuS | Sound Cards? I thought it would be using Ethenet ? |
11:25.15 | FLeiXiuS | Hence NIC. |
11:25.18 | pbxjunkie | FLeiXiuS: yes. ethernet. it's already "built in" |
11:25.59 | FLeiXiuS | hmm i'll have to keep doing my research thanks |
11:27.59 | pbxjunkie | FLeiXiuS: http://www.voip-info.org |
11:46.02 | jonathh | anyone know of good sip handset reseller in the UK |
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12:08.48 | Newbie___ | hi all |
12:09.11 | Newbie___ | exten => _95.,1,Dial(Zap/g5/${EXTEN:2},30) |
12:09.21 | Newbie___ | can anyone please tell me if the above is fine |
12:10.17 | RoyK | fine for what? |
12:10.18 | RoyK | :) |
12:10.29 | Newbie___ | RoyK: to make an outgoing call |
12:10.38 | RoyK | sure |
12:10.41 | RoyK | why not... |
12:10.50 | RoyK | only 30 secs might be a little low timeout |
12:10.53 | Newbie___ | i keep getting "Unable to create channel of type 'Zap'" |
12:11.02 | RoyK | pastebin full debug output |
12:11.06 | RoyK | ~pastebin |
12:11.53 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
12:11.53 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th || Read bug guidelines before posting bugs or face deletion. |
12:12.05 | Newbie___ | RoyK: http://pastebin.ca/9598 |
12:13.42 | Weezey | I have an SPA-3000 and the Line side is working great, but it's got a studdered dial tone. Aside from having a message, what does that mean? |
12:14.05 | Weezey | (there's no mailbox= for it) |
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12:15.54 | RoyK | Newbie___: what hardware do you have_ |
12:15.55 | RoyK | ? |
12:16.19 | Newbie___ | TE410P and just added X101P, X101P is suppose to be in g5 |
12:16.35 | RoyK | pastebin zapata.conf and zaptel.conf |
12:16.41 | Newbie___ | RoyK: ok |
12:16.42 | RoyK | hm |
12:16.43 | RoyK | really |
12:16.55 | RoyK | x101p? |
12:16.55 | elric | this is not working |
12:16.56 | elric | :( |
12:16.56 | RoyK | what is that? |
12:17.03 | RoyK | single pri card? |
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12:17.10 | Newbie___ | is a clone FXO |
12:17.13 | RoyK | ok |
12:17.14 | The_Duke | hi |
12:17.17 | RoyK | then I have no idea |
12:17.21 | RoyK | I don't do analog stuff |
12:17.36 | The_Duke | can someone help me connect a customer's cisco callmanager with my asterisk? |
12:17.41 | elric | does anyone know how to make macro execution wait till the call is actually answered? |
12:18.23 | Newbie___ | RoyK: http://pastebin.ca/9599 |
12:18.39 | Newbie___ | errr |
12:18.59 | RoyK | sorry. can't say anything about analog stuff |
12:19.19 | Newbie___ | RoyK: ok, tks |
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12:40.43 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
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12:44.17 | SuPrSluG | Newbie___:exten => _95.,1,Dial(Zap/g5/${EXTEN:2},30). Why do you have it on g5? if its an x100p clone a Zap/1 is normal. And do normal pattern. matching. _9NXXXXXX etc... |
12:45.34 | SuPrSluG | Newbie___:Do you have 5 x100p clones? |
12:46.33 | Newbie___ | i have 1 X101P and 1 TE410P |
12:46.59 | Pinhole | What hardware do I need if I want to use * as an answering machine on my normal phone line? |
12:47.45 | Newbie___ | span1-4 = g1-4. group5 = X101P |
12:48.16 | ManxPower | ~doc |
12:48.17 | jbot | it has been said that doc is The command is "~docs", moron! |
12:48.17 | bugbot | doc is assigned nothing and reported M3667, M2605. |
12:48.19 | ManxPower | ~mailinglist |
12:48.20 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
12:48.20 | bugbot | mailinglist is assigned nothing and reported nothing. |
12:48.26 | Newbie___ | Pinhole: i think u need FXO to answer call |
12:48.49 | ManxPower | Pinhole, Asterisk does NOT make a good answering machine |
12:49.01 | Pinhole | why not? |
12:50.01 | _Brian | good question...why not? Isnt a answering machine just a basic version of voicemail? |
12:50.13 | ManxPower | Pinhole, It was never designed as an answering machine. for example, imagine this: A call comes in, you don't pick it up in time and it goes to voicemail. There is no way to monitor the message that is being left, if you pick up a phone connected to the line asterisk will nto stop recording the message. |
12:50.27 | ManxPower | nto == not |
12:50.40 | Pinhole | My current answering machine has the same trouble. I have to unplug it. |
12:50.52 | _Brian | ManxPower: you can purchase answer machine service from RBOC's and you cant screen calls either... |
12:51.04 | _Brian | ManxPower: at least not with Verizon that is :) |
12:51.09 | ManxPower | Pinhole, you can buy an answering machine that works for like $20 |
12:51.27 | ManxPower | _Brian, No, you are purchasing voicemail service, not answering machine service. |
12:51.47 | Pinhole | I want to make my answering machine do "cool stuff". (read cool geeky toy) |
12:51.55 | _Brian | ManxPower: all depends on your expectations :).....for me..I MUST screen my calls at home..... ... :) |
12:52.14 | ManxPower | _Brian, then you need a real answering machine. |
12:52.16 | Pinhole | I don't screen calls, I usually answer on the first ring. |
12:52.40 | ManxPower | Pinhole, you also have to do some dialplan tricks to get asterisk not to pick up the line when the phone stops ringing |
12:52.43 | SuPrSluG | Newbie___:try dialling w/ just Zap/5 |
12:52.56 | Newbie___ | SuPrSluG: ok |
12:52.58 | _Brian | ManxPower: that is why I got one :) ....hell, i dont have the cash to setup * as a answering machine at home....besides my wife would kill me or I would kill her while trying to explain how to get messages |
12:53.45 | ManxPower | _Brian, Aparently wives are technical morons. Odd. I wonder if it's being female or getting married that keeps them from learning anything tech. |
12:54.04 | ManxPower | I suspect it's getting married. Anyone that gets married has to be pretty stupid anyway. |
12:54.15 | _Brian | ManxPower: ouch.... |
12:54.29 | Newbie___ | SuPrSluG: with zap/5, * pick up zap/5 from span1 |
12:54.51 | ManxPower | _Brian, research the origins of marriage some time. It was mostly about property rights. |
12:55.26 | ManxPower | Did she come with a good dowry? |
12:55.53 | ManxPower | Is she producing lots of kids to work as labor on your farm? |
12:55.55 | Pinhole | dual ownership == marriage. |
12:55.57 | _Brian | ManxPower: if dowry is defined as debt..yup :) |
12:56.14 | _Brian | ManxPower: she does tend to the sheep, and milk the cows....so yes |
12:56.29 | ManxPower | Anyway, you never hear "I want to use Asterisk, but it has to be easy enough for my husband to use." |
12:56.33 | _Brian | ManxPower: cant get her on the tractor though... |
12:56.46 | _Brian | ManxPower: rofl! |
12:58.09 | _Brian | does anyone know if * has any type of audio detection or call progress detection? I have an application that needs to Flashhook a call to put them on hold and then dial another extension utilizing SendDTMF. The problem I am having is that * will continue to the next step even before the remote party answers. If I utilize a Dial string, then i use another channel...... |
12:58.11 | ManxPower | It's not a popular view, but I think the whole gay marriage issue is silly. ALL marriage should be abolished from a govt stantpoint. Marriage should be personal/religious thing. The govt should be concerned with contract law and that's how a "marriage" should be set up. |
12:59.02 | ManxPower | Marriage should be replaced by a modified verison of the "LLP" or "LLC" that is common for businesses in the USA. 8-) |
12:59.32 | _Brian | OK..let me rephrase..."I want to use Asterisk, but it has to be easy enough for my kid to use." |
12:59.34 | _Brian | :) |
13:00.00 | _Brian | ManxPower: now i know you are gonna say something about children... :) |
13:00.33 | jakepdev | Brian - there's a 3rd party module I was looking at for CPD |
13:00.58 | bjohnson | ManxPower: I thought you were into the GPL version of marriage .. it's all shared baby |
13:01.00 | _Brian | jakepdev: cool......got any urls? i will check it out.. |
13:01.32 | SuPrSluG | Newbie___:did u paste your zapata and zaptel files? |
13:01.35 | ManxPower | Gads! Don't get me started about kids! |
13:01.52 | Newbie___ | SuPrSluG: http://pastebin.ca/9599 |
13:02.26 | ManxPower | bjohnson, Yes, but we don't call it that |
13:02.36 | _Brian | ManxPower: hell..we didnt mean to get you started about marriage..and look what happened |
13:02.58 | _Brian | ManxPower: :) |
13:03.07 | jakepdev | brian - http://www.voip-info.org/wiki-NVLineDetect |
13:03.39 | _Brian | jakepdev: thank you sir.. |
13:03.43 | jakepdev | np |
13:03.56 | jakepdev | if you try it, let me know if it works... |
13:04.33 | _Brian | yup..i will be looking at it shortly..got a meeting and then i will start testing.. |
13:04.49 | jakepdev | tnx - /msg me please |
13:05.06 | jefrey | it seems like when A calls B, A transfer B - C, in CDR, it shows 2 record, 1 is A to B , another is B - C (shouldn't it be A - C) ? |
13:07.19 | Delvar | jefrey: no, CDR is correct |
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13:09.58 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
13:10.28 | cervajs | hi, plz what is new method for playing mp3 (not mpg123)? |
13:10.30 | SuPrSluG | Newbie___:in zaptel fxsks=125. i'm not sure if that relates directly to the channel or # of cards.try w/ =1 |
13:10.44 | cervajs | i have someone who can code support for ogg vorbis |
13:10.52 | jefrey | Delvar: hmm.. is a flaw if the transfer is PSTN - PSTN |
13:10.55 | cervajs | but he doesnt know asterisk |
13:10.55 | Newbie___ | SuPrSluG: ok |
13:11.09 | jefrey | Delvar: how am I going to bill B if B is a not a user?! |
13:11.10 | cervajs | for MoH |
13:11.48 | Newbie___ | SuPrSluG: i did that before i use 125, X101P crashed with Zap1 in span 1 |
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13:12.10 | Delvar | jefrey: thats how it works in normal PSTN calls, if you get a PSTN call into your houm, then transfer that call out, YOU get chareged for teh transfer not the caller, thats how it works.... |
13:12.31 | Newbie___ | guys from mailing list adviced that even i only use 2 span, span 3 and 4 must be reserved |
13:13.03 | jefrey | Delvar: hmm.. you sure bout that? how can a receiver be charged? |
13:13.18 | Delvar | jefrey: they cant! |
13:13.21 | Newbie___ | damn, is there a way to tell my wife to get off my back when i am working |
13:13.22 | jefrey | Delvar: B receives a Call from A, he was transfered to C and he to be charge? |
13:14.43 | Delvar | jefrey: A calls B, B talks to A, B tansfers A to C, C talks to A. in this, A is charged for call to B, B is charged for call to C. |
13:15.06 | Delvar | jefrey: thats exactly how it should work |
13:15.25 | jefrey | hmm |
13:15.38 | jefrey | that's assuming if B is one of sip user |
13:15.40 | Delvar | pm -> |
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13:16.42 | newl | however, in his original example, A is transferring B to C which implies a second call from A to C (conferenceing all three, A drops out, B and C remain). |
13:17.08 | newl | A party placed two calls. |
13:17.16 | Pinhole | ManxPower, if I was to go against your advice and try it anyway, what would be the minimum (cheapest) hardware I would need to pull off an answering machine? |
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13:21.48 | jonathh | newbie.. but i'd say pc.. asterisk... fxo card. |
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13:22.13 | jonathh | i think it is an fxo.. the one that geos from the pc to the phone line |
13:22.52 | Newbie___ | jonathh: yes, is a FXO |
13:23.24 | jonathh | thanks |
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13:24.08 | Pinhole | It looks like FXO cards fit inside another card. Can you just by a FXO card? |
13:24.10 | Newbie___ | jonathh: why do u thank me for ? |
13:24.21 | jonathh | clarifying :) |
13:24.29 | Newbie___ | ohm ok |
13:24.36 | jonathh | i think the x100p is a clone card |
13:24.40 | jonathh | that does only that |
13:24.40 | Newbie___ | yes it is |
13:24.48 | jonathh | so for 1 line.. at home.. tis fine |
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13:25.00 | ManxPower | Pinhole, Digium sells the TDM400P with FXO module. You can still get cheap clone X100P cards, but they are no longer being made, so you won't be able to get them at some point in the future. |
13:25.36 | cervajs | do you someone using MoH+mp3? |
13:25.56 | Newbie___ | take ManxPower advice , go for digium TDM |
13:26.30 | ManxPower | "Microsoft Releases Public Beta of Data Protection" It's not April 1, is it? |
13:26.45 | jonathh | lol |
13:29.10 | darkskiez | I saw an x100p clone for sale for £9.99 |
13:29.39 | darkskiez | might as well buy a TDM400 card with an FXO and FXS module. |
13:29.40 | Gand_DJ | yeah, they are standard dialup modem cards, with MD3200 chipset |
13:29.53 | Gand_DJ | wouldn't mind getting couple |
13:30.07 | JerJer | Caveat Emptor |
13:30.12 | darkskiez | telappliant seem to be only have FXS modules for the TDM tho |
13:30.32 | darkskiez | i hear the x100p cards are echotastic |
13:30.37 | JerJer | most X100P clones have horrible echo problems and some cannot deal with caller*id (on-hook audio) |
13:30.58 | jonathh | is that why the clones are sooo much cheaper |
13:31.10 | *** join/#asterisk iq (~iq@207-224-100-229.omah.qwest.net) |
13:31.11 | jonathh | i mean sub £10 compared to ~£90 for a TDM and the correct module |
13:31.27 | Newbie___ | anyone from www.iax.cc |
13:31.57 | JerJer | jonathh: you get what you pay for |
13:31.58 | darkskiez | http://www.myphonecall.co.uk/voip/telephonycards/oem/default.aspx |
13:32.03 | darkskiez | theyve got em for a tenner |
13:32.28 | jonathh | see you really cant complain about a tenner |
13:32.55 | *** join/#asterisk BBRodriguez (~alex@pD956341D.dip.t-dialin.net) |
13:32.58 | darkskiez | jonathh, can i borrow a tenner off you at the next slug meet? |
13:33.19 | BBRodriguez | Can anybody confirm www.voip-info.org is down ? |
13:33.33 | darkskiez | Noooooooooo |
13:33.39 | jonathh | you sure can |
13:33.57 | darkskiez | Connected...Waiting for..... |
13:33.58 | jonathh | hmmm dont seem to be work |
13:34.04 | darkskiez | works for me |
13:34.14 | jonathh | it will be down with all the peeps in here checking!" |
13:34.44 | darkskiez | jonathh, I thought this was #scotlug, ooops. nevermind :) |
13:34.50 | Newbie___ | is slow |
13:35.02 | jonathh | lol |
13:35.09 | jonathh | no skint flints here! |
13:35.20 | jonathh | yeah confirm... it worked finally! |
13:35.41 | Newbie___ | what time is it now in US mountain standard time or something |
13:36.03 | jonathh | fork knows |
13:36.49 | newl | 9:36AM EST so take two hours from that for mountain. |
13:36.52 | ManxPower | 7:30am MDT |
13:36.54 | jonathh | how many UK peeps we got in here then? |
13:37.06 | Newbie___ | ok, thank you all |
13:37.52 | darkskiez | maximum length of 1000base-t on cat5e anyone? google tells me different things. |
13:38.15 | ManxPower | darkskiez, search Cisco's web site. |
13:38.22 | jonathh | not far enough? |
13:38.45 | Gand_DJ | I think 100 meters? |
13:39.01 | ManxPower | Most *Base-T is 100 meters |
13:39.02 | Newbie___ | www.voip-info.org site finally finish loading |
13:39.39 | jonathh | someone posted a voip-info article on slashdot or something? |
13:40.00 | darkskiez | apparently its 10meters for stranded |
13:42.16 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
13:43.40 | kajtzu | 1000base-t will run on cat5 or cat5e upto 100 meters. |
13:44.39 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
13:44.53 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
13:46.10 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
13:47.24 | darkskiez | kajtzu, source? |
13:47.36 | kajtzu | darkskiez: cisco |
13:47.40 | dmccollum | Hello peeps |
13:48.47 | tzanger | ugh fucking dialplan bugs |
13:51.28 | *** join/#asterisk CoolCat_ (~god@200.170.109.217) |
13:51.38 | CoolCat_ | morning |
13:52.14 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
13:52.41 | Gand_DJ | ne1 have fwd setup in *@home/ |
13:52.53 | Gand_DJ | I can dial out, but when I try to call my box, I get busy signal. |
13:53.08 | *** join/#asterisk moy (~kvirc@201.135.105.124) |
13:53.14 | Gand_DJ | using a different softphone not linked to * that is |
13:53.32 | Egonis | When I try the sample (ext1000) it works, however, pressing '2' or '#' does nothing... I tried setting dtmfmodes, but nothing changes... any ideas? |
13:53.34 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
13:53.49 | dmccollum | Is it common to have static when using x100p cards? If so is there a setting that would help eliminate the static? |
13:56.31 | foobos | dmccollum, that is most likely due to badly insulated motherboard. try changing the PCI-slot |
13:56.55 | foobos | and if you have soundcard in the same server/computer try removing it |
13:57.04 | foobos | also changing the PSU might help |
13:57.15 | jonathh | you oculd try changing the whole computer |
13:57.19 | jonathh | and if that doesn't work |
13:57.21 | CoolCat_ | does anyone has any idea how * gateway skype (i dont really used * yet, only installed *@home, but didnt got any clue how conect skype as * doesnt manage the soundcard) |
13:57.23 | jonathh | a differenct card ;) |
13:57.44 | *** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz) |
13:57.44 | foobos | there's no skype connectivity as of yet |
13:58.17 | carlosh | hello, anyone from voipjet support here ?? Thanks. |
13:58.18 | CoolCat_ | foobos =o/ i thought i heard something like it was feasable! =o( |
13:58.38 | CoolCat_ | calos o que voipjet? |
13:58.42 | foobos | coolcat_, all on design phase at the moment i think |
13:59.18 | carlosh | CoolCat: trying to get hold of them.. someone from voipjet.. |
13:59.21 | CoolCat_ | foobos =o) probably it will use the soundcard or the skype api, thought! |
14:00.40 | CoolCat_ | foobos see if my way of think is right... |
14:00.49 | CoolCat_ | i can open a FWD account |
14:00.58 | foobos | coolcat_, still a long way to working solution |
14:01.04 | carlosh | can someone recommend better providers than voipjet ? I'm outside USA, and want to terminate calls in Southamerica, need cheap and good.. |
14:01.08 | bjohnson | and limited interest |
14:01.10 | dmccollum | foobos: Thanks I'll give it a try. There's no soundcard in the server. Just an Intel dual 10/100 card and two x100p's. |
14:01.15 | *** join/#asterisk Nivex (kjotte@user-0c8hq5r.cable.mindspring.com) |
14:01.17 | CoolCat_ | and setup asterisk to redirect to my pbx, correct? |
14:01.39 | bjohnson | carlosh: teliax, nufone, livevoip, etc, etc are all possibilities. |
14:01.40 | CoolCat_ | carlos to pstn? |
14:02.40 | carlosh | CoolCat: yes to pstn in Southamerican countries.. |
14:02.50 | carlosh | iax2, ilbc |
14:02.58 | *** join/#asterisk jf_ (~jeanfranc@modemcable077.187-80-70.mc.videotron.ca) |
14:03.33 | CoolCat_ | carlosh probably try some provider from the country, thought! |
14:03.48 | CoolCat_ | bjohnson fwd is good? |
14:03.50 | jf_ | can someone tell me that xlite does not transfert anything (audio, ringtone) but iax does, why ? |
14:04.13 | jonathh | port forwarding issue? |
14:04.23 | jf_ | jonathh u think so |
14:04.26 | bjohnson | CoolCat_: for what? |
14:04.43 | jonathh | i dont knw... but i know that sip is needy ont he ports.. iax isn't so much so |
14:04.46 | CoolCat_ | jonathh im in this issue atm! |
14:04.53 | jf_ | k |
14:04.58 | jonathh | ok.. |
14:05.04 | bjohnson | CoolCat_: it's free and does certain .. it is even pretty reliable considering the price |
14:05.08 | CoolCat_ | bjohnson for sip-sip conections (i signed it, as it was for free) =o) |
14:06.11 | bjohnson | jf_: afaik, xlite is sip and requires extra screwing around to get it to work through NAT firewalls |
14:06.29 | jonathh | the conclusion i have come to is dont bother :) |
14:06.40 | bjohnson | CoolCat_: sip-sip connections you can do directly and don't need an intermediary like FWD |
14:06.47 | jonathh | use sip for intra netwrok.. and iax connectiosn to other asterisk boxes for internet connections |
14:06.53 | bjohnson | jonathh: (unless you have to) |
14:07.01 | jonathh | true. |
14:07.27 | jf_ | bjohnson: it was working yesterday night, im quite sure |
14:07.30 | CoolCat_ | well, i would need a sip server dont i? |
14:07.54 | bjohnson | CoolCat_: asterisk talks sip |
14:08.39 | bjohnson | CoolCat_: but even without *, a SIP device can be configured to allow you to directly call another sip device |
14:08.44 | CoolCat_ | bjohnson well, i am still learning...i didnt configured one byte under asterisk |
14:09.26 | CoolCat_ | bjohnson i would like to have a sip number...so ppl can dial me, and i want to redirect it to my regular pbx! |
14:09.39 | bannerman | *yawN* |
14:09.40 | CoolCat_ | bjohnson i also studing the zoomtel v3 |
14:09.40 | bannerman | mornin |
14:09.41 | bjohnson | consider FWD as kind of like a registry service and free voip voicemail |
14:09.57 | bjohnson | CoolCat_: no idea what a zoomtel is |
14:10.19 | CoolCat_ | bjohnson fwd is need for me to get the sip #, isnt it? |
14:11.08 | Egonis | When I try the sample (ext1000) it works, however, pressing '2' or '#' does nothing... I tried setting dtmfmodes, but nothing changes... i'm really lost |
14:11.29 | Egonis | I also tried ext8500 for voicemail, but entering the voicemail box number does nothing either |
14:11.50 | jonathh | egonis.. i got this with sipgate i think |
14:11.54 | jonathh | they blamed BT |
14:12.37 | Egonis | jonathh: I tried dtmfmode=inband just now.. let's see what happens |
14:13.04 | CoolCat_ | bjohnson http://www.zoom.com/products/voip_products.html v3! |
14:13.15 | jonathh | i tried all sorts of combinations in the [general] and for the sip setting of the connecting device.. also nat= effects it.. |
14:14.10 | CoolCat_ | bjohnson but i dont know exactly how it will integrate with asterisk, and if it is need, thought! |
14:14.15 | Egonis | jonathh: worked |
14:14.25 | Egonis | jonathh: what is the default passwd for mailbox 1234? |
14:14.26 | jonathh | so where did you set it? |
14:14.33 | jonathh | 4242 |
14:14.34 | Egonis | jonathh: sip.conf |
14:14.34 | jonathh | i think |
14:14.38 | Egonis | jonathh: ty! |
14:14.41 | jonathh | yeah but in general? |
14:14.45 | jonathh | or for the sip device? |
14:15.12 | Egonis | jonathh: the sip device itself |
14:15.22 | Egonis | jonathh: how do I find the mailbox passwd? 4242 was wrong |
14:15.29 | jonathh | hmm might stop mucking about with exim for a sec.. and tr |
14:15.29 | jonathh | y |
14:15.33 | jonathh | check voicmail.conf |
14:15.37 | jonathh | for that extension |
14:15.43 | jonathh | it is the first CSV |
14:16.31 | jonathh | egonis |
14:16.34 | *** join/#asterisk Moc[NX] (~mochouina@64.235.196.24) |
14:16.35 | jonathh | so you are using inband? |
14:16.38 | jonathh | and it works? |
14:16.43 | jonathh | what provider are you using? |
14:16.45 | Egonis | jonathh: works great |
14:16.49 | Egonis | jonathh: Primus |
14:16.57 | jonathh | hmm lets see what they offer |
14:17.16 | jonathh | primus.com? |
14:17.25 | Egonis | jonathh: funny, although the passwd is set to 4242, it says login incorrect |
14:17.27 | Egonis | jonathh: primus.ca |
14:17.39 | Egonis | jonathh: I'm switching to allstream tho |
14:17.57 | jonathh | well i use sipgate.. |
14:18.02 | jonathh | it gives you any uk area number |
14:18.07 | jonathh | but seems the tones dont work |
14:19.24 | *** join/#asterisk cj-rm (~cjrm@81-178-22-214.dsl.pipex.com) |
14:19.26 | cj-rm | hey ppl |
14:19.40 | cj-rm | How're things on #asterisk today? |
14:19.48 | jonathh | hmm no tones |
14:19.52 | jonathh | cant get into voicemail |
14:19.53 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
14:19.55 | Egonis | jonathh: That's odd... I wonder why... have you tried dtmfmode=info? |
14:20.01 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
14:20.02 | jonathh | bloody sipgate! |
14:20.08 | jonathh | yeah.. but i'll try it again |
14:20.24 | Egonis | jonathh: modes (from what I know) are inband, info, rfc2833 |
14:20.32 | fenlander | jonathh: sipgate does not work in incoming DTMF |
14:20.34 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
14:20.51 | _THEEND_ | someone uses spandsp for fax? |
14:21.03 | jonathh | so how can you operate menu's etc? via sipgate.. |
14:21.05 | jonathh | or cant oyu? |
14:21.11 | *** join/#asterisk Nuttah (~andrew@amber2.interdart.co.uk) |
14:21.21 | fenlander | jonathh: no, at least not last time I looked into it. They are aware of the problem. |
14:21.23 | Nuttah | afternoon guys :) |
14:21.30 | jonathh | any alternatives? |
14:22.00 | fenlander | jonathh: there are several other ITSPS in the UK that will give you a local number, but none that I know of are free |
14:22.17 | jonathh | yeah that is what sipgate is good for :-/ |
14:22.31 | jonathh | i dont mind shelling out once i have proved the tech.. but not while i am still playing |
14:22.46 | Nuttah | does anyone here have experience with DECT units connected to sipura adapters? and have ever experience random tone sounds being generated mid call |
14:23.44 | Nuttah | aye, geographical numbers are never free from what ive found as well/ |
14:24.08 | jonathh | sipgate is.. |
14:24.19 | jonathh | maybe i should quit moaning ;) |
14:24.46 | Nuttah | sipgate is also about twice as expensive as my current call rate for national calls |
14:25.08 | jonathh | what do you uise? |
14:25.08 | Nuttah | if i recall correctly that is |
14:25.23 | Nuttah | atm i'm testing out with magrathea |
14:25.25 | fenlander | For outgoing you can't really beat call18899 :-) |
14:25.41 | Nuttah | linky fen? |
14:26.10 | fenlander | www.call1899.co.uk/coip.php |
14:26.13 | _THEEND_ | someone uses spandsp for fax? |
14:26.13 | Nuttah | ta |
14:26.20 | fenlander | oops voip.php |
14:26.26 | Egonis | I just added a voicemail box to voicemail.conf -- what else do I need to do to get it to answer w/ voicemail? |
14:26.27 | jonathh | it no working :) |
14:26.43 | jonathh | lol |
14:28.00 | Nuttah | interesting.. they charge for accounts fen? |
14:28.15 | fenlander | no - just for calls iirc |
14:28.21 | Nuttah | Egonis: you will need a voicemail line in extensions |
14:28.24 | fenlander | outgoing only |
14:28.35 | Blissex | fenlander: jonathh: in the UK Gradwell.com is free for the 1st 3 months, then £4/month, and they give local numbers almost anywhere. |
14:29.04 | Nuttah | hmm I wonder how they can do that |
14:29.16 | cj-rm | how do I get asterisk itself to establish a call between two of its extensions? |
14:29.23 | cj-rm | The extensions are SIP phones |
14:29.40 | jonathh | http://www.gradwell.com/ |
14:29.41 | jonathh | ? |
14:29.41 | Egonis | Nuttah: ty! |
14:29.43 | Nuttah | notranfer=yes.. or is that iax.. |
14:30.20 | Blissex | http://VoIP.Gradwell.com/ too. |
14:30.28 | cj-rm | Is there not some kinda spool that I can dump a file into to establish a call?? Or is there a pipe or socket I can connect to?? |
14:30.48 | Blissex | Gradwell resell Magrathea more or less, and they are fairly reasonable. They support IAX too, not just SIP. |
14:31.08 | Blissex | cj-rm: are your questions serious? |
14:31.37 | Nuttah | cj: adding a canreinvite=no will make asterisk manage transfers i believe |
14:31.44 | Nuttah | cj: into sip.conf |
14:31.58 | cj-rm | Blissex: sure are :) Are they n00besque? |
14:32.12 | Egonis | Nuttah: I added voicemail for ext611, worked.. but now 8500 doesn't work |
14:32.15 | Blissex | cj-rm: well, beyond your wildest dreams :-) |
14:32.35 | Blissex | cj-rm: all that Asterisk does is to switch calls between its extensions :-) |
14:32.53 | Nuttah | egonis: usepastebin and show us you conf |
14:33.11 | Nuttah | bliss: pedantic mofo :P |
14:33.31 | cj-rm | Blissex: Yeah I know :) But I want a piece of software to tell Asterisk to call BOTH the extensions! |
14:33.35 | Blissex | cj-rm: and as to «pipe» or «socket» it uses usually VoIP, between IP phone. |
14:33.48 | Blissex | cj-rm: ahhh, thats completely different. |
14:33.59 | Blissex | cj-rm: describe then what you really want to happen. |
14:34.03 | Nuttah | seems we have a few UK guys here.. anyone use sipura and dect together? |
14:34.06 | cj-rm | Blissex: I know, its completely different, but thats what I want :) |
14:34.17 | fenlander | Nuttah: yes - never had a problem |
14:34.24 | durex | Asterisks... I just instaled Asterisk on a FBSD Box. All my desktops are with x-lite installed, and now I wanna that them talk with each other. Where do I start? |
14:34.26 | Blissex | cj-rm: asking intelligible questions is a somewhat important question. |
14:34.44 | Blissex | durex: you need to do two things... ooops. three |
14:34.49 | Nuttah | hmm never get tone beeps in the call Fen? |
14:34.51 | Egonis | Nuttah: nm, figured it out |
14:34.53 | durex | Blissex what? |
14:35.02 | Blissex | durex: the FIRST thing is to read an introduction to Asterisk :-0 |
14:35.14 | fenlander | Nuttah: no - but I have heard other people mentioning that problem |
14:35.20 | Nuttah | !docs |
14:35.20 | durex | Blissex I did it... :-) |
14:35.28 | Nuttah | man whats the command again |
14:35.29 | cj-rm | Blissex: I want a program (I'm writing) to decide that Bob (Extension: 2000) and Sue: (Extension: 2001) need to talk. I want my program to then tell Asterisk to ring bob and sue's phones, so that they can chat to each other |
14:35.37 | Blissex | durex: the other two are to define your phones as entries in 'sip.conf' or similar, and then define them as related extensions in 'extensions.conf'. |
14:35.56 | Blissex | cj-rm: that's a rather odd requirement indeed. |
14:36.02 | cj-rm | Blissex: Both my extensions are SIP phones |
14:36.14 | jonathh | im curious to know why you want auto initiated calls? |
14:36.16 | Blissex | cj-rm: I suspect that the best way would be a special purpose extension/plugin. |
14:36.41 | durex | Blissex and how do I do it? |
14:36.43 | durex | :P |
14:36.46 | Blissex | cj-rm: however you might be able to hack something together as a meeting room hack. |
14:36.47 | cj-rm | jonathh: Think scheduled calls :) |
14:36.53 | Nuttah | fenlander: out of curiousity.. what dect phones are you using? |
14:37.02 | jonathh | i like the idea.. |
14:37.03 | Nuttah | personally think its a dect problem |
14:37.06 | cj-rm | Blissex: meeting room hack? |
14:37.15 | jonathh | but maybe.. an email saying... 'hey dont forget to call thingy wahtsit' |
14:37.22 | Blissex | cj-rm: well, you create a meeting room and they connect via the meeting room. |
14:37.24 | jonathh | then maybe 'click here to initiate the call' |
14:37.33 | fenlander | Nuttah: BT Synergy range |
14:37.45 | Nuttah | right using the BT ddiverse range myself |
14:37.53 | Blissex | cj-rm: but the idea is that in general Asterisk is sort of passive, and switches calls only on request. |
14:38.00 | Nuttah | cause i'm a cheapskate :) |
14:38.21 | fenlander | Nuttah: are they dtmf tones that you hear? |
14:38.27 | jonathh | but astersik COULD call party 1.. and on party 1 answering.. call party 2? |
14:38.35 | Blissex | cj-rm: however, it could be something like call extension A, put it on hold, and transfer it to B. |
14:38.47 | cj-rm | jonathh: Thats exactly what I want :) |
14:38.54 | Nuttah | Fenlander: yes seemingly at random as well |
14:39.11 | cj-rm | Blissex: I guess thats exactly what I want |
14:39.13 | Blissex | cj-rm: but I doubt there is any logic that does that already. In part because it is hard to tell phone A to call B. |
14:39.33 | fenlander | cj-rm: use a call file? |
14:39.44 | cj-rm | fenlander: a call file? |
14:39.47 | jonathh | i can see some wisedom to this |
14:40.16 | cj-rm | jonathh: me too :) Now how do I get Asterisk to do it? |
14:40.18 | fenlander | cj-rm: http://www.voip-info.org/wiki-Asterisk+auto-dial+out |
14:40.33 | jonathh | dunno.. me new to all this :_) |
14:40.39 | jonathh | but if asterisk can initiate calls |
14:40.55 | jonathh | i see address books on the desktop.. adn double clicking on a number.. calling that perosn |
14:41.05 | *** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34) |
14:41.13 | *** join/#asterisk CoolCat_ (~god@200.170.109.217) |
14:41.20 | fenlander | Nuttah: which dtmf mode are you using? |
14:41.45 | cj-rm | jonathh: yup, its a cool idea, heh? :) click-to-dial its a big thing, thats under-understood :) |
14:42.07 | cj-rm | there are even sip specs defining best practices for it |
14:42.18 | jonathh | i was talking to a mate about this last night .. and i decided it w asn't possible |
14:42.21 | jonathh | but i take that back |
14:42.27 | Nuttah | fenlander: RFC2833 |
14:42.29 | jonathh | where? |
14:43.32 | fenlander | Nuttah: I use inband to the sipura - maybe there is a bug in the rfc2833 code? |
14:44.13 | Nuttah | possibly.. i'll test that out. |
14:44.20 | Gand_DJ | Does this info look like something you'd put into the incoming section of * or outgoing? |
14:44.24 | jonathh | any idea where sample.call is? |
14:44.41 | Gand_DJ | context=inbound |
14:44.44 | Gand_DJ | dtmfmode=inband |
14:44.50 | Gand_DJ | type=friend |
14:44.56 | jonathh | incoming |
14:45.33 | Gand_DJ | ok. I just signed up with voipforcanada and they sent me that info (and couple more lines) and told me to add it to my sip.conf for * to work for making outgoing calls |
14:45.39 | cj-rm | RFC 3725 - best practices for 3pcc |
14:45.44 | fenlander | jonathh: asterisk/sample.call |
14:46.01 | jonathh | hmm i cant see it! |
14:46.09 | Blissex | fenlander: just checking... you do remember that only the 711 codecs can do inband DTMF, and that both RFC2833 and INFO DTMF can be broken in various phones... |
14:46.40 | Blissex | jonathh: there are several sample calls files in that Wiki page... |
14:46.44 | fenlander | Blissex: yes, I use 711 everywhere at the moment |
14:46.56 | Nuttah | hmmm prefer GSM over my main pipe |
14:47.18 | fenlander | Blissex: something chaged between 1.0.5 and 1.0.6 that changes the negotiation of dtmf mode. |
14:47.45 | Nuttah | which reminds me... must upgrade *.. on 1.0.4 atm :P |
14:47.54 | *** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br) |
14:48.09 | cj-rm | Blissex, fenlander, jonathh - Thanks for your help... I'm gonna go check this stuff out :) |
14:48.35 | jonathh | hey |
14:48.38 | jonathh | cj-rm |
14:48.39 | jonathh | dude |
14:48.52 | jonathh | if you create that file.. in the outgoing spool file |
14:48.55 | Gand_DJ | Hrm, for inbound, should I have type= friend or user? |
14:48.57 | jonathh | it will prob do what your adter |
14:49.01 | jonathh | after |
14:49.08 | jonathh | need to do some tests first :) |
14:49.28 | Nuttah | fenlander: regarding 1899, seems voip service is only available for existing customers. |
14:49.51 | fenlander | Nuttah: yes, but you can become an existing customer by signing up :-) I did |
14:50.01 | Nuttah | Gand_DJ: user = inbound, peer = outbound, friend = both |
14:50.13 | vaewyn | no friend = evil |
14:50.16 | vaewyn | :} |
14:50.18 | cj-rm | jonathh: I know I've been looking into it... Best of luck man. |
14:50.19 | Nuttah | :P |
14:50.30 | cj-rm | ttfn |
14:50.48 | jonathh | l8rz |
14:54.03 | Pinhole | I just had a really great idea: You could use voice recognition to play the telephone game. You say something and * interprets it, says it to another * box and when it gets back to you, you'll be amazed at what comes back! |
14:56.02 | Gand_DJ | What is usually better.. making an outbound [] entry using peer, and then making an inbound [] entry using user.. making 1 [] entry using friend... or making an outbound & inbound [] entry using friend in both? |
14:56.30 | Nuttah | well thats your call tbh |
14:56.37 | Nuttah | there are risks using friend |
14:57.14 | Nuttah | if you have definite directions for your connections. then always best to use the user/peer approach |
14:57.27 | *** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca) |
14:57.45 | *** join/#asterisk psycodad (~obiwan@2001:4060:4419:b1:0:0:0:2) |
14:58.03 | jf_ | why when i dial an sip channel i got auto-congestion everyone is busy/congested at this time |
14:58.20 | Gand_DJ | I'm trying to configure asterisk@home for outbound & inbound. fwd works for outbound, along with simpletel... but fwd inbound gives me busy signal when I call it. |
14:58.40 | Gand_DJ | using context=from-pstn |
14:58.46 | Gand_DJ | for inbound [] |
14:58.57 | jf_ | unable to create channel type sip |
15:00.41 | Nuttah | i'm probably not going to look at this gand_dj.. currently getting harrassed by customers, but use pastebin if you want to give ppl access to you current conf |
15:01.10 | *** join/#asterisk netofsickcoder (~netofsick@202.154.225.74) |
15:01.59 | *** join/#asterisk heison (~heison@p85.n-lapop06.stsn.com) |
15:02.01 | jf_ | why when i dial an sip channel i got auto-congestion everyone is busy/congested at this time |
15:02.16 | heison | is CVS head currently broken? |
15:02.45 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
15:03.19 | heison | i get "ast_destroy is deprecated, use ast_config_destroy instead!" and a core dump |
15:05.05 | CoaxD | heison: Hah. Someone'll find it in 5 minutes. Dont worry. |
15:05.08 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
15:05.11 | Gand_DJ | http://pastebin.ca/9608 |
15:05.32 | Gand_DJ | shows extensions.conf and extensions_additional.conf |
15:05.36 | heison | it has been like that since yesterday :( and i wasn't sure if it was me... |
15:05.49 | CoaxD | heison: Hmmm. Wonder if it could be in a module |
15:05.49 | jonathh | hey peeps.. i am playinh with dial-outs.. can i get it to route the outgoing call over iax? |
15:06.04 | Gand_DJ | if your provider supports iax |
15:06.11 | CoaxD | heison: did you rm -rf /usr/lib/asterisk/modules before you did a 'make install'? |
15:06.12 | jf_ | why when i dial an sip channel i got auto-congestion everyone is busy/congested at this time |
15:06.19 | jonathh | well it would go over the intenet to my mates ia box :) |
15:06.44 | heison | coaxD: haven't try that yet... hang on |
15:07.01 | Gand_DJ | I think as long as you setup the outbound properly, and they setup inbound properly for iax.. I think it would work |
15:07.04 | Gand_DJ | I'm kinda new at this |
15:07.05 | jonathh | prlbem is.. i want it to start the first call.. on a local sip.. then connect it to another sip over iax.. |
15:07.29 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
15:07.29 | *** mode/#asterisk [+o bkw_] by ChanServ |
15:07.57 | CoaxD | heison: The thing is, when you install new versions of asterisk, if the particular module code doesnt get updated, there's a chance that an old module wont get overwritten. (Tho I think that nowadays, the makefile circumvents that from happening, so thats probably NOT the issue) |
15:07.59 | Gand_DJ | well I think IAX from softphone -> asterisk works.. then * -> * can be over iax... then * -> softphone can be sip again |
15:08.19 | jonathh | how do i specify that toh in the call thingy? |
15:08.29 | CoaxD | heison: Moreover, with a change like that, module code *would* get updated |
15:08.51 | Gand_DJ | not sure. kinda new to using *@home |
15:09.43 | heison | coaxD: that worked... but i had to move the codec_g*.so by hand |
15:09.51 | CoaxD | heison: AHHHH!!! :) |
15:10.01 | CoaxD | heison: WHO'S YOUR DADDY?!#$ |
15:10.01 | heison | lucky i didn't rm -f... phew |
15:10.19 | jf_ | why when i dial an sip channel i got auto-congestion everyone is busy/congested at this time |
15:10.23 | CoaxD | heison: Indeed! Especially if you had custom stuff in there! |
15:10.40 | heison | hmm... my DADDY? bkw_? kram? |
15:10.40 | moy | does anybody has any idea why asterisk is stoping when i use musiconhold? i start asterisk with "asterisk -vvvvvvvvvvvvvc" and the end message before stoping is: |
15:10.44 | moy | Found new ID3 Header |
15:10.44 | moy | Beginning asterisk shutdown.... |
15:10.44 | moy | Executing last minute cleanups |
15:10.44 | moy | <PROTECTED> |
15:10.44 | moy | Asterisk cleanly ending (2). |
15:10.55 | CoaxD | heison: Yeah, those two qualify to be your daddy |
15:11.01 | CoaxD | heison: Maybe i qualify to be a cousin or something. |
15:11.11 | heison | lol |
15:11.13 | CoaxD | heison: (thrice removed or something.) |
15:11.24 | heison | thanks for your help... |
15:11.29 | CoaxD | heison: Welcome! |
15:11.48 | CoaxD | hate. api. changes. |
15:12.53 | nestAr | anyone using monitor with queues? I'm using the monitor-join command |
15:13.05 | nestAr | and it leaves me with 364 byte wav files |
15:15.17 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
15:16.09 | moy | what does monitor does? i dont know that app |
15:16.19 | *** join/#asterisk netofsickcoder (~netofsick@202.154.225.74) |
15:16.21 | darkskiez | record the call |
15:17.05 | Egonis | In messages: Unable to open pseudo channel for timing ... Sound may be choppy (this is to name a few) |
15:17.31 | CoaxD | Egonis: insmod zaptel.o ; insmod ztdummy.o |
15:18.10 | Egonis | CoaxD: Non-existent |
15:18.16 | Egonis | CoaxD: Do I need to install zaptel? |
15:18.37 | CoaxD | Egonis: Um, yes |
15:18.42 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
15:18.44 | Egonis | CoaxD: Actually, Zaptel is already installed |
15:18.45 | jf_ | can someone help me about that : why when i dial an sip channel i got auto-congestion everyone is busy/congested at this time |
15:18.51 | Egonis | CoaxD: I'm using kernel 2.6 |
15:19.13 | DrWho17 | jf_: no extension match? |
15:19.14 | CoaxD | egonis: You need to install zaptel else any timing functions wont work, and you wont be able to create meetme rooms, or such |
15:19.31 | jf_ | it should. |
15:19.54 | DrWho17 | well, watch the call come in from the console, this should tell you what is happening |
15:20.06 | jf_ | exten => 103,1,Dial(sip/JF,20) |
15:20.08 | Egonis | CoaxD: adding those mods to my kernel config now |
15:20.13 | jf_ | it should be o |
15:20.27 | jf_ | i can see what is happening |
15:20.36 | Gand_DJ | ne1 know which countries require G729 to be licensed? |
15:20.47 | CoaxD | Egonis: If you can use 'zaprtc.o', use that instead of 'ztdummy.o'. More accurate timing source |
15:21.08 | CoaxD | Gand: Asterisk requires G729 to be licensed to enable it |
15:22.08 | Egonis | CoaxD: I modprobed zaptel and ztdummy, but still get the same error message about pseudo channel |
15:22.18 | CoaxD | egonis: does /dev/zap exist? |
15:22.27 | Egonis | CoaxD: let's see.. one sec |
15:22.35 | Egonis | CoaxD: nope.. :) |
15:22.43 | CoaxD | egonis: You never actually ran 'make install' |
15:22.48 | CoaxD | egonis: Or at least, not with that kernel |
15:22.55 | Egonis | CoaxD: I emerged it in gentoo |
15:22.57 | CoaxD | ('make install' on the zaptel sources) |
15:23.05 | CoaxD | Egonis: Will you frickin gentoo users quit doing that? :) |
15:23.15 | Egonis | CoaxD: LOL! |
15:23.23 | Egonis | CoaxD: re-emerging zaptel for kicks |
15:24.11 | CoaxD | Egonis: WOO |
15:24.17 | nestAr | my internal search and replace read that as the more vulgar version.. |
15:24.22 | nestAr | my mind rules |
15:24.46 | Nuttah | fenlander: you have use g729 for your 1899 calls? |
15:24.54 | *** join/#asterisk cinzas (~ashes@83.240.144.145) |
15:24.57 | cinzas | Hi ! |
15:25.00 | Nuttah | because some reason its decided to use it |
15:25.54 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.170.115.68.195.rev.coltfrance.com) |
15:25.55 | jf_ | what mean RFC3389 support incomplete |
15:26.11 | jf_ | turn off client if possible |
15:26.55 | Egonis | CoaxD: How do I restart udev? |
15:26.58 | jonathh | anyone in here familar with .call files? |
15:27.36 | fenlander | Nuttah: I use 711 over IAX2 |
15:27.52 | *** join/#asterisk lancey (Shady@support.net1.cc) |
15:27.54 | lancey | hi guys |
15:28.05 | lancey | anyone knows how to set the inter-digit timeout of LinkSys PAP2 ? |
15:28.06 | *** join/#asterisk kairo (~kairo@200.251.61.124) |
15:28.22 | kairo | hi. The asterisk is one gatekeeper too? |
15:28.34 | lancey | kairo yes it is |
15:28.38 | *** part/#asterisk Egonis (~chultay@69.194.211.129) |
15:28.40 | fenlander | jonathh: what is your problem? |
15:28.42 | lancey | far more than a gatekeeper |
15:28.52 | lancey | noone here dealing with LinkSys PAP2? |
15:29.12 | kairo | And I I can use the asterisk as gatekeeper only? |
15:29.19 | *** join/#asterisk netofsickcoder (~netofsick@stjhts23d054.nbnet.nb.ca) |
15:29.29 | lancey | kairo yes you can |
15:29.49 | ManxPower | ~docs |
15:29.50 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:29.50 | bugbot | docs is assigned nothing and reported nothing. |
15:29.57 | ManxPower | ~mailinglist |
15:29.58 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
15:29.58 | bugbot | mailinglist is assigned nothing and reported nothing. |
15:30.06 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
15:30.55 | cinzas | Anyone here using SIP phones widely ? |
15:31.34 | jf_ | what mean RFC3389 support incomplete |
15:31.35 | jf_ | turn off client if possible |
15:31.43 | ManxPower | I've deployed about 15 SIP phones and am in the process of deploying 60 more. |
15:31.48 | ManxPower | jf_, So do that. |
15:32.16 | jf_ | server side or client |
15:32.16 | cinzas | ManxPower: wich phone do you use ? |
15:32.32 | kairo | lancey: thanks a lot. |
15:32.37 | lancey | jf_ client |
15:32.39 | lancey | obviousli |
15:32.43 | lancey | *obviously |
15:32.44 | jf_ | k |
15:33.29 | ManxPower | cinzas, Polycom IP 300 and 500 |
15:33.36 | cinzas | Hmmm |
15:33.39 | cinzas | thanks ;) |
15:34.26 | jf_ | which should be turn off rtp message on xlite |
15:37.09 | lancey | anyone here with a LinkSys PAP2, last time :> |
15:38.41 | ManxPower | jf_, no, in X-Lite "Transmit Silence = YES" turns off RFC3389 |
15:41.03 | *** join/#asterisk |Vulture| (~Vulture@64.234.204.68.cfl.res.rr.com) |
15:41.21 | |Vulture| | is it possible for voicemail.conf to accept 2 pager #s or do I need to write an AGI? |
15:41.50 | ManxPower | |Vulture|, Dude, /etc/aliases |
15:42.13 | Gand_DJ | Here's a crazy question.. can you setup * to have 2 IVR systems, 1 set of prompts if call comes in from 1 trunk, and another set of prompts if call comes from trunk 2? |
15:42.30 | Egonis | CoaxD: zaptel is now working fine and is in /dev... I now have the following messages: unable to get our ip address ... Skinny disabled -- Unable to open /dev/dsp ... no such file or device (as my server has no sound board) |
15:42.31 | jakepdev | yes |
15:42.48 | DrWho17 | Gand_DJ: sure, just send them to different contexts |
15:42.55 | Gand_DJ | I want to setup zap for family IVR, and voip trunk for business ivr |
15:43.06 | Gand_DJ | k |
15:43.18 | ManxPower | Gand_DJ, CONTEXTS!!!!! |
15:44.03 | DrWho17 | hrm, is asterisk cdr csv format based on anything, or just custom to asterisk |
15:44.08 | ManxPower | Egonis, Do you want to use chan_skinny? Unable to open /dev/dsp is because you are trying to use chan_oss or chan_alsa (or it's autoloading) |
15:44.30 | Egonis | ManxPower: Do I want chan_skinny? how do I disable the chan_oss/alsa? modules.conf? |
15:44.49 | CoaxD | Oh fscking COOL. There's a new article about my Variegated Streptocarpus yahoo group in the new GHA newsletter. I'm dancing. woo. |
15:45.13 | jf_ | manxpower : got it |
15:45.49 | ManxPower | Egonis, You disable Asteirsk modules in /etc/asterisk/modules.conf Skinny is a Cisco protocol for Cisco phones |
15:46.34 | Egonis | ManxPower: no, no need for skinny.. how do I disable? |
15:47.32 | Egonis | ManxPower: when I try to access ext 8500, using box # 1234 w/ pass 5678 or 4242, access is denied |
15:49.01 | Nuttah | hmm this is a long shot, as mobiles dont have carrier dependant number prefix's, is there any way to route outbound calls via various voip providers depending on the carrier? |
15:49.04 | ManxPower | Egonis, noload =>chan_skinny.so noload => chan_alsa.so noload => chan_oss.so |
15:49.14 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
15:49.24 | ManxPower | Egonis, /etc/asterisk/voicemail.conf issue a reload on the cli after you make changes |
15:50.07 | ManxPower | Nuttah, That's only the case in the USA/Canada. In the rest of the world there mobile users use specific prefixes. |
15:50.22 | Egonis | ManxPower: I left voicemail as is... should I change the pwd? |
15:50.25 | ManxPower | In the USA it does NOT cost more to call a mobile than it costs to call a landline. |
15:50.38 | Nuttah | not USA sorry.. UK |
15:50.48 | ManxPower | Egonis, I have no comment on that. Asterisk sample config files are there to show you EVERY option available. they are not designed to work. |
15:50.50 | Nuttah | costs a fecking packet here |
15:51.15 | Nuttah | and MAnx ya wrong wit the number prefixs |
15:51.24 | Egonis | ManxPower: lol.. okay, how do I delete the voicemail waiting for ext1234 manually? |
15:51.25 | ManxPower | Nuttah, that's because the CALLER pays when using mobile. In the USA the person being called pays for the call (for mobile) |
15:51.26 | *** join/#asterisk ronn (ronn@217.46.199.162) |
15:51.32 | ManxPower | ~docs |
15:51.33 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:51.35 | ManxPower | ~mailinglist |
15:51.36 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
15:51.41 | ManxPower | ~RTFG |
15:51.42 | jbot | Another variant of rtf*, 'g' refers to Google or newsGroups |
15:51.43 | bugbot | docs is assigned nothing and reported nothing. |
15:51.43 | bugbot | mailinglist is assigned nothing and reported nothing. |
15:51.43 | bugbot | RTFG is assigned nothing and reported nothing. |
15:52.05 | tzanger | ManxPower: no that's STFW |
15:52.17 | Nuttah | in the UK you you can have aa number on one carrier, and move the number over to another carrier |
15:52.20 | ronn | hi guys |
15:52.31 | ronn | i just installed asterisk on rh el3 |
15:52.34 | tzanger | jbot, STFW is Search The Fucking Web. See http://justfuckinggoogleit.com/ |
15:52.35 | jbot | tzanger: please, watch your language. |
15:52.42 | tzanger | jbot, STFW is Search The F*cking Web. See http://justfuckinggoogleit.com/ |
15:52.43 | jbot | tzanger: please, watch your language. |
15:52.47 | tzanger | jbot, STFW is Search The F*cking Web. See http://justf*ckinggoogleit.com/ |
15:52.48 | jbot | ...but stfw is already something else... |
15:52.48 | Wonka | ha-ha |
15:52.53 | tzanger | ~stfw |
15:52.54 | jbot | stfw is, like, Search the F|_|cking Web: http://www.google.com/ |
15:52.54 | bugbot | stfw is assigned nothing and reported nothing. |
15:52.54 | Wonka | jbot: stfw? |
15:52.55 | jbot | methinks stfw is Search the F|_|cking Wiki: http://voip-info.org/tiki-index.php |
15:53.03 | tzanger | jbot no, STFW is Search The F*cking Web. See http://justf*ckinggoogleit.com/ |
15:53.04 | jbot | okay, tzanger |
15:53.08 | ManxPower | BTW, we can say FUCK here. |
15:53.17 | shido6 | oh my ... |
15:53.22 | jakepdev | double oh my |
15:53.22 | Wonka | fork it! |
15:53.23 | mishehu | my virgin eyes! |
15:53.30 | mishehu | oooooowowowowowoww!!! ManxPower swore! |
15:53.35 | Wonka | stop rape, say yes... |
15:53.40 | ManxPower | It's not considered terribly polite, but we can say it. |
15:53.42 | jakepdev | i thought this forum was G rated |
15:53.48 | Nuttah | ManxPower: UK numbers are movable from one carrier to another |
15:53.55 | ManxPower | Nuttah, That must be new. |
15:54.04 | Nuttah | so I presume the innitial answer is no :P |
15:54.15 | Nuttah | ManxPower: not really. |
15:54.19 | *** join/#asterisk brc-tux (~brc-tux@pD9E9A2F2.dip0.t-ipconnect.de) |
15:54.19 | newl | Number portability isn't anything new. :) |
15:54.26 | *** part/#asterisk brc-tux (~brc-tux@pD9E9A2F2.dip0.t-ipconnect.de) |
15:54.32 | DrWho17 | Nuttah: portable here too |
15:55.09 | jakepdev | ~STFW |
15:55.11 | jbot | somebody said stfw was Search The F*cking Web. See http://justf*ckinggoogleit.com/ |
15:55.11 | bugbot | STFW is assigned nothing and reported nothing. |
15:55.28 | ManxPower | bugbot needs to be sprayed with poison |
15:55.30 | jakepdev | case sensitive as well as profanity sensitive |
15:55.40 | *** join/#asterisk bazzz (~baz@atlnga1-ar3-4-3-007-122.atlnga1.dsl-verizon.net) |
15:55.59 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
15:56.34 | jakepdev | oh - my bad - it worked before |
15:56.48 | Juxt | hello peeps |
15:57.00 | jakepdev | hello |
15:57.19 | jakepdev | ~nickometer jakepdev |
15:57.19 | bugbot | nickometer jakepdev is assigned nothing and reported nothing. |
15:57.19 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
15:57.34 | Zeeek | gentlefolk... |
15:59.20 | Gand_DJ | Does this look right... I hope this is the right info to post. |
15:59.21 | Gand_DJ | http://pastebin.ca/9608 |
15:59.25 | jakepdev | ~kill bugbot |
15:59.27 | jbot | ACTION shoots a excited quark gun at bugbot |
15:59.27 | bugbot | kill bugbot is assigned nothing and reported nothing. |
15:59.37 | Gand_DJ | incoming fwd calls get busy signal |
16:00.39 | jakepdev | GandDJ - IAX or SIP? |
16:00.44 | Egonis | Any suggestions for testing festival w/ asterisk? |
16:00.47 | Gand_DJ | iax |
16:00.53 | Gand_DJ | I can call out fine |
16:00.57 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
16:01.01 | jakepdev | can you show your iax2 debug? |
16:01.25 | Gand_DJ | what's the command again from root? |
16:01.30 | *** join/#asterisk cp5 (~samy@chcgil2-ar7-4-3-040-086.chcgil2.dsl-verizon.net) |
16:01.41 | jakepdev | it's in the CLI |
16:01.49 | jakepdev | iax2 show debug - i think |
16:01.56 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
16:02.00 | BuckRogers | good morning |
16:02.04 | ManxPower | iax2 debug |
16:02.16 | Zeeek | and iax2 no debug |
16:02.20 | Gand_DJ | [root@asterisk1 root]# iax2 debug |
16:02.20 | Gand_DJ | -bash: iax2: command not found |
16:02.24 | ManxPower | "help" is a good command to know, BTW. It's spelled "help" |
16:02.33 | jakepdev | now Gand - in the CLI |
16:02.34 | jonathh | not in bash |
16:02.36 | jonathh | in the CLI |
16:02.38 | ManxPower | Gand_DJ, you putz. Asterisk commands are done in the Asterisk CLI |
16:02.49 | jakepdev | asterisk -r |
16:02.55 | Nuttah | christ manx... got a bug up ya ass as well? :) |
16:03.00 | Gand_DJ | sorry..lol... been ages since I played with linux |
16:03.00 | jonathh | or asterisk -cvvvvvvvv |
16:03.21 | Gand_DJ | ok.. debugging enabled |
16:03.27 | Gand_DJ | going to try to call my * box |
16:03.48 | Gand_DJ | nothing shows up. |
16:03.52 | Gand_DJ | seems to not even hit the box |
16:03.58 | Zeeek | ManxPower - have you ever heard of a problem like this? A few days ago most servers start becoming UNREACHABLE for one cycle, i.e., a few seconds then REACHABLE. It isn't the provider since several at once, so it's a network issue probably. When it happens, all flow stops to the connection. (ADSL) |
16:04.04 | Gand_DJ | verbosity is at least 3 |
16:04.11 | BuckRogers | Question, What is the largest calling capasity card that is supported by enterprise asterisk, running on sun solaris OS, looking into buying hardware probly need something that supports fiber a plus |
16:04.12 | Zeeek | This happens once every 30 minutes or so |
16:04.31 | Gand_DJ | I have port 4569 forwarded from router to * |
16:04.31 | jakepdev | Gand_DJ - do a show peers |
16:04.34 | Gand_DJ | ok |
16:04.34 | Zeeek | replaced router/modem, network= card |
16:04.50 | Zeeek | tried every possible network test |
16:05.04 | jakepdev | is FWD listed? |
16:05.11 | Gand_DJ | fwd/520214 65.39.205.121 (S) 255.255.255.255 4569 Unmonitored |
16:05.28 | Gand_DJ | I'm using *@home |
16:05.29 | algorithmn | BuckRogers: i've heard about that sbc card, but not fiber.. ;-( |
16:05.35 | Gand_DJ | want to see my setup through the amp interface? |
16:05.51 | BuckRogers | has anyone else had any large scale deployement experince here? |
16:05.55 | jakepdev | hmm.. looks like it registered |
16:06.02 | BuckRogers | Algorithmn, thanks |
16:06.13 | Gand_DJ | yeah. I can make outgoing fine |
16:06.21 | Gand_DJ | wierd |
16:06.30 | BuckRogers | Mr spencer are you watching the room? |
16:06.37 | jakepdev | outgoing calls are made within your dialplan |
16:06.56 | jakepdev | your incoming depends also on the config of iax.conf |
16:06.59 | algorithmn | BuckRogers: lol. there isn't much burocracy to circumvent if that one works... |
16:07.13 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
16:07.15 | ManxPower | I need this http://story.news.yahoo.com/news?tmpl=story&cid=573&e=20&u=/nm/science_clock_dc |
16:07.51 | jakepdev | that's great if it works |
16:07.57 | ManxPower | Zeeek, Yes, I've seen it. It seemed to be to be an ISP issue. |
16:08.24 | BuckRogers | so has anyone ran asterisk on a sun solaris os |
16:08.32 | BuckRogers | with fiber ethernet? |
16:08.34 | Zeeek | ManxPower is there something like mtr for commandline? |
16:08.56 | Zeeek | I have them checking it but it's really hard to catch (every half hour for 10 sec!) |
16:09.10 | Zeeek | I got the two upstreams to see it |
16:09.26 | Zeeek | they are now contacting the telco that owns the wires |
16:09.41 | Zeeek | anyway, it makes voip USELESS! |
16:11.54 | Zeeek | anyone know a network utility like mtr or pingplotter ? |
16:11.55 | DrWho17 | BuckRogers: what's a large scale deployment? |
16:12.05 | Zeeek | big people need a large scale |
16:12.40 | BuckRogers | trafic at super "pop" on a national network |
16:13.09 | nestAr | super size my deployment! |
16:13.10 | BuckRogers | Drwho17 needs to beable to handle super pop capistity |
16:13.42 | DrWho17 | well, that is different per "super pop" |
16:13.49 | newl | indeed |
16:13.50 | DrWho17 | how about calls/hour |
16:14.04 | ManxPower | Zeeek, run a ping and see if you drop packets during the UNREACHABLE" |
16:14.24 | Zeeek | done it, yes |
16:14.34 | Zeeek | plus at the provider she saw packet loss |
16:14.43 | Zeeek | which is why she reported to upstream |
16:14.44 | BuckRogers | DrWho17 i think at 20,000 continous calls |
16:14.47 | ManxPower | Zeeek, It COULD be some other issue. NAT timeout? Crappy Router (my old Cisco Cablemodem/NAT Router could not handle many small packets, it dropped them on the floor every few mins during an IP call) |
16:14.55 | Zeeek | but man, it's hard to troubleshoot a once in a while condition |
16:15.17 | BuckRogers | the really high capisty where national network interconnections |
16:15.19 | ManxPower | Zeeek, Try trobuleshooting a problem that only happens every few WEEKS |
16:15.21 | DrWho17 | BuckRogers: ouch, good luck with that |
16:15.21 | Zeeek | ManxPower a) this is new and I've been running for a year. b) I swapped for different router |
16:15.22 | foobos | zeek, mtr is commandline. |
16:15.49 | ManxPower | Zeeek, Prolly ISP issue then |
16:15.49 | Zeeek | foobos it won't make on my box with no win |
16:15.52 | BuckRogers | yeah i need ot contact a asterisk driver programer i think |
16:16.05 | Zeeek | what's the iptraffic prog again? |
16:16.10 | Zeeek | iptraf ? |
16:16.14 | foobos | zeeek, 0.26 version compiles with curses/ncurses atleast |
16:16.16 | ManxPower | Zeeek, SOME DSL modems allow you to see the line conditions. Poke around the mfgr's web site. |
16:16.24 | DrWho17 | well, you'll need 100's of asterisk boxes clustered together actually, you are better off buying a switch I think |
16:16.35 | newl | Zeeek: if you suspect the fault is on a PRI or DSL, have the telco place QM on the service(s). |
16:16.58 | BuckRogers | why not have asterisk running on a sun micro machine with 24 64bit processors |
16:16.59 | Zeeek | newl that's what is about to happen,but man this is gonna take forever to find |
16:17.00 | DrWho17 | 4-port T1 cards for asterisk are $1500, each handles 96 calls |
16:17.23 | ManxPower | I had a problem where my signal from the CO was too weak. Showed up as packet loss |
16:17.23 | BuckRogers | yeah we already have one of those |
16:17.25 | Zeeek | The telco HAD a monopoly, they may not be that cooperative with the ISP |
16:17.26 | DrWho17 | BuckRogers: well you could do that, but Intel is much more economical |
16:17.30 | DrWho17 | and powerful |
16:17.46 | newl | Zeeek: nah, 7-10 days for the average QM will ususally show BER if there's problems. |
16:17.50 | BuckRogers | i think we may need to use a cisco cat switch to get it to ether net sip |
16:17.56 | Zeeek | in the meantime, I have to tell everyone, ok if you stop hearing me, wait thru 10 seconds of silence and I'll be back |
16:18.16 | Zeeek | fucking annoying to say the least |
16:18.24 | BuckRogers | Drwho what duel zeons or quads i dont think it would be that power fulll |
16:18.26 | Zeeek | and unusable for bizness to say the rest |
16:19.00 | BuckRogers | how much do you think markspencer charges for a day of his time? |
16:19.16 | newl | how long is a piece of string? :) |
16:19.30 | DrWho17 | how is going to help you? You need to have a better definition of your problem |
16:19.35 | BuckRogers | and how much does he like the night life, got to boogie |
16:19.39 | darkskiez | BuckRogers, How much you got? probably $1 more. |
16:19.57 | BuckRogers | i dont think he is an unreasable man |
16:20.07 | BuckRogers | why do you inply that he is darkskiez |
16:20.28 | darkskiez | because he gave his code to the hippies. |
16:20.40 | BuckRogers | right..... |
16:20.41 | *** join/#asterisk jmacz (~jmacz@63.245.86.225) |
16:21.20 | Moc[NX] | BuckRogers: he is busy. You better reach a consultant instead of mark itself. |
16:22.09 | BuckRogers | Moc[NX]: do you have any suggestions, i think if he knew what we are working on he would make time |
16:23.06 | newl | Was that 20k concurrent calls? |
16:23.14 | BuckRogers | yes |
16:23.19 | Zeeek | newl what is BER |
16:23.27 | BuckRogers | could be more in larger meto areas |
16:23.28 | newl | Zeeek: Bit Error Rate |
16:23.30 | Zeeek | ok |
16:23.34 | newl | ~ber |
16:23.35 | bugbot | ber is assigned nothing and reported nothing. |
16:23.38 | Zeeek | heh |
16:23.47 | Moc[NX] | BuckRogers: I dont see the need to talk to mark about that |
16:23.48 | newl | bugbot needs to die |
16:24.12 | Zeeek | this is one of those "no one else is having this problem..." situations |
16:24.17 | newl | jbot: BER is Bit Error Rate |
16:24.18 | jbot | okay, newl |
16:24.20 | BuckRogers | Moc[NX |
16:24.27 | Moc[NX] | He be in Toronto next week giving a conference at Von |
16:24.31 | Moc[NX] | or just email him |
16:24.33 | BuckRogers | i adgree but i do not know who else to talk |
16:24.55 | Moc[NX] | BuckRogers: , just call digium, if it need to get to mark, they will do that |
16:24.58 | BuckRogers | that what will probly happen or ill call him |
16:25.15 | Zeeek | making mtr complains he can't find resolver library |
16:25.56 | Moc[NX] | Even if I had to do a 100k phone setup, I dont see the need to contact mark except telling him that I got this really cool sucess story ;) |
16:26.00 | *** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca) |
16:26.52 | jf_ | is there any to configure xlite that if im local to the * use subnet ip address, if im remote (internet) using the dns of * |
16:27.03 | Moc[NX] | salut jf |
16:27.06 | BuckRogers | it for the use of abstract hardware and signaling at that large cap level |
16:27.09 | jf_ | Moc: salut |
16:27.25 | newl | 20k concurrent users..that's definately a shirtload. Would be easier to buy a switch that'd handle that number of users no sweat like AXE or if all you're doing is routing, a S12. hehe |
16:27.30 | BuckRogers | perferable sunsolaris |
16:27.46 | BuckRogers | its more than routing though |
16:28.09 | Moc[NX] | if I were you, I would stick with Linux |
16:28.16 | BuckRogers | its a whole knew level of voip trafficing |
16:28.26 | Moc[NX] | you will get all kind of weird problems if you try to use anything else |
16:28.32 | newl | Well, S12 is capable of hanging subscribers off of as well..it's just not primarily designed for that purpose. |
16:28.39 | BuckRogers | yeah but you can run linux on some sun hardware |
16:29.02 | foobos | even if linux runs on sun hardware, its not necessarily a good idea |
16:29.03 | Moc[NX] | who care about sun hardware ? |
16:29.17 | BuckRogers | high processing power |
16:29.31 | BuckRogers | large memory capasity |
16:29.39 | algorithmn | you cannot use 120gb of system ram nor 180mb of proc cache using linux with sun hardware... |
16:29.43 | Moc[NX] | BuckRogers: Server price are soo low these day... just buy multiple of them |
16:30.09 | foobos | i'd just go with cheap opterons and link them together |
16:30.10 | BuckRogers | Moc that is an alternative |
16:30.16 | foobos | it would be pity if that one expensive box go down |
16:30.17 | BuckRogers | IBM blade servers |
16:30.21 | BuckRogers | ect. |
16:30.30 | Moc[NX] | yep, and if 1 fail, only 1 fail |
16:30.55 | *** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
16:31.04 | Moc[NX] | if you need to do maintenance, you can ofload 1 blade, and do it on that one, and switch it back in production |
16:31.05 | foobos | only thing is of course heat output, powerusage and rack space constraints |
16:31.59 | *** part/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
16:32.01 | BuckRogers | yeah it has some advantages, but that still doesnt address my problem of asterisk not being able to handle fiber cards? |
16:32.09 | BuckRogers | oc signalling |
16:32.22 | denon | BuckRogers: no OC cards, but it does handle a DS3 card now |
16:32.25 | foobos | why does asterisk has to handle that directly |
16:32.52 | denon | I cant imagine why you'd ever want to handle more than a single DS3 per asterisk switch anyway |
16:33.09 | Moc[NX] | same here |
16:33.12 | denon | unless you wanted to run an entire carrier solution on a single switch |
16:33.14 | BuckRogers | there is reasons that i can not disclose |
16:33.34 | denon | you're going to have to disclose them if you want us to help you with a solution :) |
16:33.53 | DrWho17 | denon: routing calls via PRI |
16:33.54 | BuckRogers | i hear ya |
16:34.30 | Moc[NX] | 1 DS3 per 1U server is a good ratio I think |
16:34.31 | foobos | buckrogers, well you could buy an expensive Juniper router and run asterisk inside it (its basically freebsd) |
16:34.53 | CoaxD | Moc[NX]: Um |
16:35.11 | denon | I dont think asterisk is really designed to be a high-density termination server... |
16:35.14 | DrWho17 | Moc: 2 DS3 per 2U then |
16:35.14 | CoaxD | Moc[NX]: Thats a hell of a lot to ask out of asterisk on a 1U server |
16:35.37 | BuckRogers | who said anything about termination |
16:35.41 | DrWho17 | denon: yea, probably not, I'm still investigating |
16:35.56 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
16:36.02 | CoaxD | Hell, there are people reporting good results w/ a quad T1 card in a dual Xeon |
16:36.36 | Moc[NX] | CoaxD: I got a Dual Xeon 2.8 1gig ram, PowerEdge 1850 (1U) and it could have being bosted |
16:36.37 | DrWho17 | no transcoding |
16:36.52 | Moc[NX] | so we are talking of about 27k channel per Rack |
16:36.53 | DrWho17 | just hairpinning calls around |
16:37.07 | BuckRogers | have you ran transcoding, or pri to sip signalling |
16:37.18 | CoaxD | Moc[NX]: Hmmm. Wasn't aware that a xeon proc could fit in a 1U |
16:37.21 | moy | hi everybody, how can i increment the volume in wich the Playrecord() files are reproduced? |
16:37.25 | bkw_ | this _. vs _X. is pissing me off |
16:37.33 | CoaxD | Moc[NX]: (Much less *2* Xeons.) |
16:37.38 | Moc[NX] | CoaxD: yes they do hehe |
16:37.51 | CoaxD | Moc[nx]: They must be making xeons considerably smaller than P3 Xeon days |
16:37.56 | Moc[NX] | check the PowerEdge 1850 |
16:37.58 | moy | bkw, whi? |
16:37.58 | newl | bkw_: go back to working on SMS then. :D |
16:38.03 | CoaxD | Moc: They were *huge* |
16:38.10 | bkw_ | newl, waiting on modem to arrive |
16:38.10 | bkw_ | :P |
16:38.15 | Moc[NX] | CoaxD: it not using cardrige anymore |
16:38.28 | CoaxD | Moc[NX]: Socket based now? |
16:38.33 | Moc[NX] | yep |
16:38.38 | CoaxD | Moc[nx]: Much better. yes. |
16:38.45 | BuckRogers | all i need this asterisk sun machine to do is pass traffic |
16:38.50 | CoaxD | Moc[nx]: Those big ass cartridge monsters were for the birds. *lol* |
16:38.58 | BuckRogers | at high capsity= |
16:39.08 | CoaxD | BuckRogers: Its not just "passing traffic" |
16:39.32 | BuckRogers | What makes you say that? |
16:40.05 | newl | If handling traffic is all that you want, considered SER? |
16:40.45 | BuckRogers | still need to run a mysql database and api'modes also |
16:40.55 | DrWho17 | newl: he's going to put something to handle SIP in a "super pop"? |
16:40.55 | *** join/#asterisk [Outcast] (~bill@c-24-218-94-11.hsd1.ma.comcast.net) |
16:41.16 | DrWho17 | newl: the calls will come in via PRI handoff, or SS7 Trunks |
16:41.31 | BuckRogers | DrWho17, exactly |
16:41.40 | DrWho17 | asterisk is not a solution for this |
16:41.46 | newl | DrWho17: well, PoP implies termination. He didn't say anything about termination. |
16:42.13 | jmacz | hi everyone, I have a problem with the atxfer function. I edited the features.conf en reload the "res_features.so" at the CLI but it doesn't work. Any idea? |
16:42.17 | DrWho17 | newl: right, he wants to take calls in and send voip calls to their voip destination and terminate modem calls probably |
16:42.27 | BuckRogers | Super pops have interconnections from one service provider to another |
16:42.36 | DrWho17 | where asterisk makes the switching decision, but asterisk won't be good at that |
16:42.43 | BuckRogers | DrWho17, no |
16:42.47 | DrWho17 | especially not for a deployment such as 20000 calls |
16:43.13 | BuckRogers | needs to foward traffic or not foward traffic |
16:43.24 | jmacz | normal trasfer works ok but I can't get atxfer to work on 1.0.3 |
16:43.32 | [Outcast] | DrWho17: you can do it with about 10 SGI servers |
16:44.02 | DrWho17 | Outcast: well, they are connected to media gateways |
16:44.11 | [Outcast] | yep |
16:44.25 | DrWho17 | which handle a lot of the heavy lifting |
16:44.26 | _Brian | does anyone by any chance have a copy of the app: NVLineDetect ? |
16:44.45 | [Outcast] | you can still let asterisk handle the transcoding on those machines |
16:45.09 | BuckRogers | it needs to come in and out in the same signalling format.. |
16:45.19 | DrWho17 | well, I'd rather just get a switch that does it all you know |
16:45.29 | BuckRogers | there not smart enough |
16:45.40 | DrWho17 | if you are talking 20,000 simultaneous calls, you should have enough $$$ to buy one |
16:46.14 | BuckRogers | we developed the programming to run in the envoriment no one else has it to sell so we developed it our selves |
16:46.17 | DrWho17 | probably less then 2000 96-port digium cards |
16:46.35 | [Outcast] | convergent network has a level 5 switch that will do it I think |
16:46.50 | [Outcast] | big $$$ though |
16:46.53 | DrWho17 | yea, or santerra |
16:46.57 | DrWho17 | or metaswitch |
16:47.04 | DrWho17 | or sentito |
16:47.10 | DrWho17 | whatever, plenty of them |
16:47.15 | [Outcast] | convergent is easy to setup though |
16:47.47 | newl | $150k will get you an AXE switch capable of 32k subs outta the crate. :) |
16:47.49 | CoaxD | There is not enough processing power within a 1U box to handle 2,000 calls, let alone 20,000 calls |
16:47.54 | *** join/#asterisk goldenear (~goldenear@d149.dhcp212-198-168.noos.fr) |
16:47.57 | DrWho17 | newlP oh, not bad |
16:48.19 | DrWho17 | if it does SIP/ATM Switching/MGCP/h.248 I'll take it |
16:48.24 | CoaxD | hell, there's not enough processing power in a powerhouse quad xeon 4U boxes to handle 2,000 calls |
16:48.28 | BuckRogers | nah we figure we would need rack size equipment |
16:48.30 | newl | DrWho17: Alcatel isn't all that bad. ;) |
16:48.31 | fearnor | there's telica 1U switch now ;) |
16:48.40 | DrWho17 | yea, lucent bought them |
16:48.47 | tclark | why does vm play the voice mails msg in reverse date sequence in the OLD msg folder .. |
16:48.49 | fearnor | coax: it depends what exactly you are doing. single SER box can handle 2000 calls no sweat |
16:48.55 | tclark | err doesnt |
16:49.13 | fearnor | and it also matters if you are just doing mgc, in which case, you need to worry about call setup per second, not concurrent calls |
16:49.16 | CoaxD | fearnor: Hmm. really? 2000 calls, 1 box? |
16:49.20 | fearnor | concurrent calls take almost no resources |
16:49.24 | fearnor | call setup/teardown does |
16:49.39 | CoaxD | fearnor: Hmm. Well, a sip proxy doesn't really have to do any transcoding, i wouldnt think. |
16:49.45 | fearnor | that's right ;) |
16:49.46 | CoaxD | fearnor: Nor does it actually have to process the data coming into it |
16:49.51 | CoaxD | fearnor: It just has to relay it |
16:49.52 | fearnor | that's exactly right. |
16:50.04 | CoaxD | fearnor: Which basically makes it a data relayer. Dumb mode ON. |
16:50.10 | fearnor | this is why the world has a distinction between MG and MGC. |
16:50.11 | CoaxD | fearnor: 2000 calls @ 64kbps, tho.. Oww |
16:50.17 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
16:50.20 | fearnor | it doesn't even relay the data. |
16:50.27 | DrWho17 | newl: yea, can't find those on Ebay though |
16:50.29 | fearnor | it just CONTROLS THE CALLS |
16:50.45 | newl | DrWho17: hahaha no, you probably wouldn't. :) |
16:50.46 | CoaxD | fearnor: (if ser is taken out of the media path right after start of the sip conversation, yea) |
16:51.04 | fearnor | its all about differentiating media path from call control path |
16:51.12 | fearnor | ktnxbye |
16:52.30 | CoaxD | fearnor: Heh :) |
16:52.47 | [Outcast] | msg if you know how to get in touch with anthem outside of mirc |
16:53.54 | *** join/#asterisk eidolon (~eido@seawall.homeport.org) |
16:54.59 | eidolon | hihifolks. i have a quick question. i have a PBX that I scrounged from eBay about 10 years ago, and it's really starting to fall apart. We have 12+ extensions feeding 3 POTS lines into the house. I'd love to toss the whole thing and replace it with Asterisk, but i'm not sure how costly it would be. I don't want to park PC's at every extension - have VOIP phones gotten cheap enough to get a dozen or so for, oh, < $500? |
16:55.58 | lancey | LinkSys PAP2 costs $63 |
16:56.03 | lancey | and you get 2 POTS lines with it |
16:56.04 | lancey | :) |
16:56.16 | eidolon | Hmm. |
16:56.17 | reallost1 | How do you detect phone numbers that are out of service? The T1 line never answers and doesn't seem to return a code. |
16:56.18 | lancey | and it really works :) |
16:56.45 | [Outcast] | hey is possible to unlock the sip adapter on the linksys |
16:57.01 | eidolon | ohhh. that's pretty neat. |
16:57.18 | eidolon | so i could use the PAP2 as a gateway to POTS phones, and it'll talk to the Asterisk server? |
16:57.19 | gambolputty | grandstream bt102 about $75 |
16:57.23 | eidolon | (2 phones at a time) |
16:57.30 | gambolputty | bt101 about $65 |
16:58.07 | Qwell | eidolon: You need the PAP2-NA, make absolutely sure it has the NA |
16:58.15 | Qwell | anything else will not work with astrerisk |
16:58.17 | Qwell | asterisk... |
16:58.24 | [Outcast] | has anyone been having problem with res_perl in the cvs head? |
16:58.40 | eidolon | okay. |
16:58.52 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
16:58.59 | eidolon | that's good, i could get a couple pap2's and use existing analog pots phones, plus get some BT101's for the power-users (like me :) |
16:58.59 | Qwell | eidolon: those ones are generally locked to a specific provider, and its difficult, if not impossible, to unlock them |
16:59.19 | eidolon | i'll need an FXO card for the server to hook into the POTS lines from the service, yes? |
16:59.20 | jets | Is there any commercial vendors supporting asterisk text to speach? |
16:59.24 | jets | speech even |
16:59.24 | Qwell | eidolon: correct |
16:59.27 | eidolon | okay. |
16:59.29 | eidolon | good :) |
16:59.37 | Qwell | eidolon: something like a tdm400p, or maybe a SPA3000 |
16:59.43 | [Outcast] | voice geine |
16:59.48 | Qwell | I hear the SPA's are good |
17:00.00 | eidolon | i was looking at the voicetronix openline4 |
17:00.07 | [Outcast] | the spa's rock |
17:00.10 | Qwell | I'm all for the tdm though |
17:00.17 | lancey | bye guys |
17:01.08 | Qwell | eidolon: if you were to use a TDM for the pots lines, it would also give you a trusty timing device for things like MeetMe() |
17:01.10 | eidolon | okay, last question. i have a dual-PIII-700 rackmount server that would make an excellent host for asterisk (1gig RAM) - is that enough horsepower to handle no more than 4 sessions at a time? |
17:01.16 | Hmmhesays | got a guy telling me he's got a loop start pri |
17:01.26 | Qwell | eidolon: That would probably handle a little more then 4 at a time |
17:01.30 | eidolon | heh |
17:01.37 | eidolon | okay, good 8) |
17:01.58 | eidolon | oh good, and asterisk is even in debian Sarge. life is good. |
17:01.59 | Moc[Work] | www.voncanada.com |
17:02.04 | *** join/#asterisk ManxPower (~eric@adsl-35-236-60.msy.bellsouth.net) |
17:02.06 | |Vulture| | eidolon: that would easilly handle it |
17:02.16 | Qwell | eidolon: people have had problems with the debian packages |
17:02.21 | eidolon | oh? |
17:02.23 | Hmmhesays | you should get 1.07 if you apt-get asterisk and have testing as your default |
17:02.26 | Qwell | You'd probably be better off using the source, and compiling stuff yourself |
17:02.39 | gambolputty | Does Asterisk support a plus sign in front of an E164 number? |
17:02.46 | eidolon | getting compilation working on debian stuff is... tedious. mostly because debian by default doesn't install all the supporting stuff needed for builds. |
17:02.50 | eidolon | but maybe i'll try it. |
17:02.52 | |Vulture| | hmmm strange... spandsp doesn't seem to be saving inbound faxes |
17:03.05 | Qwell | eidolon: well, try the (testing) packages, and see how it works out |
17:03.10 | eidolon | okee. |
17:03.35 | eidolon | what VOIP client do most folks under linux use? |
17:03.37 | Qwell | compiling is good, because you can switch to cvs head easily if you need to test anything |
17:03.47 | Qwell | For a softphone you mean? |
17:03.53 | eidolon | yeah. |
17:04.02 | Qwell | I like iaxcomm |
17:04.06 | eidolon | (i'm VERY new at this, sorry for the n00b questions) |
17:04.24 | eidolon | i'd like to use the internal speaker/mic for a headset, and just call right through the PC. not using an external handset. |
17:06.15 | goldenear | iaxcomm is nice indeed |
17:06.49 | goldenear | but it doesn't support for call transfer or hold when native bridged :( |
17:08.03 | *** join/#asterisk jmacz (~jmacz@63.245.86.225) |
17:08.28 | *** join/#asterisk Mike (~mike@201.135.48.119) |
17:08.59 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
17:08.59 | *** mode/#asterisk [+o bkw_] by ChanServ |
17:09.09 | fearnor | hrmph |
17:09.14 | fearnor | early media is pissing me off |
17:09.45 | *** join/#asterisk kmelillo (~Armitage@wghi.net) |
17:10.30 | eidolon | iaxcomm looks good. |
17:10.31 | Hogie | http://gallery.cyberjunky.net/Work_Pictures/P0009081 <-- Ever seen a Cisco box taped up like that? |
17:10.35 | eidolon | not in debian packages, unfortunately :) |
17:10.56 | eidolon | hogie: that host fails. |
17:11.48 | Hogie | eidolon: works for me, and everyone else I give it too... |
17:11.57 | fearnor | tape packaging++ |
17:12.08 | eidolon | 'gallery.cyberjunky.net could not be found.' |
17:15.32 | goldenear | eidolon, did you try the call tranfer function ? |
17:16.03 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
17:16.06 | eidolon | i was just looking at the site and the sscreenshots. i don't even ahve a server active yet. |
17:16.14 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
17:16.19 | goldenear | ok |
17:17.43 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
17:17.46 | goldenear | But I guess the call transfer issue of iaxcomm is an iax conception issue... |
17:18.50 | eidolon | in all honesty, i'd rather use a SIP phone. but having a softphone can be nice too. particularly if i can tunnel through to the asterisk server remotely. :) |
17:20.06 | goldenear | that's the big bennefit of IAX |
17:21.00 | goldenear | but the fact that the sig doesn't go thrue the server during a native bridging has some bad consequences ... |
17:21.04 | eidolon | well, phase 1 will be getting a platform and asterisk running on it :) |
17:21.20 | *** join/#asterisk asteriskn00b (asteriskn0@wsip-68-15-113-233.ok.ok.cox.net) |
17:21.43 | asteriskn00b | anyone here using or had any success with the xten eyebeam softphone? |
17:22.03 | eidolon | hm. guy on ebay selling 85 grandstream bt101 phones for $65 each. |
17:22.08 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
17:23.10 | nvrswork | can you use a hostname for externip, ie. externip = my.dyndns.org |
17:23.13 | tzanger | ManxPower: do you really think I'm that stupid? (re: dialplan 'i' and DIALSTATUS bugs) |
17:25.44 | *** join/#asterisk AngelGabriel (~angel@host81-133-190-29.in-addr.btopenworld.com) |
17:26.31 | AngelGabriel | The whole concept of asterisk, has blown my mind, WELL DONE to the developers, and people that have taken thier time to make it work |
17:27.17 | AngelGabriel | I have just one query - is it possible to route calls out onto the internet? And if so, how do I pay for them? |
17:27.34 | tzanger | AngelGabriel: you have a voice over internet provider and send the calls to them, and they bill you for the minutes used |
17:27.55 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
17:28.26 | Gand_DJ | are there any voip providers that offer iax for inbound? Some offer it for outbound, but all seem to be sip for inbound |
17:28.42 | kmelillo | would it be possible to host Asterisk on the net, and give subscribers extension numbers, and have them use Soft Phones to dial each others extensions? |
17:29.24 | eidolon | kmelillo: i don't see why not. |
17:29.33 | eidolon | that's basically what skype does, isn't it? |
17:29.42 | kmelillo | I dont know.. heh |
17:29.53 | eidolon | www.skype.com :) |
17:30.02 | kmelillo | I just installed Asterisk on 2 machines, and I am trying to get them to communicate together... via prefix dialing |
17:31.00 | AngelGabriel | tzanger, Thanks ... I'll now go google for the info I need! I'm in the UK, can anyone recommend a supplier? and can I have more than one supplier? |
17:32.50 | tzanger | AngelGabriel: you can have as many as you like. :-) |
17:33.18 | goldenear | eidolon, skype is p2p based... |
17:33.19 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
17:34.36 | goldenear | kmelillo, do you plan to have many subscribers ? |
17:34.37 | eidolon | true, but they use a central accounting / auth system, as well as a gateway i'd assume. |
17:34.53 | kmelillo | goldenear, 20 or so... |
17:35.12 | goldenear | ok so should not be a problem |
17:35.23 | kmelillo | we do a radio show, and need a way to record, once we are all connected... can Asterisk do conference recording? |
17:35.54 | bjohnson | tzanger: I find it advisable to avoid asking other people if they think I'm stupid |
17:36.12 | kmelillo | bjohnson, heh.. if you have to ask?!? |
17:36.40 | bjohnson | AngelGabriel: lots of voip suppliers listed on the wiki |
17:36.51 | tzanger | bjohnson: :-) |
17:36.55 | *** part/#asterisk Egonis (~chultay@69.194.211.129) |
17:36.57 | bjohnson | AngelGabriel: maybe you should take a cold shower |
17:37.05 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
17:38.01 | bjohnson | kmelillo: I think it can record conference calls .. but you'd better google on that |
17:38.32 | bjohnson | the normal call conferencing app is called meetme .. but I understand there are one or two other public ones and a few private (commercial) ones |
17:38.55 | drumkilla | the standard meetme has a recording option |
17:39.03 | bjohnson | btw .. meetme is p2p based as well |
17:40.20 | *** join/#asterisk Signuts (~signuts@209.172.11.54) |
17:41.04 | Signuts | This is all speculation, but Anyone aware of gmail marking all messages from Asterisk-Users as spam? Are they trying to pull a fast one, Perhaps this is related to them starting a Asterisk-test on google groups. |
17:43.16 | Pinhole | It's because of all the ex-M$ employees working at google that don't know they've changed employers. |
17:43.19 | jf_ | is it possible for * to call a vonage number |
17:44.34 | Signuts | gmail just recently (perhaps yesturday) starting marking every message for the Asterisk-Users mailing as spam on my gmail account. |
17:44.45 | Signuts | This is rediculous. |
17:45.03 | Pinhole | that's why I use hotmail. |
17:45.09 | *** join/#asterisk AntiPool (~bostjan@BSN-77-143-148.dsl.siol.net) |
17:45.21 | Pinhole | they don't have any ex-M$ employees over there. |
17:45.38 | Signuts | man |
17:45.47 | AntiPool | i'm reading asterisk docs and if u understand correctly i can use isdn card for a start ? |
17:46.31 | goldenear | if it's isdn4l or capi compatible, yes |
17:46.44 | Egonis | how do I listen to the hold music? |
17:46.48 | AntiPool | what about on fbsd ? |
17:46.55 | Egonis | is there an extension I can dial? |
17:47.13 | *** join/#asterisk jeffik (jefik@69.158.30.24) |
17:47.18 | goldenear | you have to create an extension for that |
17:47.27 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
17:47.29 | Egonis | goldenear: Where would I find a howto on this? |
17:47.29 | jf_ | is it possible for * to call a vonage number directly trough the internet |
17:48.13 | goldenear | jf_, why could not * do this ? |
17:48.16 | sivana | ~tzanger |
17:48.17 | jbot | tzanger is probably some kind of fcking idiot |
17:48.17 | bugbot | tzanger is assigned nothing and reported nothing. |
17:48.28 | Signuts | jf_, yes, it's called SIP, you must have a vonage account. |
17:49.13 | sivana | ~toyk |
17:49.17 | sivana | ~royk |
17:49.18 | jbot | extra, extra, read all about it, royk is .no body |
17:49.19 | bugbot | toyk is assigned nothing and reported nothing. |
17:49.19 | bugbot | royk is assigned nothing and reported nothing. |
17:49.21 | jf_ | ok can u tell me the parameter to do that, or do u have a tut |
17:50.10 | Dutts | hi guys how do I find out which version of * is running from the CLI? show version doesn't seem to give me a proper version number like 1.x.x |
17:50.14 | sivana | jbot: no, sivana is one of the brightest stars out there, ok? |
17:50.16 | jbot | okay, sivana |
17:50.26 | goldenear | Egonis, many howtos on voip-info.org |
17:50.41 | tzanger | jbot no, sivana is not one of the brightest stars out there |
17:50.42 | jbot | tzanger: okay |
17:50.45 | sivana | lol |
17:51.09 | tzanger | damn I was hoping you weren't paying attention |
17:51.21 | sivana | ~sivana |
17:51.22 | jbot | rumour has it, sivana is not one of the brightest stars out there |
17:51.22 | bugbot | sivana is assigned nothing and reported M2515. |
17:52.07 | Dutts | my show version says Asterisk CVS-HEAD-03/09/05-02:14:09 |
17:52.40 | RoyK | bugbot: I'm indeed reporting stuff |
17:53.00 | drumkilla | ~drumkilla |
17:53.02 | jbot | somebody said drumkilla was the Asterisk v1.0-stable maintainer. ph33r him. |
17:53.02 | bugbot | drumkilla is assigned M2338, M3154, M3758, M3857, M3320, M3012, M2140, M2790, M2983, M3979, M3989, M1595, M3733, M2968, M3977, M2755, M3150, M2662, M3188, M2669 et al. and reported M2814, M4000, M3746, M3046, M3842, M3254, M3124, M3858, M3838, M3864, M3280, M3130, M3083, M3749, M3997, M3990, M3876, M3934, M3989. |
17:53.14 | drumkilla | soo mannnyy buuuggsss ... |
17:53.22 | CoaxD | ~coax |
17:53.23 | jbot | i heard coax is a wierdo who screws around with asterisk for fun |
17:53.23 | bugbot | coax is assigned nothing and reported nothing. |
17:53.47 | goldenear | Egonis, try this : in extension.conf : exten => 1800,1,Answer exten => 1800,2,MusicOnHold |
17:53.49 | CoaxD | ~botsnack |
17:53.49 | jbot | CoaxD: thanks |
17:53.49 | bugbot | botsnack is assigned nothing and reported nothing. |
17:53.55 | CoaxD | jbot: welcome |
17:53.58 | Dutts | can anyone hear me? =) |
17:54.11 | tzanger | ~~bugbot |
17:54.12 | bugbot | ~bugbot is assigned nothing and reported nothing. |
17:54.13 | jbot | ...but bugbot is already something else... |
17:54.15 | AntiPool | anyone runing asterisk on bsd ? |
17:54.19 | tzanger | damn |
17:54.21 | Qwell | umm... |
17:54.23 | tzanger | ~jbot |
17:54.24 | bugbot | jbot is assigned nothing and reported nothing. |
17:54.28 | jf_ | anyone can help me with sending call from asterisk to vonage number |
17:54.29 | tzanger | ~~jbot |
17:54.32 | Qwell | hmm |
17:54.33 | CoaxD | antipool: I DONT USE ASTERISK IN BED! |
17:54.41 | CoaxD | antipool: er. I misread you. :) :) :) |
17:54.45 | Qwell | tzanger: we got the same idea then, heh |
17:54.45 | AntiPool | hahaha |
17:54.52 | *** join/#asterisk MeTaBSD (metabsd@BlackBox.black4est.org) |
17:54.57 | MeTaBSD | hi all |
17:54.57 | tzanger | ~~jbot no, jbot is heading for a crash |
17:54.59 | jbot | I think you lost me on that one, tzanger |
17:54.59 | bugbot | ~jbot no, jbot is heading for a crash is assigned nothing and reported nothing. |
17:55.00 | jbot | bugbot: okay |
17:55.03 | tzanger | hehehe |
17:55.04 | Qwell | ~north |
17:55.06 | jbot | hmm... north is up today. |
17:55.08 | MeTaBSD | i have compile problem with asterisk-addons :( |
17:55.09 | *** part/#asterisk Dutts (~dutts@81.168.70.41) |
17:55.13 | bugbot | north is assigned nothing and reported M3457. |
17:55.13 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
17:55.14 | CoaxD | antipool: Asterisk on BSD works fine. Trouble is, zaptel for it aint that great, from what iv'e heard |
17:55.15 | MeTaBSD | app_addon_sql_mysql.c:162:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given |
17:55.21 | Dutts | hello guys can you hear me now? |
17:55.26 | CoaxD | antipool: Oh, i'm sure it does get better every day. But. |
17:55.35 | Qwell | tzanger: evil :p |
17:55.49 | MeTaBSD | i have mysql-server,clien,devel,shared-compat installed . |
17:55.50 | Qwell | Why did bugbot ignore me? heh |
17:55.53 | CoaxD | antipool: My advice is just to run it on linux |
17:56.08 | CoaxD | antipool: (Contrary to what some BSDers believe, it won't actually kill you to do so.) |
17:56.19 | AntiPool | CoaxD: linux - me no go, |
17:56.27 | CoaxD | ANtiPool: Guess you're screwed then :) |
17:56.28 | MeTaBSD | http://www.pastebin.com/271685 |
17:56.33 | AntiPool | CoaxD: i only wonder if it can work with isdn card |
17:56.35 | AntiPool | on bsd |
17:56.41 | MeTaBSD | can you help me :) |
17:56.42 | CoaxD | AntiPool: what, BSD? Probably not. |
17:56.55 | CoaxD | AntiPool: The APIs for talking to ISDN stuff are completely different between the OSs, i would think |
17:56.57 | Egonis | how do I install more music on hold? b/c it appears that 'default' is the only one that works, when I set to 'loud' the extension is busy |
17:57.10 | sivana | ~jobt |
17:57.11 | bugbot | jobt is assigned nothing and reported nothing. |
17:57.12 | tzanger | Qwell: I need a way to get jbot to and bugbot to talk to each other |
17:57.12 | sivana | ~jbot |
17:57.14 | jbot | somebody said jbot was heading for a crash is assigned nothing and reported nothing. |
17:57.14 | bugbot | jbot is assigned nothing and reported nothing. |
17:57.21 | Qwell | tzanger: yeah... |
17:57.24 | reallost1 | MetaBSD, did you check the zaptel-bsd mailing list? |
17:57.30 | Qwell | ~test |
17:57.31 | jbot | Test Passed! |
17:57.31 | bugbot | test is assigned nothing and reported nothing. |
17:57.32 | Qwell | ~test abc |
17:57.34 | jbot | Testing abc... ROM BASIC NOT FOUND$#$ |
17:57.34 | bugbot | test abc is assigned nothing and reported nothing. |
17:57.38 | Qwell | wtf |
17:57.44 | RoyK | jbot: lart bugbot |
17:57.45 | AntiPool | CoaxD: true, do you know anything about X100P cards ? |
17:57.46 | Qwell | How is that even there? |
17:57.55 | MeTaBSD | reallost1 im not on BSD im on Linux |
17:57.58 | tzanger | ~~jbot no, jbot is jbot no, jbot is going recursive |
17:57.59 | jbot | tzanger: I think you lost me on that one |
17:58.03 | Qwell | ~M2501 |
17:58.09 | bugbot | ~jbot no, jbot is jbot no, jbot is going recursive is assigned nothing and reported nothing. |
17:58.10 | jbot | bugbot: what are you talking about? |
17:58.10 | bugbot | M2501 is assigned nothing and reported nothing. |
17:58.16 | reallost1 | oh, antipool sorry. |
17:58.19 | Qwell | ~~M2501 |
17:58.20 | bugbot | ~M2501 is assigned nothing and reported nothing. |
17:58.21 | jbot | okay, bugbot |
17:58.25 | tzanger | hahaha |
17:58.34 | pgpkeys | as long as the card is supported it should work. I have freebsd, asterisk installed (not yet configured) which is in ports, and as i said so long as your ISDN card is supported it shoudl work |
17:58.36 | reallost1 | nm |
17:58.38 | pgpkeys | s/dl/ld/ |
17:58.40 | Qwell | wait, is that a valid username? heh |
17:58.40 | tzanger | jbot forget M2501 |
17:58.40 | jbot | tzanger: i forgot m2501 |
17:58.44 | Qwell | ~totallyfakeuser |
17:58.46 | bugbot | totallyfakeuser is assigned nothing and reported nothing. |
17:58.49 | AntiPool | reallost1: why sorry ? |
17:58.51 | Qwell | guess so |
17:58.56 | tzanger | Qwell: no it worked because you said ~~ |
17:59.08 | sivana | bugbot should need a msg word, like ~bug M2515 |
17:59.15 | sivana | s/msg/special |
17:59.15 | Qwell | sivana: yeah, likely |
17:59.22 | sivana | dont' ask how I screwed that one up |
17:59.45 | Dutts | if I downlaod the latest Asterisk v1.0.7 how do I upgrade my current install? |
18:00.08 | sivana | ~M2515 |
18:00.11 | bugbot | M2515 is assigned nothing and reported nothing. |
18:00.11 | reallost1 | antipool, I confused you with MeTaBSD in a conversation. |
18:00.25 | sivana | M2515 |
18:00.25 | bugbot | M2515 is a tweak bug that is closed (markster): [patch] cleaned up cdr_mysql.c. It was filed by sivana and was last updated on 01-10-05. http://bugs.digium.com/bug_view_page.php?bug_id=2515 |
18:00.31 | reallost1 | AntiPool, I'm running asterisk on BSD on several systems. |
18:00.34 | AngelGabriel | does anyone in here use SIPGATE? |
18:01.12 | AntiPool | reallost1: any expiriences with isdn on bsd ? i have only once isdn card, i'm poor and i want to try ip telephony :) |
18:01.30 | tzanger | I need an ISDN card that works in north america |
18:01.34 | tzanger | BRI of course |
18:01.40 | *** join/#asterisk Dutts (~dutts@81.168.70.41) |
18:01.50 | Dutts | hello guys |
18:02.04 | AngelGabriel | SIPGATE.CO.UK - they are a VoIP provider |
18:02.20 | Qwell | tzanger: mind a msg? heh |
18:02.39 | tzanger | depends on what it says |
18:02.40 | reallost1 | AntiPool, which ISDN card? |
18:02.44 | Qwell | tzanger: its funny. :) |
18:02.45 | tzanger | I have been known to shoot the messenger before |
18:03.35 | AntiPool | reallost1: truth is i forgot which one, and i lost dmesg, how can i list pci devices without scanpci ? |
18:03.48 | pgpkeys | pciconf |
18:04.09 | pgpkeys | this is a *bsd box correct? (net,free,open) |
18:04.50 | *** join/#asterisk Balu (~balu@foghorn.bartels-schoene.de) |
18:05.30 | AntiPool | pgpkeys: correct |
18:05.36 | pgpkeys | pciconf -lv |
18:05.42 | Balu | Hi everyone |
18:05.55 | Balu | Just a quick question... |
18:05.57 | *** join/#asterisk chris78 (~dg1nsw@saturn2.franken.de) |
18:06.06 | pgpkeys | questions are never quick |
18:06.12 | Balu | hm |
18:06.19 | Balu | Your answer is hopefully ;) |
18:06.23 | Balu | lemme phrase it 8) |
18:06.28 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
18:06.50 | *** join/#asterisk WGFreewill (~WGFreewil@24-75-221-174.miamfl.adelphia.net) |
18:06.58 | *** part/#asterisk eidolon (~eido@seawall.homeport.org) |
18:07.14 | Balu | I am one of the guys who likes to start from scratch, so I thought to clean /etc/asterisk (containing the debian examples) and start with a clean sip.conf and extensions.conf |
18:07.26 | Balu | is that ok or will I miss something absolutely needed? |
18:08.31 | Balu | asterisk.conf probably :) |
18:08.34 | pgpkeys | if you have to ask that then you've not read the manual or any of the docs that come with it. claiming 'i like to start from scratch' is a little over the top |
18:08.45 | AntiPool | reallost1: i have HFC-S PCI A Cologne Chip |
18:08.52 | AntiPool | reallost1: ever heard of it ? |
18:09.02 | Balu | the problem with the manuals and example config files is that they are cluttered with things I don't need |
18:09.07 | Balu | :) |
18:09.09 | tzanger | AntiPool: wow I bet that card stinks |
18:09.18 | pgpkeys | then ignore the things you don't need |
18:09.31 | AntiPool | tzanger: why ? they have support for linux on page :) |
18:09.33 | Balu | I've red some docs and tutorials and now think I can start with my own configs |
18:09.58 | tzanger | AntiPool: nevermind |
18:10.00 | chris78 | Balu: i started from scratch as well .. like noted in the asterisk-documentation-project .. its a good way i think |
18:10.19 | AntiPool | tzanger: but probably sux elephant's ass in real life :) |
18:10.20 | Balu | pgpkeys: But then I will not know if my stuff is not working because it conflicts with some not needed example in some file I don't know yet |
18:10.24 | pgpkeys | you still need to read the manual at the very least from end to end |
18:10.29 | Balu | of cours |
18:10.30 | Balu | e |
18:10.36 | Signuts | pgpkeys, end to end is easy |
18:10.41 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
18:10.47 | pgpkeys | Signuts: hehe |
18:10.52 | chris78 | Balu: then step by step go through and taking look at the example-configs helped me understand |
18:10.52 | pgpkeys | ok, start to finish ;) |
18:10.56 | Signuts | :) |
18:11.15 | AntiPool | tzanger: http://www.colognechip.com/isdn/controllers/hfc-pci-a.gif <- diagram of that card maybe you can tell if sucks ? |
18:11.30 | tzanger | AntiPool: I didn't say it sucked, I said it stunk |
18:11.39 | pgpkeys | chris78: excellent comment. |
18:11.51 | Balu | asterisk is fairly chatty on problems as I've seen so far, so it will help me too |
18:12.17 | pgpkeys | i've built more LFS style deployments than I care to count in the last 10 years. even I read the docs from top to bottom and look through the example configs and try to fathom what's being done. |
18:12.29 | pgpkeys | it's really the only way to do it if you're a 'start from scratch' guy |
18:12.51 | Balu | I was very active in LFS up to V4 - some parts are from me :) |
18:13.01 | pgpkeys | Signuts: or would that be bottom to top as well, depending on the language? ;) |
18:13.15 | Balu | pgpkeys: if we are talking about the same LFS ;) (Linux From Scratch) |
18:13.21 | pgpkeys | i was also one of the LFS mirrors |
18:13.25 | pgpkeys | Balu: yes |
18:13.26 | Balu | :) |
18:13.27 | pgpkeys | very same |
18:13.38 | [Outcast] | how does the amd chips handle asterisk? |
18:13.48 | pgpkeys | i started out as one of the original beta testers, before there was a 1.0 ;) |
18:14.12 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
18:14.29 | pgpkeys | [Outcast]: well it would come down to the horsies you have. the cpu itself has no issues |
18:14.52 | Gand_DJ | ne1 here installed G729 on their * box. I just got a non-commercial license but can't find info for installing the ipp. :( |
18:15.01 | Balu | :) |
18:15.02 | pgpkeys | i have a duron 1.3GHz for a fbsd dev box with 1GB RAM. i've yet to see any app it can't handle or an app that can't handle the cpu |
18:15.18 | [Outcast] | pgpkeys: is there a proformance gain from using the amds at all? |
18:15.29 | pgpkeys | not really except for games. |
18:15.36 | [Outcast] | k |
18:15.43 | pgpkeys | which is why i happened to have an AMD around in the first place :) |
18:15.48 | Balu | pgpkeys: I've learned a lot from LFS even though I was a linux user for >7(?) years at that time :) |
18:15.59 | pgpkeys | my game server box became my development box when i got tired of gaming |
18:16.04 | WGFreewill | Gand_DJ: I have commercial g729 |
18:16.06 | WGFreewill | digium |
18:16.08 | WGFreewill | works great |
18:16.09 | PBXtech | Gand_DJ, non commerical? |
18:16.36 | Gand_DJ | Yeah, non commercial license from intel website |
18:16.36 | PBXtech | oh that hack, yea nevermind |
18:16.48 | pgpkeys | Balu: so did I. I started out with SLS Linux (Slackware's father) and followed along for several years. then LFS came along and even having worked in the industry for some time, I learned more working through the LFS than i had previously |
18:17.00 | pgpkeys | VERY good choice for those that really are into the technicals of distrib building |
18:17.13 | Balu | yep |
18:17.15 | DrWho17 | Outcast: my Opteron boxes destroy my Xeon boxes for SQL database serving |
18:17.23 | Sedorox | :-p |
18:17.32 | Balu | pgpkeys: I've created a Dreamcast-Linux with my LFS-knowledge |
18:17.43 | Balu | pgpkeys: :) |
18:17.49 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
18:18.03 | pgpkeys | cool. |
18:18.07 | Sedorox | thats alotta 2's |
18:18.15 | Grooby | wierd |
18:18.17 | pgpkeys | i should email gerard and say hello |
18:18.22 | Grooby | has anyone have problem with speex codec? |
18:18.28 | pgpkeys | he and chris were great. |
18:18.35 | Balu | yep |
18:18.49 | Balu | pgpkeys: I wonder if there still are the irc-servers |
18:18.57 | pgpkeys | probably |
18:19.38 | Balu | yep irc.linuxfromscratch.org |
18:19.41 | Balu | no Gerard though |
18:19.57 | pgpkeys | he never stayed connected |
18:20.05 | Balu | true |
18:20.07 | Balu | :) |
18:20.10 | pgpkeys | he's busy has hell. i wouldn't expect him to stay connected all the time |
18:20.33 | Balu | don't even know what he is doing now |
18:20.39 | Balu | he switched and moved a lot |
18:20.56 | pgpkeys | well his company gave him permission to write the book in the first place, gave him the down time to do it. |
18:21.21 | pgpkeys | so now that it's written and it's well into new renditions he's probably back working and offloaded the majority to individual maintainers |
18:21.33 | Balu | Wasn't he going/working to do some LinuxLab work? |
18:21.48 | Balu | no |
18:21.48 | pgpkeys | i don't know for sure. i hven't touched a linux in oh neigh on 2 years now. |
18:21.58 | pgpkeys | everything i've got is all *bsd these days |
18:22.14 | pgpkeys | so i don't know the goings on anymore like i used to with most of the linux projects |
18:22.34 | Balu | :) |
18:22.40 | tzanger | heh |
18:22.42 | Balu | No BSD FS? :) |
18:22.57 | pgpkeys | free, net, and open. only things i run |
18:23.08 | tzanger | lfs is pretty good for learning but if you're going to just copy and paste the text you ain't learning any more than watching hours of compiler output scroll by with gentoo |
18:23.15 | tzanger | I tried BSD |
18:23.18 | tzanger | I tried really hard to like it |
18:23.20 | Balu | :) |
18:23.25 | pgpkeys | well occasionally i handle a solaris box or two on contract but not much beyond that |
18:23.33 | pgpkeys | tzanger: exactly |
18:23.40 | Balu | The learning in LFS is if you help people when they have problems |
18:23.45 | tzanger | yup |
18:23.51 | tzanger | same as asterisk, but really same as anything |
18:23.53 | pgpkeys | Balu: the learning is if you take the LFS apart |
18:23.57 | pgpkeys | to see what it's doing. |
18:24.03 | tzanger | I used LFS to build small firewalls |
18:24.05 | pgpkeys | the passing ON of what you've learned is in the helping |
18:24.30 | tzanger | I had a firewall with ipsec, perl, iproute2 and an xmlrpc configuration server fit into a 16M CF |
18:24.32 | ManxPower | Gads! _., "i", and "h" is generating almost as many messages as the GPL stuff! |
18:24.34 | tzanger | uncompressed was like 40 |
18:24.39 | pgpkeys | there's some learning in the helping too but nothing like when you take something apart |
18:24.39 | tzanger | ManxPower: that is good though |
18:24.48 | Balu | I started using LFS because I got fed up with all distros |
18:24.50 | tzanger | it's like stuffing 10 pounds of shit in a 5 pound bag |
18:24.52 | Balu | not did what I liked |
18:25.03 | tzanger | Balu: I felt the same way about everything which is why I use slackware |
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18:25.06 | *** mode/#asterisk [+o jmhunter] by ChanServ |
18:25.19 | Balu | Now I am back to "grah - all that compiling, fixing errors, recompiling - I don't have that time!!1!" |
18:25.31 | tzanger | Balu: heh |
18:25.40 | tzanger | I am trying out suse 9.2 for a lug presentation next month |
18:25.43 | pino | tazanger: i'm going to try with the uClibc-based debian sooner or later |
18:25.44 | tzanger | it's an interesting experience |
18:25.54 | tzanger | pino: that's what I did with lfs... uclibc+busybox for most things |
18:26.04 | pgpkeys | the only distro i touch these days is debian. I've even let my RHCE lapse out, but that's for political and internal differences with Red Hat's dealing with things. nothing they've done so far invalidates my RHCE knowledge so i can easily do what i need to do on the redhat boxes i controct on |
18:26.08 | pgpkeys | err contract even |
18:26.08 | Balu | Yep, nice for workplaces and my fathers box :) |
18:26.21 | tzanger | I hate hate hate HATE debian |
18:26.36 | pgpkeys | then you hate one of the strongest if not THE strongest linux distribution going |
18:26.49 | tzanger | pgpkeys: debian is going nowhere fast, don't kid yourself |
18:26.51 | Balu | I am also a Debian dude, but they absolutely need to do something about their release cycles |
18:26.53 | pgpkeys | NO ONE does the level of testing and auditing on their distros that debian does |
18:26.55 | tzanger | redhat's got more pull than debian |
18:26.58 | Grooby | this is really wierd |
18:26.59 | pgpkeys | tzanger: you can think that if you want |
18:27.01 | tzanger | and I hate redhat too :-) |
18:27.03 | pino | tzanger: no flame intended but... i love it :) so i'm curious to know why... |
18:27.05 | Grooby | i am getting 1 way audio with iax |
18:27.10 | Grooby | and just start happening recently |
18:27.18 | Gand_DJ | is there a way to see what audio formats * supports for transcoding? |
18:27.20 | pgpkeys | anyways.. i'm not getting into a distro or old vs. new war |
18:27.24 | Gand_DJ | or what's installed |
18:27.25 | tzanger | pino: no no people can use whatever distro they please, but I really really hate debian |
18:27.26 | pgpkeys | so.. how bout that asterisk! |
18:27.49 | Balu | asterisk? |
18:27.55 | pino | tzanger: exactly, i was asking why *you* hate debian! :) |
18:28.02 | Balu | ah - the number of people in here let me think it is #debian :) |
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18:28.21 | tzanger | pino: I hate the politics behind debian, I hate how unless you stick to stable, it's unstable. I hate how old stable is, but I understand why |
18:28.32 | pgpkeys | tzanger: that's horse shit, but ok |
18:29.02 | Balu | pino: I am using debian on all servers too, but I hate all the backports.org-stuff in my sources.list |
18:29.03 | tzanger | I hate how everyone says how stable it is yet they all run unstable and end up butchering the package tree, yet debian is still somehow immune to it |
18:29.06 | ManxPower | tzanger, It's pretty obvious that _. is confusing to new users and makes things happen that they don't want. |
18:29.07 | pino | tzanger: ok, release cycles mostly :) |
18:29.09 | tzanger | pgpkeys: refute it if you like :-) |
18:29.18 | pgpkeys | debian's testing is far more stable than any other distro's stable release is. the versions are usually no more than 1 behind the current upstream version and that's to ensure that they have a known codebase to secure and audit from |
18:29.25 | tzanger | ManxPower: _. is confusing, period |
18:29.34 | tzanger | pgpkeys: -stable is not 1 rev behind |
18:29.39 | tzanger | -testing is not -stable |
18:29.44 | tzanger | and -unstable is not -stable |
18:29.58 | Qwell | Therefore, -unstable must be -testing |
18:30.01 | tzanger | haha |
18:30.04 | Balu | pgpkeys: no security support in testing and unstable |
18:30.04 | ManxPower | tzanger, You hate Debian? I KNEW there was a reason I liked you. 8-) |
18:30.09 | tzanger | ManxPower: :-) |
18:30.11 | pgpkeys | that's stable, that's the official release. and the debian project's goal isn't to be latest and greatest. it's to build THE most stable distribution possible for administrators |
18:30.18 | tzanger | I mean yes, if everything you want is in -stable then it is *amazing* |
18:30.24 | tzanger | if you can put up with the politics |
18:30.29 | pgpkeys | Balu: unstable there is none |
18:30.34 | pgpkeys | but testing most assuredly does |
18:30.37 | tzanger | pgpkeys: I am not arguing that -stable is not stable |
18:30.49 | ManxPower | tzanger, if "i" worked as people expect it would not be as much of an issue. |
18:30.51 | tzanger | pgpkeys: as I said, if everything you want is in the -stable tree, you have a very well tested and very stable distribution |
18:30.57 | tzanger | ManxPower: I intend on making 'i' work as intended |
18:31.04 | pgpkeys | it's 1 day behind stable and that's because stable is an official release. they have to get it out for the official before they start on the unofficial |
18:31.21 | ManxPower | I think "i" should be fixed to work as expected and then remove _. as a valid pattern. Require at least 1 X or Z or N before the . |
18:31.38 | tzanger | nah accept _. still but it does not match any of the oshiat extensions |
18:31.56 | pgpkeys | tzanger: well considering that debian has almost 3 times the number of packages available than most of the other distros have available i'd say you'd have a VERY hard time not fiding something in stable |
18:32.03 | tzanger | pgpkeys: bullshit |
18:32.07 | tzanger | utter, complete bullshit |
18:32.13 | pgpkeys | tzanger: that's NOT bullshit,. that |
18:32.15 | pgpkeys | is FACT |
18:32.24 | Balu | pgpkeys: was it last year when all those ssh-problems came up? |
18:32.27 | tzanger | I have MANY utilities that do not appear in debina -stable or even -unstable |
18:32.44 | pgpkeys | Balu: that was ssh upstream which translated to ALL the distros, not to debian alone |
18:32.49 | ManxPower | pgpkeys, Um, I want ease of management, reasonably up to date packages, package format supported by many vendors. Mandrake does that for me. |
18:32.57 | Gand_DJ | how do you force linux to delete a directory that is not empty? |
18:33.01 | *** join/#asterisk DeeJayTwo (~deejay2@office.abi.ca) |
18:33.03 | pgpkeys | rm -rf |
18:33.08 | DeeJayTwo | hi guys.. |
18:33.16 | Egonis | Gand_DJ: also, to see WHAT you are deleting... ls .* |
18:33.23 | Balu | pgpkeys: I switched to a different ssh software (not sure what it's name was, need to look up), which had a bug too (found a few days later :-) |
18:33.26 | DeeJayTwo | Has anybody tried to use postgresql with odbc? |
18:33.41 | pgpkeys | ManxPower: apt is the #1 package management system out there which is why redhat supported so heavily porting apt to handle RPMs |
18:33.42 | Balu | pgpkeys: that was fixed in upstream, but not in the stable packages after two months |
18:33.55 | pgpkeys | as for ease of management debian/rules is as easy as hell |
18:34.00 | Gand_DJ | thx |
18:34.13 | Balu | pgpkeys: I contacted security-mailinglist and they told me there are problems backporting the stuff :-( |
18:34.28 | Balu | pgpkeys: not sure if it was ever fixed... |
18:34.49 | pgpkeys | Balu: yes, backporting an upstream issue that's filtered into over 12 architectures (which is more support for various arches than any otehr distro has) is not exactly easy |
18:35.02 | pgpkeys | s/otehr/other/ |
18:35.16 | tzanger | pgpkeys: psi (jabber client) is in stable, but it's quite a bit more than "one rev behind". openswan (ipsec gateway). unstable only. postgresql. hardly one rev behind. MANY revs behind. xen - unstable only. |
18:35.20 | tzanger | need more examples? |
18:35.33 | Balu | pgpkeys: of course, but it gave me a bad feeling all over... |
18:35.37 | Balu | anyway |
18:35.39 | pgpkeys | tzanger: you did exactly what i expected you to do, you ignore the keyword in my comment |
18:35.40 | ManxPower | pgpkeys, Cite your source. |
18:35.44 | pgpkeys | i said USUALLY |
18:35.47 | tzanger | pgpkeys: what was the keyword in your comment? |
18:35.51 | tzanger | usually what |
18:35.55 | pgpkeys | ManxPower: I was ON the project. i AM onje of the sources |
18:35.56 | pgpkeys | :) |
18:36.10 | pgpkeys | i said USUALLY one revision behind |
18:36.12 | tzanger | How can you tell a tough lesbian bar? ...Even the pool table doesn't have balls. |
18:36.16 | tzanger | hahahaha |
18:36.16 | ManxPower | pgpkeys, I mean, cite an OBJECTIVE source. |
18:36.19 | Gand_DJ | I noticed that for manitoba canada, all voip providers have access to 204-480 area only.. where would I look to getting ability to resell DID for that area also |
18:36.23 | pgpkeys | usually is not equal to subjective |
18:36.35 | pgpkeys | ManxPower: Umm redhat support what is it, 6 architectures? |
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18:36.49 | pgpkeys | mandrake supports the same amount as does suse |
18:36.53 | pgpkeys | same with slackware |
18:37.10 | pgpkeys | look at the stable release support of number of architectures |
18:37.18 | Balu | pgpkeys: and the debian people are thinking to reduce all their supported architectures |
18:37.21 | Balu | :) |
18:37.23 | sivana | lol |
18:37.27 | *** join/#asterisk corlis (~corlis@HSI-KBW-082-212-051-230.hsi.kabelbw.de) |
18:37.30 | pgpkeys | if that doesn't constitute subjective then i don't know what to tell you |
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18:37.47 | corlis | heyas |
18:37.52 | pgpkeys | Balu: on the more difficult ports like sparc which even redhat had issues with porting to. |
18:37.58 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
18:38.03 | tzanger | pgpkeys: the difference is that on slackware, building from source does not fuck up your package manager |
18:38.07 | Balu | Ok, stop bashing distros now |
18:38.08 | Balu | :) |
18:38.15 | durex | hello folks |
18:38.19 | pgpkeys | which people are naming as the competitor (who i also worked for) so I'll speak on that score as well :) |
18:38.21 | tzanger | pgpkeys: that's the EXACT reason I don't fuck around with these "dep tracking" distros... too much fucking work |
18:38.27 | *** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca) |
18:38.48 | durex | it´s my first time I instaled Asterisk. I have two computers with X-lite installed in the same network thank asterisk, and I wanna make them talk with each other |
18:38.51 | durex | does anybody can help me? |
18:39.15 | jf_ | is it possible to remove the girl saying in the voicemail please leave your message after, hangup or press the # key, by a simple beep |
18:39.16 | pgpkeys | tzanger: hehe. well trying to mix source and pkgs is always hell on wheels |
18:39.21 | corlis | durex: you been googling around and read that manymany examples? |
18:39.28 | tzanger | my main main main GNU/beef with GNU/debian is GNU/the GNU/politics |
18:39.30 | BuckRogers | durex you need to set up a sip config and extenctions |
18:39.33 | Balu | gtg dudes - I will come back if I totally ruined my ACFS (Asterisk Configuration From Scratch) :) |
18:39.40 | DrWho17 | slackware has a package manager? |
18:39.42 | tzanger | pgpkeys: exactly. with slackware it's trivial |
18:39.46 | tzanger | DrWho17: yup |
18:39.52 | durex | corlis yes... and I got the following error in Xlite: Call failed: 408 Timeout |
18:39.54 | MeTaBSD | its ok my problem solve :) |
18:40.01 | MeTaBSD | but i have other question :) |
18:40.01 | pgpkeys | tzanger: it's only trivial because slackware by default doesn't exactly have package management |
18:40.02 | MeTaBSD | Asterisk RealTime Voicemail |
18:40.12 | Balu | later |
18:40.13 | DrWho17 | hah, I ditched slackware 6-7 years ago, because they were so far behind |
18:40.14 | MeTaBSD | where i specify the Privilege user |
18:40.16 | corlis | durex: have you checked that the firewall is open/ok? |
18:40.28 | durex | corlis yes... it's ok |
18:40.30 | tzanger | pgpkeys: untrue. slackware has great simple package management. Slackware does not have DEPENDENCY TRACKING which is the problem with every other package manasgement system out there... they try too damned hard |
18:40.59 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
18:41.06 | DrWho17 | moved to that new Redhat thing, with glibc2 and rpm |
18:41.10 | pgpkeys | tzanger: ok, so it's cool when slackawre let's you overwrite libperl.so by some perl related package without checking to see if it's already installed |
18:41.11 | pgpkeys | cool |
18:41.27 | durex | corlis sorry... I think really could be a firewall problem... let me see... |
18:41.28 | tzanger | pgpkeys: dep tracking is 90% solved with ldd and a search of the manifest. debian (and the others) try for 100% and they all fail because if something's not in the package tree and you need it, you're fucked... now you have the package manager the disto has and you have the one you've now set up in your head. thanks but no thanks |
18:41.39 | Gand_DJ | Is there a way to change a peer / registry entry from being unmonitored to monitored? |
18:41.47 | tzanger | pgpkeys: cite me a real world example, not some contrived bullshit one |
18:42.04 | jf_ | is it possible to remove the girl saying in the voicemail please leave your message after, hangup or press the # key, by a simple beep |
18:42.07 | DrWho17 | tzanger: mysqlclient, glibc version |
18:42.10 | pgpkeys | i AM citing real world. you just choose not to let the possibility enter into YOUR world |
18:42.12 | pgpkeys | :) |
18:42.33 | tzanger | pgpkeys: hardly. I do a LOT of work with perl and never have I had libperl.so overwritten. Your example is contrived |
18:42.39 | tzanger | DrWho17: huh? |
18:42.44 | DrWho17 | yea perl, eek |
18:42.47 | pgpkeys | it's an example of a VERY real world possibility |
18:42.52 | tzanger | pgpkeys: no it's not |
18:42.57 | pgpkeys | ok, if you say so. |
18:42.59 | tzanger | pgpkeys: name a perl module that creates libperl.so |
18:43.01 | tzanger | go ahead |
18:43.06 | pgpkeys | i didn't say creates it |
18:43.28 | pgpkeys | now you're dodging the issue in an attempt to prove your disaffection |
18:43.30 | pgpkeys | i won't play that game |
18:43.30 | tzanger | pgpkeys: you actually give me great ammunition -- perl has its own package management: CPAN. If Debian doesn't have a package for a specific perl module you need, now you're dealing with CPAN too |
18:43.36 | DrWho17 | well, don't mix and match that is for sure |
18:43.42 | Grooby | hmmmm |
18:43.54 | DrWho17 | don't compile and install via the distro package system |
18:43.54 | tzanger | pgpkeys: I'm not dodging anything, I am calling bullshit. I have been working with slackware and perl for 7+ years... I have NEVER had a perl module overwrite libperl.so. |
18:43.55 | pgpkeys | *sigh* |
18:44.02 | MeTaBSD | --> Asterisk RealTime Voicemail i configure all but where i specify the username and password for databases and table privilege |
18:44.11 | tzanger | pgpkeys: that's like me saying I can create a deb package in -unstable that does the same... your point is meaningless |
18:44.12 | Grooby | what could cause the suddent change from working setup to only 1 way sound? |
18:44.20 | pgpkeys | tzanger: and I'ev been working on various distros for neigh on 10 years and this is a VERY real world situation. |
18:44.20 | DrWho17 | MeTaBSD: in extconfig |
18:44.25 | pgpkeys | so think as you wish. |
18:44.41 | tzanger | pgpkeys: again... where in the blue fuck do you get a perl module that overwrites libperl.so? Any distro |
18:44.43 | pgpkeys | i suggest we stop bashing at each other now because we are boviously in the opposition camp for each other's thoughts |
18:44.45 | DrWho17 | extconfig.conf |
18:44.53 | corlis | Anyone here could help me with this, when i enable CAPI debug: "found capi with omsn = 123456", next is: "Segmentation fault" |
18:44.59 | tzanger | pgpkeys: I just want a real world example, not a "what if" that can happen on any distro |
18:45.01 | pgpkeys | i used the libperl.so as an example of a potential |
18:45.06 | pgpkeys | people KNOW what libperl.so is |
18:45.36 | tzanger | pgpkeys: but the same thing cna happen in debian |
18:45.37 | DrWho17 | I had problems with mysqlclient and dbi yesterday |
18:45.41 | pgpkeys | tzanger: you CAN'T overwrite libperl.so on debian via any known package unless you --force |
18:45.48 | MeTaBSD | DrWho17 voicemail => mysql(methode),asterisk(DB),voicemail_users(Table) but where is the username and password ? |
18:46.03 | DrWho17 | updating mysql from source, broke the rpm installed perl DBD modules |
18:46.10 | tzanger | pgpkeys: and you can't do that on slackware without a bad package either... I fail to see what makes Debian better in this particular example? |
18:47.02 | jeffik | all: using *@home, need to add a line to extensions.conf to allow access to long on to voice mail by pressing * during greeting |
18:47.02 | DrWho17 | MeTaBSD: well, that's not how I access the voicemail users database from within asterisk |
18:47.16 | *** join/#asterisk AlexCeli (~Alex@200.37.85.95) |
18:47.19 | pgpkeys | tzanger: the last slackware i used (7 iirc) did not do adequate checks of versioning to ensure that proper care was taken to stop the overwrite. it was automatic assumption that it was ok TO overwrite from another package. |
18:47.26 | pgpkeys | and THAT is intolerable |
18:47.31 | DrWho17 | I just store the mailbox information in a table, and access it just like a normal extensions |
18:47.47 | tzanger | pgpkeys: yes we are on opposing sides... as I said, if -stable has exactly everything you want, it is *GREAT*. It really and truly is... but at least for my installations, it doesn't... and -testing and -unstable are no better than no package manager in my opinion because you either end up breaking the dep tree by stepping around it to get what you want or end up keeping two dep trackers around |
18:48.12 | jeffik | DrWho17: need to call in from outside |
18:48.18 | pgpkeys | why not make packages and submit if the ones you want are missing? |
18:48.19 | tzanger | pgpkeys: that's correct; slackware 10.1 does that too. you say "Installpkg somepkg.tgz" and it will, and anything in it will overwrite anything that was on the system before |
18:48.30 | tzanger | pgpkeys: that has a VERY simple answer. |
18:48.49 | pgpkeys | you don't have to maintain it, just submit it and ask the project to assign a maintainer |
18:49.01 | pgpkeys | if it has intrinsic value they'll do it |
18:49.04 | DrWho17 | jeffik: add it to your context to allow the * |
18:49.07 | *** join/#asterisk barmal (~1@adsl-19-109-17.asm.bellsouth.net) |
18:49.28 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
18:49.29 | DrWho17 | exten => a,1,VoiceMailMain(${CALLEDTO}) ; If they press *, send the user into Voic$ |
18:49.29 | DrWho17 | exten => a,2,Hangup |
18:49.44 | tzanger | pgpkeys: 1) I must create a patch to GNU/ify the entire documentation tree. 2) I must include in that patch moving the document tree to where Debian prefers it. 3) I must write proper startup/shutdown scripts and 4) I must now MAINTAIN that abomination since a) the software author will never accept the debian-ified setup as "stock" and b) nobody else will take it over because it's such a holy shit pile of work to keep up :-) |
18:49.44 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
18:49.58 | DrWho17 | just tack those two lines to the end of your voicemail context |
18:50.07 | MeTaBSD | DrWho17 i dont need privilege ? |
18:50.12 | *** join/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
18:50.13 | jeffik | ok i'll try it |
18:50.17 | tzanger | pgpkeys: I had no idea that someone else would accept a paackage maintenence job from lil ole me :-) |
18:50.21 | barmal | if Voicemail was left then do something. Is there something that tells you the voicemail has been left? |
18:50.28 | DrWho17 | MeTaBSD: you specify the priviledges in extconfig.conf |
18:50.31 | corlis | What should a normal extension look like for a call from a IAX-client to an isdn? |
18:50.44 | pgpkeys | tzanger: i think we'd both agree that the only 100% way to manage something is to remove package management completely, move the management back to the admin and let him face the tedious task of version control and overwrites. packages were simply meant to lower the level of time-consumption the admin had to face. |
18:50.44 | tzanger | maintaining slackware packages is dead simple in comparison |
18:51.17 | MeTaBSD | DrWho17 how |
18:51.17 | DrWho17 | pgpkeys: statically link everything !!! |
18:51.35 | pgpkeys | tzanger: again, as i said create the package and submit. if it has value (as in your not the only one, or only a small subset of the userbase would actually USE teh damn thing) they will usually assign a maintainer |
18:51.45 | DrWho17 | MeTaBSD: have you read the realtime instructions? They are very clear |
18:51.56 | pgpkeys | DrWho17: hehe ok, you just shot your memory usage up by probably 70% |
18:51.57 | tzanger | pgpkeys: correct -- slackware packages are easy to maintain and work with but you have to have a bit of clue to be able to use them effectively. What every other distro (lfs and gentoo aside) do is lower the level of clue required... it's a noble and laudable goal but it ties my hands as an admin |
18:51.59 | barmal | is there anything what returns anything if voicemail has been left? |
18:51.59 | MeTaBSD | Copy asterisk-addons/configs/res_mysql.conf.sample to /etc/asterisk/res_mysql.conf |
18:52.00 | DrWho17 | extconfig.conf.sample is probably in your default asterisk install as well |
18:52.04 | MeTaBSD | i think i find |
18:52.15 | pgpkeys | (arbitrary number to show the extreme increase in memory you just caused) |
18:52.26 | *** join/#asterisk nexIAX (~logger@telux.net) |
18:52.28 | nexIAX | <PROTECTED> |
18:52.29 | tzanger | pgpkeys: as I siad I had no idea you could do that, that reduces my number of arguments against debian's package system a little. :-) |
18:52.37 | pgpkeys | hehe |
18:53.03 | DrWho17 | MeTaBSD: well, if you are using the mysql you specify database user/pass/host in res_config_mysql.conf |
18:53.11 | pgpkeys | there are tons of maintainers in the queue looking for packages. if you did NOT find at least one looking for something I'd be truly suprised |
18:53.22 | DrWho17 | but you specify how to get to your database in extconfig.conf |
18:53.25 | jeffik | DrWho17: I will try it thank yuo |
18:53.31 | FLeiXiuS | Does asterisk come with a SIP server? |
18:53.33 | tzanger | ManxPower: quit fucking with bkw, that's hilarious :-) |
18:53.57 | *** part/#asterisk yertle (yertle@ip68-6-98-122.sb.sd.cox.net) |
18:53.57 | *** join/#asterisk TEKjacob (~chris@70-32-21-41.frdrmd.adelphia.net) |
18:54.08 | TEKjacob | Happy Friday! |
18:54.08 | barmal | gotoif (Voicemail has been left = true?3:2) how can I use it? does anybody know? |
18:54.36 | MeTaBSD | ok |
18:54.41 | *** join/#asterisk nexIAX (~logger@telux.net) |
18:54.42 | nexIAX | <PROTECTED> |
18:54.43 | pgpkeys | tzanger: anyways, it *has* been clueful and enjoyable speaking with you. need to make some coffee and step into an area *I* have no experience in, asterisk setup :) |
18:54.46 | tzanger | barmal: you can't |
18:54.52 | MeTaBSD | i dont have the good version ... i download the asterisk-addons 1.07 |
18:54.55 | corlis | TEKjacob: Happy Friday? with asterisk segfaulting here? *sniff* no |
18:54.56 | TEKjacob | Oh wise ones... Do I need to use a T1 Cross over or straight cable to go from the dmarc to the Digium T1 card? |
18:55.13 | DrWho17 | straight through all wires wired |
18:55.15 | tzanger | pgpkeys: :-) I enjoy flaming as much as anyone but this wasn't a flame (at least I hope you didn't think it was) |
18:55.43 | pgpkeys | naww, i've met far rougher. you have clue, whichis always good. i enjoy clue so long as it's clue and not clue+ego. |
18:55.46 | TEKjacob | groovy thanks folks |
18:56.03 | tzanger | pgpkeys: yup... I have no ego when it comes to this stuff, it's just what I've found to work (or not work) over the years |
18:56.07 | pgpkeys | i can always debate with clue. his brother ego has to go though. which (i think) neither of us brought to the party |
18:56.12 | tzanger | I didn't write the slackware package manager so none of my ego's in it |
18:56.20 | pgpkeys | hehe |
18:56.22 | tzanger | nope it was a good (heated) debate |
18:56.39 | pgpkeys | little bit of heat always makes the meat taste better :) |
18:56.44 | barmal | tzanger: thre is no way to do the function when the voicemail has been left and user hangs up * calls back to given number? |
18:56.50 | *** part/#asterisk jmhunter (~jacob@wire3-215.razzolink.com) |
18:56.59 | tzanger | barmal: of course, but it doesn't work with just asterisk |
18:57.03 | tzanger | you need a cron job and a shell script |
18:57.13 | DrWho17 | well |
18:57.13 | tzanger | because the voicemail will only go on properly if they hit # and 1 to accept the message |
18:57.16 | tzanger | which is bullshit |
18:57.17 | pgpkeys | ok, coffee time. wake up fully enough to start laying down this config |
18:57.17 | tzanger | been there, done that |
18:57.29 | tzanger | it's far easier with a shell script to watch for new messages and copy a .call file over |
18:58.36 | barmal | damm the problem is I am not good with shell script. Do you know if thre is any example would appreciate man |
18:58.47 | tzanger | barmal: yeah I have mine |
18:58.50 | tzanger | it ain't pretty bu tit works |
18:59.18 | barmal | can you share please... |
18:59.29 | tzanger | yueah just looking it up |
18:59.31 | barmal | litcomp@bellsouth.net |
18:59.54 | *** join/#asterisk In-Side (~Lowgitek@es-217-129-30-41.netvisao.pt) |
18:59.55 | In-Side | hi |
18:59.59 | barmal | hi |
19:00.01 | cypromis | lo |
19:00.37 | In-Side | anybody knows what that means? pr 15 20:00:05 WARNING[1527]: chan_sip.c:603 __sip_xmit: sip_xmit of 0x812b21c (len 414) to operator_ip returned -1: Bad file descriptor |
19:04.01 | In-Side | anybody ? |
19:04.18 | In-Side | I have no clue what hell is that |
19:04.20 | *** join/#asterisk mbranca_home (~matteo@host-84-222-7-10.cust-adsl.tiscali.it) |
19:04.56 | tzanger | barmal: http://pastebin.ca/9612 |
19:05.00 | *** part/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
19:05.09 | tzanger | that's my messy callback script; it's called every 5 minutes from crontab |
19:05.29 | *** join/#asterisk redG (~nik@67.107.241.3.ptr.us.xo.net) |
19:05.49 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
19:05.56 | In-Side | tzanger: can you help me ? |
19:06.57 | *** join/#asterisk redG ([U2FsdGVkX@67.51.185.15) |
19:06.59 | redG | <PROTECTED> |
19:07.17 | tzanger | bad file descriptor? weird |
19:08.02 | ManxPower | WOW! Useful suggestions to _. and h |
19:08.59 | tzanger | ManxPower: which one |
19:09.11 | ManxPower | http://bugs.digium.com/bug_view_page.php?bug_id=0004036 |
19:09.24 | tzanger | yeah |
19:09.32 | In-Side | let take a look |
19:09.35 | In-Side | damn message ... |
19:09.35 | ManxPower | tzanger, info about proper use of _. in extensions.conf.sample and add a URL to the warning message to the proper use of _. |
19:09.39 | In-Side | very unusefull |
19:09.40 | tzanger | ManxPower: uh 4036 is about parking |
19:10.22 | ManxPower | tzanger, I pasted that before I saw your "whoich one" |
19:10.27 | In-Side | the url is for me ? |
19:10.27 | tzanger | ManxPower: 4038 is the one |
19:10.37 | ManxPower | my paste for 4036 is just desperation to get it fixed. |
19:10.42 | ManxPower | I'll post a bounty if |
19:10.44 | ManxPower | i have to. |
19:10.59 | ManxPower | We go live with an office that will use a lot of parking on Friday |
19:11.09 | tzanger | Corydon-w: wow that was fast |
19:11.17 | tzanger | is it easy to do that for SIP and IAX too? |
19:12.18 | In-Side | IU just got franzy it that |
19:12.35 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp64197.qc.sympatico.ca) |
19:12.40 | In-Side | my * don't get registered in my sip provider |
19:13.13 | ManxPower | EGADS! I have a negative karma! |
19:13.24 | In-Side | how can i set a realm on my resister |
19:13.47 | In-Side | I trying with realm=name.realm in providerconfiguration |
19:14.19 | In-Side | and it still persist to use the host name after @ as realm in register in provider how can I set another one ? |
19:15.23 | *** join/#asterisk jmacz (~jmacz@63.245.86.225) |
19:16.12 | *** join/#asterisk darby_t (~tom@182-tor-6.acn.waw.pl) |
19:16.24 | corlis | any isdn-god here, that could help me with my lill problem calling out? |
19:16.38 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
19:17.15 | Corydon-w | tzanger: dunno, haven't looked |
19:17.34 | corlis | or any c-guru/asterisk-guru here, that could help me to find out why my asterisk SEGFAULTS? |
19:17.38 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
19:17.43 | Corydon-w | That's the advantage of having poked around in the code for a long time... |
19:17.44 | WGFreewill | can anybody help with H.323 codec negotiation problem |
19:17.47 | corlis | woops, i hate caps :P |
19:17.52 | WGFreewill | Nortel is beating me about asterisk |
19:17.59 | *** join/#asterisk Hymie (hymie@L8R.NET) |
19:18.02 | tzanger | Corydon-w: I'm looking at it and I don't know if that's right |
19:18.08 | tzanger | wouldn't you use ast_canmatch_exten() ? |
19:18.11 | Hymie | hey guys, I seem to be getting a lot of problems with hard drive usage and clipping / noise in asterisk. |
19:18.16 | Hymie | I don't think it's purely interrupts, but I am monitoring each call... so if I have 8 or 9 calls occuring at once, there's some thrashing.. add a cp operation, and it seems that asterisk freaks out at the inability to dump the buffer fast enough |
19:18.20 | Hymie | anyone else notice behaviour like this? |
19:18.39 | In-Side | hey guys how can i set a realm in register ? |
19:18.42 | Nivex | <PROTECTED> |
19:18.45 | Corydon-w | tzanger: I'm looking for a single extension called "i". Why would I use a different function? |
19:18.52 | Corydon-w | tzanger: btw, have you tried it? |
19:18.59 | tzanger | Corydon-w: no I haven't tried it yet, it seems too simple |
19:19.01 | In-Side | my sip provider doesn't accept the host name as realm |
19:19.11 | tzanger | Corydon-w: you are looking for an 'i' exten but what will match it later on? |
19:19.16 | In-Side | how in register i can set a new realm ? |
19:19.24 | WGFreewill | http://bugs.digium.com/bug_view_page.php?bug_id=0003980 |
19:19.33 | FLeiXiuS | Does Asterisk already include SIP? |
19:19.34 | Corydon-w | tzanger: once you start the dialplan, the dialplan takes care of the invalid extension |
19:19.41 | tzanger | Corydon-w: ahhhhhhhhhhh |
19:19.42 | *** join/#asterisk easimon (~easimon@baghira.kawo2.RWTH-Aachen.DE) |
19:19.43 | corlis | FLeiXiuS: yes |
19:19.57 | In-Side | FLeiXiuS: see sip.conf |
19:20.02 | DaLion | !seen bkw_ |
19:20.02 | FLeiXiuS | corlis: It includes the SIP server also correct? |
19:20.09 | In-Side | FLeiXiuS: shure |
19:20.11 | Corydon-w | tzanger: that's the nice thing about how pbx.c's dialplan logic works |
19:20.22 | corlis | FLeiXiuS: yes, yes and yes |
19:20.33 | FLeiXiuS | lol cool :-p |
19:20.42 | Corydon-w | Because, in the words of that Ronco commercial, we can "set it and forget it" |
19:20.56 | In-Side | doesanybody knows how to set the realm at register in sip.conf? |
19:21.32 | eKo1 | realm=myrealm under [general] in sip.conf |
19:21.35 | tzanger | Corydon-w: ok, I did something simialr for chan_iax2.c |
19:21.41 | eKo1 | I think |
19:21.42 | tzanger | except I return CANEXIST |
19:21.48 | tzanger | testing now |
19:21.49 | In-Side | eKo1: no that is only for clinets |
19:21.50 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
19:21.55 | FLeiXiuS | I'm looking to just use Asterisk locally, without connecting to a VoIP provider. Would I need SER to do the routing? Or does Asterisk already deal with this. |
19:22.01 | In-Side | it doesn't uset it at register |
19:22.14 | Corydon-w | tzanger: I would test it myself, but I'm in the middle of a drastic change to manager.c |
19:22.15 | In-Side | FLeiXiuS: asterisk is all in one solution |
19:22.20 | In-Side | FLeiXiuS: is enough for you |
19:22.32 | tzanger | Corydon-w: :-) I will test it tonight |
19:23.22 | tzanger | bah got 3 active calls |
19:23.31 | ariel_ | FLeiXiuS, in Between 95% to 99% of the time you will never need to use SER with asterisk. |
19:23.48 | tzanger | er two now |
19:24.00 | ariel_ | Good afternoon everyone |
19:24.43 | FLeiXiuS | ariel_ and In-Side: Thank you. |
19:24.47 | corlis | Why does asterisk crash, when using: "exten => _0049XXXXXXXXXX,1,Dial(CAPI/12345678:${EXTEN},90)" |
19:25.30 | corlis | When i try to call out with my IAX-client, asterisk does a segmentation fault |
19:26.07 | ariel_ | corlis, seems like you have an issue with your drivers. Are you using CVS Head or stable? |
19:26.20 | corlis | cvs |
19:26.46 | ariel_ | corlis, make sure that your capi is the right one for cvs head and not stable. |
19:27.09 | corlis | using chan_capi 3.5, as it's the only one out there? |
19:27.52 | corlis | modified it a little, tho, as it won't compile with cvs |
19:28.20 | *** join/#asterisk drbrown (~chatzilla@65.121.240.182) |
19:28.28 | pino | corlis: there's a patch set for running chan_capi on cvs-head, AFAIK |
19:28.39 | foobos | corlis, depending on card, you can also run isdn with chan_misdn http://www.beronet.com/?PageID=3017 |
19:29.17 | corlis | pino: i applied that patch, but it still was referring to a non-existant channel_pvt.h |
19:29.25 | DeeJayTwo | is there any stable asterisk version with "realtime" ? |
19:29.44 | corlis | foobos: hrm. gonna check that one out, thanks |
19:29.50 | pino | corlis: the one I have does not... yours is probably old |
19:31.01 | pino | http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 does no longer use channel_pvt.h. |
19:31.03 | ariel_ | DeeJayTwo, no |
19:31.11 | Hymie | hey guys, I seem to be getting a lot of problems with hard drive usage and clipping / noise in asterisk. I don't think it's purely interrupts, but I am monitoring each call... so if I have 8 or 9 calls occuring at once, there's some thrashing.. add a cp operation, and it seems that asterisk freaks out at the inability to dump the buffer fast enough |
19:31.17 | durex | does anybody can help me with asterisk and x-lite ??? |
19:31.21 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
19:31.38 | LoRez | Hymie: using scsi? |
19:31.43 | Hymie | LoRez: ide |
19:31.57 | corlis | pino: Hrm. then there went something wrong... gonna recheck that.... |
19:32.19 | ariel_ | durex ask a question lots of people here use both. |
19:32.55 | pino | Hymie: if I ran * on Linux with IDE disks, I'd try hdparm -u1 -3c -d1 on them. |
19:33.06 | barmal | tzanger: usage is exten => voicemail_callback.sh u101 ??? |
19:33.07 | pino | s/3c/c3/g |
19:33.36 | *** join/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
19:33.42 | tzanger | just 101 |
19:33.53 | barmal | ok |
19:34.42 | DrWho17 | DeeJayTwo: no |
19:36.40 | corlis | pino: weird, here it's still referring to channel_pvt.h.... |
19:38.22 | _Brian | anyone here ever play with AVST? |
19:38.28 | FLeiXiuS | Could I just use any ordinary phone adacpter from RJ11->RJ45(VoIP)? I was looking at the vonage one, but it says subscirption required. |
19:38.33 | pino | corlis: the applications do, but the channel does not |
19:38.49 | DeeJayTwo | DrWho17: So there's still no asterisk version able to be feeded dynamically from a SQL database? |
19:39.04 | DrWho17 | not the stable branch |
19:39.14 | DrWho17 | CVS has had it for quite some time |
19:39.14 | DeeJayTwo | ok... |
19:39.37 | DeeJayTwo | can we expect some "unstable" version to be technically stable? ;) |
19:39.40 | corlis | pino: hrm? but then "make install" will fail.... |
19:40.01 | DrWho17 | DeeJayTwo: well, stable is a new thing to asterisk, the CVS is pretty stable |
19:40.19 | pino | you just need chan_capi.so to compile OK with the patch and without further editing. then you have a starting point to investigate the crash! |
19:40.38 | DeeJayTwo | HEAD ? |
19:40.43 | DrWho17 | yes |
19:41.05 | DeeJayTwo | you'd build an mid-telco on it? |
19:41.06 | DrWho17 | I've always run out of CVS |
19:41.08 | DeeJayTwo | (1000 users) |
19:41.14 | *** join/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com) |
19:41.30 | DrWho17 | DeeJayTwo: well, like I said, it's only a new thing, that asterisk has had a "stable" branch |
19:41.45 | DeeJayTwo | ok |
19:42.07 | DrWho17 | I've got a device without a reload in 6 months |
19:42.14 | DrWho17 | running off CVS |
19:42.35 | DeeJayTwo | we built a management interface for "realtime"... |
19:42.37 | DrWho17 | and that was because I upgraded it |
19:42.42 | DaLion | man i ahad a sun box i didnt reboot for 790 DAYS |
19:42.47 | DeeJayTwo | we're gonna get our server monday...and we're gonna start testing in 1 week.. |
19:42.59 | DrWho17 | Daliion: well, I'm talking about an asterisk reload |
19:43.03 | corlis | pino: ok, and where do i have to put chan_capi.so? |
19:43.07 | DrWho17 | of course the machines have been up fine |
19:43.09 | DaLion | it could take load avg of 1500 before goign to a crawl.. in the good old 1995's days.. where porn sitres where huge ;) |
19:43.16 | DeeJayTwo | with Adtran TA750 channel banks.. |
19:43.18 | DeeJayTwo | for 250 users |
19:43.20 | DaLion | ah |
19:43.21 | DaLion | heheh |
19:43.23 | pino | usually, in /usr/lib/asterisk/modules/ |
19:43.26 | DaLion | that nice then |
19:43.36 | DrWho17 | DeeJayTwo: oh, we have some of those |
19:43.41 | DaLion | i think we reload asterisk each night at 5 am when 0 calls |
19:43.47 | DaLion | made a small script for that |
19:43.52 | DrWho17 | and some carrier access bits lying around |
19:43.58 | DaLion | sits andwaits til 0 calls then does whatevr |
19:44.07 | eKo1 | I restart * every day at 1:00 AM. |
19:44.13 | DaLion | 1am is busy here |
19:44.14 | DrWho17 | really, hrm |
19:44.19 | DaLion | yeah |
19:44.24 | *** join/#asterisk ChristianK (~Christian@p54A3FD7C.dip.t-dialin.net) |
19:44.26 | PBXtech | you do that in a cron job? |
19:44.37 | DrWho17 | why do you do that? |
19:44.37 | DaLion | no |
19:44.45 | eKo1 | It will restart regardless of how busy it is here. |
19:44.52 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
19:44.53 | corlis | pino: still the same: executes the dial, then segfaults |
19:45.16 | DaLion | i do a while look in bash .. test for lock file (means it restarted) then asterisk -rx 'show channels' | grep -a 'active'| gawk -F' ' '$1==0 {print "./reloadast.sh" }' |/bin/bash |
19:45.17 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
19:45.24 | eKo1 | My problem is, by the end of the day, * is full of zombie channels and the only way to get rid of them is using a restart. |
19:45.27 | DaLion | that reload is just what ever i want to do |
19:45.29 | corlis | gonna check out that misdn thing |
19:45.59 | *** join/#asterisk ves1820 (~root@o-254-108.hosts.cablelink.de) |
19:46.01 | DrWho17 | eKo1: really, what type of channels? |
19:46.20 | DaLion | eko u want it ? |
19:46.26 | *** part/#asterisk xai (~pasta@cpe-70-112-17-10.austin.res.rr.com) |
19:46.30 | eKo1 | SIP, which are related to my FXO gateways. |
19:46.37 | eKo1 | Stupid analog crap. |
19:46.40 | DaLion | u can test with showing print" I WOULD RELOAD" instead ofdoing anyshit |
19:46.51 | eKo1 | Otherwise, I would never reload or restart. |
19:46.55 | DrWho17 | eKo1: so they are losing connection or something? |
19:47.19 | eKo1 | Well, no but those zombie channels eat memory. |
19:47.20 | DrWho17 | I've got my asterisk boxes sipping over OC3/OC12/ethernet only |
19:47.24 | DrWho17 | stable ethernet links |
19:47.30 | eKo1 | Lucky you. |
19:47.42 | DrWho17 | and TDM inside, never noticed an issue |
19:47.57 | DrWho17 | TDM -> SIP |
19:48.05 | eKo1 | Are there any good FXO gateways out there? |
19:48.14 | DrWho17 | pretty high volume per machine, I never reload unless making a config change |
19:48.33 | *** join/#asterisk mbranca_home (~matteo@host-84-222-11-21.cust-adsl.tiscali.it) |
19:49.05 | DrWho17 | If I had to reload/restart services like that, I'd look for something else |
19:49.15 | eKo1 | Well, you should reload ONLY when you make config. changes. |
19:49.15 | DaLion | ??? |
19:49.35 | eKo1 | Well, I don't have other choices. |
19:49.43 | ves1820 | hi, has somewone got MWI working with CCM 4.2 ? |
19:49.43 | DrWho17 | ok |
19:49.47 | PBXtech | buy a HP server |
19:49.58 | ves1820 | yo |
19:50.00 | DaLion | hey .. yeah .. but a config change could be a simple as changing folow me .. or new password for vm |
19:50.01 | eKo1 | I'm at the mercy of these shitty analog gateways. |
19:50.01 | ChristianK | srt: how are you? :D |
19:50.54 | DrWho17 | Damin: yea, got most of those things in a MySQL table though |
19:51.03 | DrWho17 | DaLion rather |
19:51.10 | DaLion | yeah |
19:51.19 | DaLion | so whats the prob again ? |
19:51.32 | PBXtech | i want to play with that ds3 card |
19:51.45 | tzanger | Corydon-w: unfortunately the fix isn't that simple in chan_iax2 :-( |
19:51.51 | *** join/#asterisk netofsickcoder (~netofsick@cpe-24-170-74-115.stx.res.rr.com) |
19:53.22 | eKo1 | I don't get this. I ping my gateway from one machine and I get <10 ms ping times. I ping it from another machine and the ping times >100 ms. |
19:53.54 | *** join/#asterisk don_oles (~bill@pajaro.alfabank.kiev.ua) |
19:54.28 | tzanger | Corydon-w: oh wait, I think I found it |
19:54.35 | don_oles | hello coolhackers ;-) |
19:55.36 | don_oles | someone can tell me why this stuff uses 97% of CPU on my 2 Ghz FreeBSD box? |
19:55.49 | don_oles | .. when doing nothing???? |
19:56.00 | *** join/#asterisk bobessutio (~bobessuti@c-67-180-96-152.hsd1.ca.comcast.net) |
19:56.20 | Corydon-w | don_oles: because the FreeBSD port is a hack? |
19:56.38 | bobessutio | can you have one did spread across multiple origination providers? |
19:57.57 | barmal | tzanger: thanks something works only it sends a sms message but i'll firure out the rest, Thanks |
19:59.35 | sivana | jbot: no, sivana is one of the brightest stars out there, ok? :) |
19:59.36 | jbot | okay, sivana |
20:01.33 | corlis | hrm. i already hate my idea of using asterisk at my office, so i can receive calls wherever i have my laptop.... |
20:02.36 | corlis | first day that it works, and already 3 calls... although it's 10pm here |
20:02.59 | *** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca) |
20:03.42 | jf_ | is there any to make * dial faster on a zap channel, for instance if i use my sip phone and want to call over a zap channel it take like 4 seconds before start rigging |
20:05.14 | CoaxD | jf: It should start ringing immediately on the SIP phone and dial silently and bridge the SIP and the ZAP channel together |
20:05.25 | *** part/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
20:06.22 | tzanger | what the blue fuck |
20:06.59 | tzanger | Corydon-w: I've replaced every instance of ast_exists_extension() with one that checks for the original AND "i" and it still doesn't work |
20:07.35 | Corydon-w | tzanger: I haven't looked, so I don't know what to tell you |
20:07.36 | Qwell | tzanger: Whats the problem with i? I tend to ignore the list |
20:07.39 | jf_ | CoaxD: right now i dont think it does |
20:08.26 | jf_ | can i configure something |
20:08.39 | kairo | Reading, ok the asterisk is one gatekeeper, but it is a sip gk? I use the gnugk, one gk h.323 and I need sip on the momment. |
20:08.40 | CoaxD | jf: Hmm. no, it just does what it does |
20:08.49 | CoaxD | jf: But if yours is doing something different than mine, there's obviously a difference |
20:08.55 | tzanger | Qwell: read the list |
20:09.10 | Qwell | tzanger: Too much drama |
20:09.21 | tzanger | <PROTECTED> |
20:09.24 | tzanger | ooh |
20:09.24 | tzanger | getting closer |
20:09.43 | *** part/#asterisk bobessutio (~bobessuti@c-67-180-96-152.hsd1.ca.comcast.net) |
20:10.07 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
20:10.44 | TUplink | how do i set an umtamate timeout? |
20:11.07 | jf_ | CoaxD: does i ring right now how it take like 4 sec before |
20:11.53 | tzanger | ~google IAX2 RFC |
20:11.54 | bugbot | google IAX2 RFC is assigned nothing and reported nothing. |
20:12.02 | tzanger | no no no |
20:12.06 | tzanger | ~google IAX2 reference document |
20:12.07 | bugbot | google IAX2 reference document is assigned nothing and reported nothing. |
20:12.14 | tzanger | oh for fuck sakes |
20:12.53 | *** part/#asterisk DrRighteous (~DrRighteo@ool-182c867b.dyn.optonline.net) |
20:13.39 | tzanger | there we go |
20:13.39 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
20:13.42 | tzanger | ~google iax2 white paper |
20:13.48 | tzanger | ha |
20:13.50 | bugbot | google iax2 white paper is assigned nothing and reported nothing. |
20:15.36 | Hymie | hey guys, I seem to be getting a lot of problems with hard drive usage and clipping / noise in asterisk. I don't think it's purely interrupts, but I am monitoring each call... so if I have 8 or 9 calls occuring at once, there's some thrashing.. add a cp operation, and it seems that asterisk freaks out at the inability to dump the buffer fast enough |
20:16.10 | *** join/#asterisk Veryhot (~tho@adsl-69-109-159-239.dsl.sndg02.pacbell.net) |
20:16.33 | Veryhot | hi all, anyone using netlogic.net? |
20:17.38 | Veryhot | do they have good network for netlogic.net? |
20:17.44 | tzanger | Corydon-w: are you sure that check for 'i' will work? I see * falling back to 's' if the exten doesn't exist, not 'i' |
20:18.01 | jf_ | is there any to make * dial faster on a zap channel, for instance if i use my sip phone and want to call over a zap channel it take like 4 seconds before start rigging |
20:18.16 | Corydon-w | tzanger: but it DOESN'T fall back to s |
20:18.46 | tzanger | hmm |
20:18.46 | Corydon-w | s is only for if you don't have ANY extension when you start |
20:18.46 | tzanger | chan_iax2 is working very differently from zap |
20:18.50 | tzanger | Corydon-w: yeah, chan_iax2 is wrong |
20:19.00 | tzanger | it says "extension 5342432 doesn't exist, falling back to 's'" |
20:19.18 | Corydon-w | Interesting |
20:21.46 | CoaxD | The taxes are DONE, MAN |
20:22.09 | Corydon-w | CoaxD: You don't have your refund, already? |
20:22.15 | CoaxD | Corydon: I had to pay in $3k |
20:22.20 | Corydon-w | Ouch |
20:22.25 | CoaxD | Corydon: You think I'm gonna give 'em money BEFORE 4/15?? |
20:22.36 | Corydon-w | I'm thinking extension... |
20:22.45 | tzanger | Corydon-w: pbx.c around line 2250 |
20:22.48 | CoaxD | Corydon: Heh :) You can't get an extension for money |
20:22.54 | CoaxD | Corydon: You can get an extension for the RETURN |
20:22.56 | tzanger | I think it should check ast_strlen(c->exten) for nonzero |
20:23.00 | tzanger | and only drop to s if it is zero |
20:23.08 | tzanger | oterwise jump to 'i' |
20:23.14 | Corydon-w | You mean ast_strlen_zero |
20:23.47 | tzanger | yes |
20:23.48 | tzanger | :-) |
20:24.02 | Corydon-w | Go for it... |
20:24.17 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
20:24.30 | shmaltz | who overhere is an admin to the asterisk-users list? |
20:24.30 | PBXtech | im getting quite a few dropped IAX <->IAX calls over a lan (no internet calls) any ideas off the top of your head? |
20:24.33 | tzanger | what the hell |
20:24.34 | tzanger | <PROTECTED> |
20:24.48 | tzanger | copy "s" to c->exten for a size of the orignal exten length? |
20:24.55 | tzanger | that seems wrong |
20:24.57 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
20:25.48 | cp5 | it says don't copy more than what can fit in c->exten |
20:26.26 | *** join/#asterisk terracon (~tc@CPE0050da608e99-CM0012254076d6.cpe.net.cable.rogers.com) |
20:26.51 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
20:28.14 | Bentley | hi all - can anyone recommend a utility for measuring packet loss on a network? |
20:28.57 | riksta | good question |
20:29.25 | TUplink | packet loss.... packets dont get lost inles their is a colision |
20:29.34 | TUplink | rite |
20:29.50 | riksta | not just that |
20:29.51 | PatrickDK | tuplink, you wish |
20:29.52 | TUplink | ifconfig i belive will tell you Collisions |
20:29.56 | TUplink | ok... |
20:30.03 | TUplink | howels dose it die? |
20:30.10 | riksta | howels? lol |
20:30.14 | riksta | how else? |
20:30.16 | PatrickDK | most routers drop packets |
20:30.23 | PatrickDK | if a link is over 80% full |
20:30.31 | riksta | TUplink: what about when you receive more data than you can process? |
20:30.36 | PatrickDK | also, interference |
20:30.44 | PatrickDK | buffer overflows :) |
20:30.56 | TUplink | yea... true |
20:30.59 | PatrickDK | riksta, that doesn't happen too much now a days though |
20:31.16 | riksta | true |
20:31.25 | PatrickDK | I have some 3c501 cards, that can't do back toback packet reception :) |
20:31.37 | PatrickDK | on 10base-2 |
20:31.38 | Bentley | i was about to try out iperf: http://dast.nlanr.net/Projects/Iperf/iperfdocs_1.7.0.html |
20:31.45 | Bentley | thought i'd check here 1st tho |
20:31.59 | riksta | i actualy didnt know of any tools for it Bentley |
20:32.16 | riksta | i dunno if ntop does that kinda stuff |
20:32.45 | riksta | Bentley: i think you want smokeping |
20:34.00 | Bentley | looks interesting riksta |
20:35.03 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
20:45.20 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
20:47.43 | shmaltz | who overhere is an admin to the asterisk-users list? |
20:48.07 | Corydon-w | There are no mail admins |
20:48.21 | Corydon-w | Not any one person dedicated to the task, anyway |
20:48.50 | denon | hmm .. am I losing my mind here? I thought asterisk would playback .wav files by default as well .. |
20:48.55 | denon | its whining cause its not a gsm .. |
20:50.00 | *** join/#asterisk toyos (~gabri@216.90.111.178) |
20:50.15 | *** join/#asterisk h3x0r (Justino@ip70-180-167-6.lv.lv.cox.net) |
20:50.22 | h3x0r | wtf, ds3000p |
20:50.27 | toyos | How do I do IAX2? |
20:50.31 | h3x0r | this is a joke right |
20:50.38 | Qwell | h3x0r: no |
20:50.39 | toyos | nope |
20:50.43 | shmaltz | Corydon-w, someone unsubscribed me from the list |
20:50.53 | shmaltz | and when I try to resubscribe, it doesn't work |
20:51.00 | h3x0r | youd need like 8 machines to codec translate anyway |
20:51.01 | h3x0r | heh |
20:51.26 | h3x0r | maybe more |
20:51.33 | Qwell | h3x0r: 8 way opteron? :p |
20:51.52 | h3x0r | yeah right |
20:52.13 | h3x0r | it would cost more than a brand spanking new sonus, telica, apx 8000, anything |
20:52.19 | h3x0r | probably |
20:53.05 | Corydon-w | shmaltz: have you considered that your ISP may have instituted an overly aggressive spam policy that it dumping all your list traffic? |
20:53.21 | shmaltz | Corydon-w, no |
20:53.28 | shmaltz | 1. My ISP is Gmail |
20:53.50 | shmaltz | 2. I got a message from the list telling me that I'm unsubscibed |
20:53.51 | Corydon-w | And you've checked all your spam folders? |
20:54.00 | shmaltz | of course |
20:54.12 | Corydon-w | Well, try emailing someone at Digium |
20:54.19 | shmaltz | oh, I just got my confirm for resubscribing |
20:54.39 | Corydon-w | Could also have been too many bounces which got you unsubscribed |
20:55.14 | h3x0r | and the new IAXy looks like a chinese radar detector |
20:55.20 | shmaltz | what do ya mean too many bounces? |
20:55.53 | shmaltz | like NDRs? |
20:56.53 | Veryhot | anyone using Netlogic.net? |
20:57.19 | *** join/#asterisk jhowardPA (~jhoward@12.25.177.120) |
20:57.28 | jhowardPA | Hello people! |
20:58.11 | jhowardPA | riksta: I threw a copy of Xorcom Rapid on my box, rather than A@Home... working great now. |
20:58.26 | jhowardPA | riksta: sorry I didn't get back to you - I was out sick yesterday. |
20:58.36 | *** join/#asterisk greg_work (~greg@d221-73-240.commercial.cgocable.net) |
20:58.40 | Veryhot | jhowardpa: yeah I seem to like xorcom too |
20:58.43 | greg_work | what are the IAX2 and SIP ports? |
20:59.10 | shmaltz | greg, SIP is 5060 by default |
20:59.17 | jhowardPA | Anyone know why my hold music would be silent, though I''ve got mpg123 (0.59r) playing properly? |
20:59.18 | shmaltz | check iax.conf, and sip.conf |
20:59.19 | Veryhot | jhowardpa: A@home should be good in couple more version |
20:59.26 | greg_work | and iax is 4569/udp ? |
20:59.32 | jhowardPA | Veryhot: I hope so ;) |
20:59.57 | shmaltz | jhowardPA, any messages on the CLI? |
21:00.14 | Veryhot | jhoward: I have problem recording my menu on the A@home |
21:00.17 | jhowardPA | shmaltz: none - none at all. |
21:00.17 | *** join/#asterisk bah (048830696@AC869EBC.ipt.aol.com) |
21:00.39 | jhowardPA | Veryhot: I gave up on it. Made me sad. |
21:00.41 | shmaltz | jhowardPA, not evan telling you starting music or stopping music? |
21:01.03 | Veryhot | jhowardpa: then I have to do custom menu :) |
21:01.06 | jhowardPA | shmaltz: I believe it says it's starting on Asterisk startup.. double-checking... |
21:01.25 | *** join/#asterisk lohelle (~post@213.161.252.253) |
21:01.29 | shmaltz | jhowardPA, when you put someone on hold, is it syaing anything? |
21:01.54 | Veryhot | anyone know of a opensource CDR program? |
21:02.29 | jhowardPA | shmaltz: no messages regarding music at all. |
21:02.39 | Veryhot | for doing billing. |
21:02.40 | jhowardPA | shmaltz: nothing. |
21:02.44 | shmaltz | jhowardPA, even when putting on hold? |
21:02.50 | jhowardPA | shmaltz: yeah, not a thing. |
21:02.55 | shmaltz | then your device is not putting on remote hold |
21:03.00 | shmaltz | what are you using? |
21:03.12 | jhowardPA | shmaltz: I think I don't have either oss or alsa initialized in modules.conf - is that the problem? |
21:03.14 | shmaltz | how many vvvvvvvvvvvvvvv are you using with asterisk? |
21:03.33 | shmaltz | jhowardPA, how are you testing it? |
21:03.43 | shmaltz | no you don't need oss or alsa |
21:03.54 | psycodad | did anybody ever run * on a vmware guest system ? I have linked a real system and a test host in virtual machine, an when i dial the *-demo on the remote via iax2 the sound is about 3 times toooo slow.. |
21:04.29 | jhowardPA | shmaltz: I've got 2 Cisco 7940G's, and I've got extn 888 pointed at WaitMusicOnHold(30) |
21:04.45 | lohelle | Does anyone have an example ser.cfg (SER proxy) to forward all all sip "traffic" to two (or more) asterisk servers? |
21:04.48 | shmaltz | and you are dialing 888? |
21:05.26 | jhowardPA | If I do, it waits a bit, then says "something is terribly wrong" audibly, and hangs-up. Placing a call on hold from phone-to-phone doesn't do anything (audible) |
21:05.32 | riksta | jhowardPA: great |
21:05.42 | jhowardPA | riksta: thanks :) |
21:05.45 | riksta | jhowardPA: i'm just about to head out to the pub |
21:05.49 | shmaltz | and what is the CLI output |
21:05.54 | shmaltz | ? |
21:05.57 | jhowardPA | riksta: I'm insanely jealous. |
21:06.03 | jhowardPA | shmaltz: none |
21:06.07 | riksta | jhowardPA: uhh? ok! hehe, laters |
21:06.09 | shmaltz | make sure you have like 20 vvvvvvvvvvvvvvvvvvvvvvvvvv when starting the console |
21:06.25 | jhowardPA | shmaltz: "-vvvvvvvvvvvvvvvvvvvvvvvvvvv" ? |
21:06.30 | jhowardPA | for verbose? |
21:06.33 | shmaltz | yep |
21:06.37 | jhowardPA | ok, one sec... |
21:07.09 | *** join/#asterisk habakuk (~chatzilla@24-119-164-129.cpe.cableone.net) |
21:07.21 | jhowardPA | executing moh on new stack... started class 'default' |
21:07.41 | shmaltz | is that all it is doing? |
21:07.47 | jhowardPA | playing tt-somethingwrong |
21:07.52 | jhowardPA | hangup |
21:07.54 | shmaltz | now check your musiconhold.conf |
21:08.02 | shmaltz | what does default say? |
21:08.24 | jhowardPA | It timed-out before the somethingwrong. |
21:08.28 | Veryhot | can someone recommend a opensource cdr program? |
21:08.29 | shmaltz | then check the path that musiconhold.conf shows and confirm that you are hearing whats in the path |
21:08.47 | jhowardPA | Default says mp3:/usr/share/asterisk/mohmp3 |
21:08.53 | shmaltz | Veryhot, doesn't really exist under GPL |
21:08.58 | shmaltz | try ASTPP |
21:09.12 | Veryhot | shmaltz: or something like under $500 :) |
21:09.37 | shmaltz | and ls in /usr/share/asterisk/mohmp3 returns what? |
21:09.47 | jhowardPA | In /usr/share/asterisk/mohmp3, I've got the two mp3s that come with AMP (AMP is not installed): QuajiroPromo.mp3 and TristeAlegriaPromo.mp3 |
21:10.33 | jhowardPA | they were playing fine when I had Asterisk@Home installed on this machine, so I assume they're in the right format. |
21:10.37 | jhowardPA | (I copied them by hand, I didn't leave the drive intact between installs) |
21:10.39 | shmaltz | I have no clue what this mp3 are so do yourself a favor move those out of there, and put in your own that you know play something and test it |
21:10.54 | habakuk | I'm trying an Originate from the manager interface. However I notice that CDR's generated are way out of whack. Has Anyone else noticed this problem? |
21:11.11 | jhowardPA | shmaltz: I'm playing one through the speaker right now, works fine. |
21:11.14 | habakuk | I'm using CVS today btw |
21:11.21 | shmaltz | habakuk, yeah |
21:11.25 | jhowardPA | Tried the second, works as well. |
21:12.21 | habakuk | shmaltz: did you find out why that's happening? |
21:12.25 | shmaltz | do pstree -G -a do you see mpg123 holding on to those files? |
21:12.28 | jhowardPA | Hmmm... Is there a config which binds the output of mpg123 to a channel on the PBX? :\ |
21:13.30 | jhowardPA | shmaltz: files in pstree? I see the mpg123 procs... let me run lsof |
21:13.39 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
21:13.54 | bannerman | Is asterisk@home pretty cool? |
21:14.07 | foobos | bannerman, not really |
21:14.13 | jhowardPA | Yeah, it's holding the first one open. |
21:14.52 | bannerman | foobos: how so? |
21:15.06 | jhowardPA | shmaltz: where does the mpg123 output go? |
21:15.18 | jhowardPA | ahh, stdout |
21:15.21 | shmaltz | where ever you tell it to |
21:15.31 | shmaltz | asterisk redirects it to itself |
21:15.32 | foobos | bannerman, too many noobies breakng the pre-programmed extensions.conf thingie |
21:15.54 | bannerman | foobos: newbs break all sorts of things, whether they use asterisk@home or not |
21:16.00 | bannerman | foobos: I know this, I am newb. |
21:16.18 | jhowardPA | shmaltz: what does asterisk do with it, once it's piped in? |
21:16.26 | jhowardPA | ie, is it configged somewhere? |
21:16.35 | shmaltz | it will wake it up whenever MOH starts |
21:16.41 | shmaltz | nah |
21:16.56 | *** join/#asterisk PBX_Boy (L0ck@82-37-180-88.cable.ubr04.wals.blueyonder.co.uk) |
21:17.52 | jhowardPA | Hmmm... worked on A@H, so something's configged different... is Asterisk calling mpg123 to wakeup by name? Maybe it's off because I've got it in as mpg123-oss which is linked from mpg123... |
21:18.41 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com) |
21:19.20 | jhowardPA | Hmmm... strace mpg123 is showing very little going on. |
21:19.26 | barmal | tzanger: you there? |
21:19.36 | sivana | he's gone |
21:20.00 | barmal | ok.... |
21:20.39 | bannerman | are there any other Bad Things in asterisk@home? |
21:22.28 | shmaltz | jhowardPA, try downloading mpg123 again and compile |
21:23.00 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
21:23.00 | *** mode/#asterisk [+o bkw_] by ChanServ |
21:23.09 | jhowardPA | shmaltz: I did that, too. |
21:23.16 | *** join/#asterisk file[laptop] (~file@mctn1-7126.nb.aliant.net) |
21:23.20 | jhowardPA | before I came here. |
21:23.23 | jhowardPA | 0.59r |
21:25.10 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
21:26.29 | jhowardPA | WHOA! |
21:26.50 | *** join/#asterisk jmacz (~jmacz@63.245.86.225) |
21:26.59 | jhowardPA | Manually running mpg123 with "--mono -r 8000 -b 2048 -f 8192" outputs "No supported rate found!" |
21:27.10 | jhowardPA | there's a quick answer, I'd bet. |
21:30.07 | bannerman | quick, need handholding! I've got to go, and I would like to delay a download by an hour and a half (or make it run at 4 pm my time) |
21:30.23 | bannerman | how do I schedule a command like that? is cron the way? |
21:32.01 | eKo1 | Why are you asking such a question here? |
21:32.19 | pgpkeys | exactly |
21:32.27 | pgpkeys | #unix or #linux |
21:32.47 | eKo1 | You're assuming he/she uses a *nux. |
21:32.53 | shmaltz | or maybe #downlad |
21:32.59 | shmaltz | I mean #download |
21:33.11 | jhowardPA | If my mpg123 (0.59r) doesn't like the "--mono -r 8000 -b 2048 -f 8192" set of options, what's the matter? |
21:33.31 | eKo1 | mpg123 sucks. Avoid using it if you can. |
21:33.44 | shmaltz | jhowardPA, get the defaults that come with asterisk |
21:34.17 | eKo1 | You can always go the the source code dir. and type 'make mpg123'. |
21:34.47 | *** join/#asterisk jf_ (~jeanfranc@toronto-HSE-ppp4024266.sympatico.ca) |
21:35.02 | jf_ | any way to make * dial faster on zap channel |
21:35.14 | eKo1 | Dial faster? |
21:35.20 | jhowardPA | I was wrong, it works if I pipe to stdout... but it gives me a different error: "Warning, flexibel rate not heavily tested!" |
21:35.25 | jhowardPA | Is that important? |
21:35.28 | Qwell | no |
21:35.34 | eKo1 | Ignore warnings. |
21:36.02 | jhowardPA | I would, if things were working ;) |
21:36.13 | jf_ | eK01: yes, when i use my sip phone, i want to call someone on zap channel (fxo) it take like 4sec before i hear the ring |
21:36.25 | jf_ | in sip phone |
21:36.30 | jf_ | it's long 4 sec |
21:36.30 | eKo1 | jf_: Press the send or pound button on your sip phone. |
21:37.00 | jhowardPA | I see, looks like that's due to the mp3 being VBR? |
21:37.29 | eKo1 | Do not use variable-bit-rate mp3s. |
21:37.35 | jf_ | ek01: u mean end the number by a # |
21:37.42 | eKo1 | Yes. |
21:38.01 | jf_ | let me try |
21:38.08 | jhowardPA | eKo1: Hmmm... it's no a vbr mp3 |
21:38.18 | jhowardPA | brb |
21:39.11 | jf_ | eko1: is it possible to auto do that |
21:39.34 | eKo1 | What do you mean 'auto do' |
21:39.51 | jf_ | so i do not have to press #à |
21:39.53 | jf_ | # |
21:40.14 | eKo1 | uh, what's wrong with pressing pound? |
21:40.37 | jf_ | i want to * do that not me |
21:41.19 | *** join/#asterisk hypa7ia (~leigh@67.71.86.109) |
21:41.32 | eKo1 | I think there's a way to set the amount of time * will wait for digits when you type a number but I don't recall. |
21:42.31 | eKo1 | But if you make it too short, then you will have trouble dialing. |
21:42.44 | jf_ | oh ok |
21:43.04 | Qwell | or, don't use . as part of your extension matches |
21:44.54 | bannerman | eKo1: Because I need to schedule an Asterisk-related ISO download sometime after hours, and I was in here already. Sorry.. |
21:45.02 | *** join/#asterisk doughecka (~dheckaman@doughecka.user) |
21:45.16 | Qwell | use `at` |
21:45.19 | doughecka | what would cause asterisk to not recognize dtmf tones from a cisco phone |
21:45.36 | doughecka | mine works fine, but this new setup I have its not wanting to see dtmf |
21:45.45 | eKo1 | doughecka: check the dtmf mode. |
21:45.59 | doughecka | its set to that rtf thing |
21:46.02 | doughecka | orwhateveritis |
21:46.11 | doughecka | rfc |
21:46.15 | *** join/#asterisk Tili (~Tili@202.133.65.241) |
21:46.18 | doughecka | but it wasnt set before |
21:46.37 | doughecka | and it said inband wont work with other codecs and info doesnt work with voicemail |
21:46.41 | eKo1 | rfc2833 |
21:47.03 | eKo1 | Make sure _both_ * and the phone are using that mode. |
21:47.27 | doughecka | how do I set it on the cisco |
21:47.44 | eKo1 | Hell if I know. Read its manual. |
21:47.59 | doughecka | lol |
21:48.09 | bannerman | I have an occasional issue where I can hear the other side, but they can't hear me. Happens on both incoming and outgoing calls. |
21:48.13 | doughecka | I am acting like a user today cause I feel like it |
21:48.14 | doughecka | :P |
21:49.54 | bannerman | Mid-call.. the call works fine.. then incoming audio quits, for about 20 seconds. |
21:50.57 | corlis | Hrm. great. misdn won't work with asterisk-cvs.... |
21:52.31 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
21:52.59 | doughecka | huh, odd |
21:53.00 | FuriousGeorge | does the source for the driver that i need to compile for the tdmp400 come with the zaptel package |
21:53.02 | doughecka | it doesnt see numbers |
21:53.05 | doughecka | but it sees the pound sign |
21:53.41 | Qwell | FuriousGeorge: yes |
21:53.51 | *** join/#asterisk likwid-- (likwid@nc-205-240-44-137.dyn.sprint-hsd.net) |
21:53.54 | FuriousGeorge | make <drivernaqme>? |
21:54.00 | FuriousGeorge | <drivername>* |
21:54.04 | Qwell | no |
21:54.09 | *** join/#asterisk girabraz (~christian@200.121.129.178) |
21:54.09 | Qwell | its included |
21:54.11 | FuriousGeorge | make config? |
21:54.18 | Qwell | just make |
21:54.54 | FuriousGeorge | i get a module not found |
21:55.00 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
21:55.01 | FuriousGeorge | isnt it wctdm? |
21:55.07 | FuriousGeorge | for the tdmp4000 |
21:55.10 | Qwell | did you make install? |
21:55.26 | FuriousGeorge | whoops |
21:55.32 | JohnnyC | Portugal esta por ai alguem ? |
21:55.44 | FuriousGeorge | eu fallo espanhol |
21:55.55 | *** join/#asterisk gein (~gein@213.134.110.241) |
21:56.11 | JohnnyC | estou mesmo a procura de um tuga ! |
21:56.48 | CoaxD | JohnnyC: no hablo el portuguesa o italiano pero.. sabes espanol? :) |
21:57.01 | eKo1 | s/espanol/español |
21:57.03 | FuriousGeorge | que e uma tuga? |
21:57.18 | CoaxD | eko1: o algo asi. jeje |
21:57.20 | JohnnyC | um portugues |
21:57.38 | FuriousGeorge | nos os chamamos "pork chops" ;) |
21:57.52 | CoaxD | FuriousGeorge: mmm. carnes! |
21:57.55 | eKo1 | eu no soi portugues |
21:58.00 | CoaxD | furiousgeorge: yo queiro yo quiero! |
21:58.03 | FuriousGeorge | lol |
21:58.09 | FuriousGeorge | chorizo na brassa |
21:58.12 | FuriousGeorge | yummy |
21:58.18 | CoaxD | Mmmmmmmm |
21:58.20 | CoaxD | siii |
21:58.36 | eKo1 | Chuleta? |
21:58.38 | FuriousGeorge | frango assado, verdad, johnnyc. o chraasco e muinto bon |
21:58.59 | CoaxD | he estado estudiando espanol hace tres anos o algo.. necesito aprender mas pero entiendo, mas o menos |
21:59.19 | eKo1 | CoaxD: You needs a spanish keyboard. |
21:59.22 | FuriousGeorge | mudase a newark NJ. apprenderas espanol, portuguese, todo |
21:59.52 | FuriousGeorge | lo malo es que to olvidaras del ingles ;) |
21:59.56 | CoaxD | eko1: nahh |
22:00.00 | CoaxD | eko1: just a lappytop |
22:00.04 | FuriousGeorge | but the food is soooo good |
22:00.05 | DaLion | heheh |
22:00.06 | DaLion | yeah |
22:00.15 | CoaxD | furiousgeorge: jeje si!! pero poco a poco :) |
22:00.24 | DaLion | creo que estas un poco loco cabron |
22:00.40 | FuriousGeorge | best in new jersey, if you are near by, check it out. i sh*t u not |
22:00.43 | eKo1 | Cuantos anos tienes? |
22:00.43 | CoaxD | DaLion: hahahahaha |
22:00.50 | FuriousGeorge | dalion: no manches, carnal |
22:00.52 | CoaxD | eko1: moi? |
22:01.06 | eKo1 | jbot, translate es en Cuantos anos tienes? |
22:01.13 | eKo1 | lol |
22:01.16 | CoaxD | hahahahaha |
22:01.17 | FuriousGeorge | lol |
22:01.20 | Hmmhesays | ~e&m |
22:01.21 | bugbot | e&m is assigned nothing and reported nothing. |
22:01.29 | Hmmhesays | ~wink |
22:01.31 | jbot | ACTION winks at hmmhesays |
22:01.31 | bugbot | wink is assigned nothing and reported nothing. |
22:01.32 | Hmmhesays | heh |
22:01.49 | Hmmhesays | i'm looking for some info on wink start |
22:01.54 | CoaxD | furiousgeorge: tengo 28 anos ahora.. pero tengo mi cumpleanos en "may" |
22:01.59 | Hmmhesays | so I don't sound like an idiot when this carrier calls back |
22:02.38 | FuriousGeorge | coaxd: ya eres un "viejete" en este mundo |
22:02.48 | FuriousGeorge | (el mundo de *) |
22:03.13 | FuriousGeorge | ps. may is ez: mayo |
22:03.54 | CoaxD | furiousgeorge: mayo si |
22:04.05 | CoaxD | furiousgeorge: como se dice "viejete" en ingles? |
22:04.18 | FuriousGeorge | "geezer" |
22:04.24 | FuriousGeorge | more or less |
22:04.29 | CoaxD | furiousgeorge: jeje siii. de acuerdo :) |
22:04.30 | |Vulture| | yea |
22:04.34 | |Vulture| | geezer lol |
22:04.37 | |Vulture| | old fart |
22:05.00 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
22:05.15 | eKo1 | pedo viejo |
22:05.19 | FuriousGeorge | i was brought up speaking gallego in a portuguese/spanish speaking neighborhoos, so i take up spoken language much better than * |
22:05.32 | |Vulture| | CoaxD: que dia en mayo? |
22:05.38 | CoaxD | Vulture: 2nd |
22:05.40 | Qwell | FuriousGeorge: Are you suggesting a speech config for asterisk? |
22:05.41 | CoaxD | Vulture: tu? |
22:05.50 | |Vulture| | 29th |
22:05.53 | FuriousGeorge | qwell: no |
22:05.55 | Qwell | :p |
22:05.57 | CoaxD | Vulture: mmm. not taurus! |
22:05.58 | FuriousGeorge | but should i be? |
22:06.09 | |Vulture| | hehehe no I don't even know my sign |
22:06.27 | CoaxD | FuriousGeorge: The thing is, dude, i'm the exact opposite. I pick up config languages like mad. but try to get me to learn a language.. |
22:06.31 | CoaxD | FuriousGeorge: I *suck* at it |
22:06.49 | CoaxD | FuriousGeorge: Only reason I learned so much spanish is because of this chick I used to work with; she didn't speak a word of spanish, but.. Man, was she ever hot. |
22:06.52 | FuriousGeorge | mais savoire des idomes est tres important |
22:06.54 | FuriousGeorge | ! |
22:07.23 | FuriousGeorge | coaxd: thats great, what women will make us do |
22:07.40 | CoaxD | FuriousGeorge: I went out with her one night. Got the spanish cook to come take her order and everything |
22:09.11 | FuriousGeorge | lol, chics dig spanish speakers, unless they are rednecks and you look like one too |
22:09.17 | CoaxD | FuriousGeorge: Hahahaha |
22:09.23 | FuriousGeorge | iow, cant speak for middle americans |
22:09.37 | CoaxD | FuriousGeorge: Honestly, dude, I'm a total redneck. Always ahve been |
22:09.50 | CoaxD | FuriousGeorge: Note: High functioning redneck. I don't have old beat up cars in my front yard, etc |
22:09.51 | FuriousGeorge | would you date a borriqua? |
22:09.53 | eKo1 | Chicks don't dig spanish speakers. |
22:10.00 | eKo1 | Where did you get that from? |
22:10.00 | CoaxD | FuriousGeorge: wtf is that? |
22:10.02 | FuriousGeorge | lol, high functioning |
22:10.12 | FuriousGeorge | borriqua is a porto rican chic |
22:10.13 | CoaxD | Eko1: Chicks like guys who can speak more than 1 language. they think its interesting |
22:10.18 | CoaxD | FuriousGeorge: Oh, sure I would |
22:10.29 | FuriousGeorge | ek01: then i guess chics dig me |
22:10.30 | CoaxD | FuriousGeorge: Not just because she was hot, tho. I never would do that. Gotta be somebody like me |
22:10.30 | eKo1 | Fuck, I can speak three. |
22:10.39 | CoaxD | eKo1: Fluently? |
22:10.40 | eKo1 | I don't see chicks falling all over me. |
22:10.48 | CoaxD | eKo1: You hang in the wrong crowds then |
22:10.56 | FuriousGeorge | eKo1: were you baeten with the ugly stick? |
22:11.09 | eKo1 | That's besides the point. |
22:11.22 | FuriousGeorge | right, because the fairer sex is never superficial |
22:11.38 | marlowe | Is anyone using livevoip w/ sip using g.729 ? |
22:12.00 | eKo1 | I'm kidding. I get with 3 new chics every week. |
22:12.44 | FuriousGeorge | eKo1: look out for rashes |
22:12.50 | CoaxD | eko1: Heh |
22:13.05 | eKo1 | I use boots instead of condoms so don't worry. |
22:13.06 | CoaxD | eKo1: The crabs are easy to get rid of, too. Just go to walmart. they'll hook you up |
22:13.43 | FuriousGeorge | the HPV is harder though |
22:13.44 | eKo1 | I wouldn't know. Never had crabs. Looks like you have though. |
22:14.19 | FuriousGeorge | which is green the fxo or fxs |
22:14.30 | eKo1 | Green? |
22:14.54 | FuriousGeorge | i bought a tdmp400 and i got 2 fxos and 2 fxss, which is which |
22:14.58 | FuriousGeorge | two are green two are red |
22:15.58 | FuriousGeorge | CoaxD: so you would date a PRcan, where are u from |
22:16.13 | eKo1 | I think red indicates an error condition. |
22:16.36 | FuriousGeorge | keep in mind i spanish, so you wont offend me, i'm probably more gringo que tu, con mis pelos rubios |
22:16.46 | eKo1 | You're from spain? |
22:16.48 | FuriousGeorge | eKo1: the actual daughter cards are made of green or red |
22:17.06 | FuriousGeorge | no im from nj but everyone else in my fam is |
22:17.16 | eKo1 | What? I have a bunch of FXO daughter cards and they don't have any color. |
22:17.46 | eKo1 | Unless you mean the PCB. |
22:17.56 | FuriousGeorge | i do |
22:18.00 | FuriousGeorge | i forgot the name |
22:18.02 | eKo1 | and if you're from NJ, then you're not spanish. |
22:18.03 | CoaxD | eKo1: Nah. never actually had crabs |
22:18.20 | CoaxD | eKo1: Had lice tho. Just as bad. (Same genus, different species.) |
22:18.26 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
22:18.52 | FuriousGeorge | eK01: we are getting into semantic arguments. ask people what they are and they will say "I'm Irish" having never met anyone from ireland. i actually have spanish citizenship, and as per you, im not spanish |
22:19.07 | FuriousGeorge | standards keep getting tougher. u must be against affirmative action |
22:19.09 | eKo1 | I'm talking about nationality. |
22:19.24 | FuriousGeorge | i hope ur not talking culturally |
22:19.24 | CoaxD | FuriousGeorge: Have mercy on the guy. he's dating a puerto rican |
22:19.29 | FuriousGeorge | cuz u'd be way off base |
22:19.48 | eKo1 | Not ethnicity (as there rarely are any 'ethnic' people around). |
22:20.26 | FuriousGeorge | i have spanish citizenship when i go through customs in spain, i go through the "nationals" line. thanks for playing |
22:20.41 | FuriousGeorge | eKo1: u live somewhere with no ethnic people? |
22:20.42 | eKo1 | So you are spanish then. |
22:20.51 | FuriousGeorge | did we just make a circle? |
22:20.57 | FuriousGeorge | i have dual citizenship |
22:21.03 | CoaxD | eKo1: Do you think one must be spanish to have citizenship in spain? |
22:21.15 | eKo1 | Well yes. |
22:21.21 | CoaxD | eKo1: (Do you have to be white to have citizenship in the USA?) |
22:21.41 | FuriousGeorge | and conversely, asians born here must be asian nationals not americans |
22:21.44 | eKo1 | No, you can be a dog for all I care. |
22:21.48 | CoaxD | No. People here in the USA might well be Hmong, but they'll answer to "HEY, AMERICAN!" |
22:22.00 | eKo1 | I like to keep things simple. |
22:22.08 | CoaxD | eKo1: Nothing is ever simple. |
22:22.17 | FuriousGeorge | lol, uh oh |
22:22.33 | eKo1 | Well, I try. |
22:22.45 | FuriousGeorge | so anyway, back to * |
22:22.48 | FuriousGeorge | which has the red pcb? |
22:23.01 | eKo1 | I would imagine the FXS. |
22:23.48 | FuriousGeorge | fxs connects office phone to station (verifying, i always get those confues) |
22:23.51 | FuriousGeorge | right? |
22:23.59 | eKo1 | ~fxs |
22:24.00 | jbot | hmm... fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
22:24.00 | bugbot | fxs is assigned nothing and reported nothing. |
22:24.02 | eKo1 | ~fxo |
22:24.06 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
22:24.12 | bugbot | fxo is assigned nothing and reported nothing. |
22:24.18 | FuriousGeorge | oh that clever jbot |
22:24.32 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
22:25.11 | eKo1 | Que hambre tengo, joder. |
22:25.20 | CoaxD | eKo1: :/ |
22:26.07 | eKo1 | jbot, I want a gyro. |
22:26.28 | eKo1 | ~gyro |
22:26.30 | bugbot | gyro is assigned nothing and reported nothing. |
22:26.39 | eKo1 | Hmm...no gyros. |
22:29.01 | FuriousGeorge | eKo1: looks like the fxo is the red one |
22:32.10 | BoRiS | I know this probably doesn't exist but is their such thing as a virtual modem(sip that can talk back to *) adapter for windows? |
22:33.29 | Elshar | I suppose I don't see what you mean. If you wanted a modem modem, you could just buy one. If you wanted a sip phone for windows, there's a couple out there.. |
22:33.29 | *** part/#asterisk DaLion (~DaLion@toronto-HSE-ppp3983233.sympatico.ca) |
22:33.36 | Elshar | along with an iax phone program too |
22:34.05 | FuriousGeorge | yeah i think u r talking about www.xten.com |
22:34.49 | Elshar | Yep, that's one |
22:35.02 | Nugget | well, it isn't a "virtual modem". |
22:35.10 | BoRiS | Elshar: Its like that, I want to emulate a modem on my windows box... This emulation adapter connects back to asterisk via SIP. Then I can use winfax or some softfaxing programs that will use the virtual adapter to send faxes |
22:35.19 | Elshar | I see |
22:35.20 | Elshar | hmm |
22:35.27 | *** join/#asterisk tsetane (~tsetane@212.4.33.58) |
22:35.36 | Nugget | modems don't talk SIP, real ones or virtual ones. |
22:35.50 | Elshar | I could've swore I've seen something like that though. |
22:36.10 | Nugget | but more importantly, trying to mix faxing and asterisk is nothing but disappointment and headache. |
22:36.15 | BoRiS | virtual ones *can* as long as someone had written a driver/program to do that. |
22:36.35 | Nugget | a "virtual modem" is a concept which exists only in your head. |
22:36.48 | Nugget | you're using that nonsense phrase to describe a device whose real name you don't know |
22:36.49 | fearnor | nugget: uhhh |
22:36.51 | BoRiS | Ummm, no |
22:36.51 | fearnor | you are so wrong. |
22:36.58 | fearnor | there is 'virtual modem' thang. |
22:36.58 | BoRiS | You are wrong Nugget |
22:37.00 | Nugget | ok. |
22:37.03 | fearnor | my TNT supports it just fine. |
22:37.07 | fearnor | modem emulation via TCP |
22:37.13 | Nugget | and those virtual modems connect to asterisk for faxing? |
22:37.25 | Nugget | interesting. ok, I'm wrong. |
22:37.29 | fearnor | there's a rfc for that even. so you can control modem DCD and other things via tcp ;) |
22:37.44 | BoRiS | Nugget, You were saying that a "virtual modem" concept was only in my head........Pffffff |
22:37.48 | fearnor | not to asstricks, but to my PC, it looks just a remote serial port. |
22:37.58 | fearnor | and i can use windows fax to send faxes and shiznit ;) |
22:38.07 | Nugget | clever |
22:38.24 | fearnor | this concept is about 10 years old |
22:38.31 | fearnor | gotte shiva!@# |
22:38.37 | Nugget | what's it used for? |
22:38.38 | BoRiS | hehe |
22:38.47 | Nugget | modem pooling or something? |
22:38.52 | fearnor | nugget: yep pretty much |
22:39.00 | fearnor | original application was modem pooling |
22:39.09 | fearnor | but you can use it for whatever |
22:39.25 | fearnor | buy a 200$ max 6000 and rock on, 4 T1s worth of that ;) |
22:41.48 | *** join/#asterisk guugmember (~guugmembe@200.6.223.209) |
22:42.00 | eKo1 | virtual modem? You mean the ones on a T1 modem card? |
22:42.33 | guugmember | hello, I am in a proyect where I have to put 48 remote extensions, besides IAXy is there other hardware solution more unexpensive and that talks IAX? |
22:42.55 | eKo1 | no |
22:42.56 | eKo1 | next |
22:43.08 | fearnor | guug: why iax. |
22:43.14 | fearnor | fuck iax. |
22:43.23 | eKo1 | eh, no. |
22:43.23 | guugmember | because sip is blocked in my ISP |
22:43.24 | eKo1 | iax rocks |
22:43.27 | Sedorox | fearnor: |
22:43.33 | guugmember | fearnor, what do you recommend then? |
22:43.39 | Sedorox | Gunnar: call them and bitch |
22:43.39 | fearnor | guug: get a non-ghetto ISP or just change ports, and SIP will probably work. |
22:43.42 | tzanger | iax > * |
22:43.55 | fearnor | if your ISP is blocking something, they'll block your iax too. |
22:43.58 | eKo1 | iax >> ISP |
22:44.26 | fearnor | my main complaint about iax is that it makes no distinction between call control path and media path |
22:44.39 | fearnor | well, that and the fact that only ghetto low-density devices support iax ;P) |
22:44.39 | tzanger | fearnor: that's its only shortcoming IMO but it's also a strength |
22:44.44 | eKo1 | Who cares. It works, especially through NAT. |
22:44.56 | fearnor | so does SIP |
22:44.58 | tzanger | all you need to do to add it is to allow callbacks from dropped out calls |
22:44.59 | fearnor | ktnxbye |
22:45.07 | fearnor | :) |
22:45.25 | eKo1 | sip + nat = pain |
22:45.32 | guugmember | so, besides of the protocol, anyting besides the IAXy? |
22:45.40 | eKo1 | h.323 |
22:45.44 | tzanger | I've heard from the yate crew that they have a sip stack that owrks very well through nat |
22:45.48 | tzanger | but I've not tested it yet |
22:45.51 | fearnor | bingo |
22:45.59 | fearnor | its all in the stack, not teh protocol |
22:46.18 | eKo1 | Well, there's nothing wrong with SIP conceptually. |
22:46.31 | eKo1 | It's just that it sucks in NATed environments. |
22:46.31 | tzanger | eKo1: I think it's a hideous protocol |
22:46.35 | tzanger | tries to do everything for everyone |
22:47.01 | fearnor | tzanger: once you realize what *could* be done with sip, you'll understand *why* it is so complex. |
22:47.06 | eKo1 | What's wrong with that. |
22:47.09 | FuriousGeorge | i followed http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf and i get a line 142: Unable to open master device '/dev/zap/ctl |
22:47.13 | tzanger | fearnor: I don't want my VOIP stack to do everything |
22:47.23 | tzanger | fearnor: I want my VOIP stack to handle voice and maybe video, that's it |
22:47.25 | FuriousGeorge | im trying to load the drivers for the tdm400 i just got |
22:47.36 | Hogie | FuriousGeorge: 2.6 kernel? |
22:47.38 | jhowardPA | Anyone know how to debug mpg123 MoH problems? |
22:47.45 | FuriousGeorge | hogie: yessir |
22:47.46 | jhowardPA | I'm still not getting anywhere... |
22:47.52 | Hogie | FuriousGeorge: udev? |
22:47.54 | fearnor | tzanger: well, how about "remote presence"? |
22:47.55 | FuriousGeorge | yup |
22:47.59 | fearnor | jabber-like |
22:48.00 | tzanger | fearnor: I have jabber for that |
22:48.04 | fearnor | or VMWI-like |
22:48.06 | sivana | hehe |
22:48.09 | nvrs | Hogie, you are on the ball |
22:48.09 | fearnor | note, VMWI *is* remote presence. |
22:48.10 | Hogie | did you read README.udev (or is it udev.README?) in the zaptel source dir? |
22:48.21 | tzanger | fearnor: I don't need my VOIP stack to handle mail and presence and everything |
22:48.28 | FuriousGeorge | thanks, i didnt know abut that, ill check it out |
22:48.32 | tzanger | I am a firm believer in using the right protocol for the right job |
22:48.36 | Hogie | it says so in the compile |
22:48.38 | jhowardPA | ...nothing but silence on hold, though it looks like it's working. |
22:48.39 | Hogie | you just have to watch for it |
22:48.49 | Hogie | er, says to do that |
22:48.54 | FuriousGeorge | hogie, i use gentoo, which compile |
22:48.54 | fearnor | gee, why not? its one protocol to support on your media gateway controller |
22:49.05 | tzanger | fearnor: no thanks |
22:49.05 | Hogie | did you emerge? |
22:49.15 | tzanger | fearnor: I already have a lot of jabber rolled out |
22:49.21 | fearnor | your MGC *needs* to know about remote presence to properly signal VMWI |
22:49.24 | tzanger | I already have a lot of FTP, SMTP and web services rolled out |
22:49.26 | eKo1 | tzanger: so what would you rather use? |
22:49.31 | fearnor | shrug |
22:49.37 | fearnor | tzanger: how does that scale for you ;) |
22:49.41 | Hogie | I know if you compile zaptel from cvs, it will say "it looks like you are using udev, please read README.udev" or something like that |
22:49.41 | tzanger | eKo1: IAX2 + Jabber really |
22:49.49 | tzanger | fearnor: it works pretty well so far |
22:49.49 | FuriousGeorge | i did en "emerge this && emerge that && emerge the other && rc-update add thisd default" overnight and went to sleep |
22:49.57 | tzanger | one protocol doesn't mean it'll scale well |
22:49.57 | Hogie | lol |
22:49.58 | eKo1 | I thought Jabber was just a chat client/server. |
22:50.00 | nvrs | # Section for zaptel device |
22:50.01 | nvrs | KERNEL="zapctl", NAME="zap/ctl" |
22:50.01 | nvrs | KERNEL="zaptimer", NAME="zap/timer" |
22:50.01 | nvrs | KERNEL="zapchannel", NAME="zap/channel" |
22:50.01 | nvrs | KERNEL="zappseudo", NAME="zap/pseudo" |
22:50.01 | nvrs | KERNEL="zap[0-9]*", NAME="zap/%n" |
22:50.07 | sivana | ~pastebin |
22:50.09 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
22:50.09 | bugbot | pastebin is assigned nothing and reported nothing. |
22:50.09 | tzanger | eKo1: presence too |
22:50.10 | FuriousGeorge | just installed. i have distcc running on 4 boxes and it took 8 hours ;) |
22:50.12 | Hogie | and dont forget the permissions nvrs |
22:50.16 | nvrs | stick that in the bottom of /etc/udev/rules.d/50-udev.rules |
22:50.21 | nvrs | permissions?!~?! |
22:50.26 | Hogie | yes nvrs, lol |
22:50.27 | nvrs | OMG |
22:50.31 | nvrs | Ive been doing it all wrong!! |
22:50.34 | Hogie | heh |
22:50.48 | goldenear | eKo1: sip + nat = pain --> sip + STUN + nat = ok |
22:50.55 | Qwell | nvrs: README.udev |
22:51.03 | guugmember | goldenear, what is STUN? |
22:51.22 | Qwell | SIP + /dev/urandom = pain |
22:51.27 | eKo1 | STUN = SIP Tryint to Use NAT. |
22:52.13 | Sedorox | ahahah |
22:52.19 | nvrs | the readme doesnt mention anything about permissions |
22:52.25 | goldenear | guugmember, http://www.voip-info.org/wiki-STUN |
22:52.36 | guugmember | goldenear, im there |
22:52.56 | eKo1 | Bottom line is, stun will not solve all your nat issues. |
22:53.20 | nvrs | Holgie: the readme doesnt mention anything about permissions |
22:53.23 | Weezey | whoa, when I change my phone to g726 the at.gsm is irritating. |
22:53.28 | goldenear | stun doesn't work only for symmetric nat IFAIK |
22:53.37 | *** join/#asterisk verge (~jfargen@rrcs-24-227-48-10.se.biz.rr.com) |
22:53.37 | sivana | Weezey: hey |
22:53.57 | Weezey | hey hey! |
22:54.00 | goldenear | IAX doesn't work with snat neither AFAIK |
22:54.02 | Weezey | been a while. |
22:54.10 | guugmember | goldenear, whatever I want to run IAX, besides IAXy any other less expensive product? |
22:54.12 | tzanger | goldenear: it works pretty well with SNAT actually |
22:54.20 | Hogie | nvrs: mine does: http://pastebin.ca/9627 |
22:54.55 | tzanger | both endpoints natted makes it difficult to initiate a call without a port forward but otherwise it works just fine |
22:55.20 | goldenear | tzanger, you can't native bridge two IAX clients between SNAT |
22:55.20 | nvrs | Hogie, different file |
22:55.24 | Weezey | is g729 worth buying? |
22:55.33 | eKo1 | Anybody here use the Mediatrix FXO gateway? |
22:55.34 | Hogie | nvrs: mine was from 1.0 stable on april 11 |
22:55.36 | verge | Hello #asterisk |
22:55.38 | nvrs | Holgie: which distro? |
22:55.40 | tzanger | goldenear: not without a little work |
22:55.48 | jhowardPA | I'm getting "Spawn extension (default, 888, 1) exited non-zero on 'SIP/501-26fa'" when I end a call that should be playing music on hold. Anyone know what's up? |
22:56.16 | Hogie | nvrs: cvs |
22:56.19 | nvrs | Holgie, think im doing 1.0.7 |
22:56.30 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
22:56.39 | eKo1 | jhowardPA: If you end a call, it will exit. This is normal. |
22:56.51 | verge | When I call people they can hear me, but when people call me I can't hear them. |
22:56.54 | goldenear | tzanger, you have to have something (like an asterisk server) in the path for two IAX clients behind SNAT to work |
22:56.57 | verge | Is that a problem with my dialplan? |
22:57.17 | eKo1 | verge: We need more specifics. |
22:57.20 | tzanger | goldenear: or port forward, or do some stack magic... however a lot less than SIP would have to do to get the RTP stream to do the same |
22:57.21 | goldenear | tzanger, the same can be accomplish with sip and RTP proxy... |
22:57.34 | jhowardPA | eKo1: I assumed that a non-zero return was not a good thing...? |
22:57.41 | tzanger | goldenear: what is the difference between having an RTP proxy and having an unnatted * box in the middle? none |
22:57.55 | eKo1 | jhowardPA: eh, how else do you think commands exit. |
22:58.15 | verge | I am using a SIP ATA. It's behind a NAT box, but the * server is not behind NAT. |
22:58.20 | eKo1 | IAX was made with * in mind. |
22:58.25 | jhowardPA | eKo1: With "zero" - meaning clean exit? |
22:58.47 | FuriousGeorge | Hoagie: im still getting the same error after editing the rules and permissions file as instructed. do i gotta restart udev or something |
22:58.50 | eKo1 | No. Zero usually means the command failed. |
22:58.59 | *** join/#asterisk netofsickcoder (~netofsick@cpe-24-170-74-115.stx.res.rr.com) |
22:59.01 | jhowardPA | eKo1: I'm also not getting any music, still. |
22:59.03 | verge | What doe you think eKo1? Do you need any more info? |
22:59.16 | jhowardPA | eKo1: I think you're mistaken. |
22:59.36 | goldenear | tzanger, right an rtp proxy is like an unnatted * box |
22:59.37 | eKo1 | jhowardPA: I write * apps. I think I know what I'm talking about. |
22:59.42 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
23:00.08 | *** part/#asterisk moy (~kvirc@201.135.105.124) |
23:00.13 | jhowardPA | eKo1: Usually, a non-zero return code implies an error state, most of those error codes identify the error state. Is this not the case for Asterisk? |
23:00.29 | eKo1 | jhowardPA: If you don't believe me, ask in the #asterisk-dev |
23:01.08 | eKo1 | That or look at the code. |
23:01.19 | eKo1 | You'll see that most apps return -1. |
23:01.31 | jhowardPA | Ahhh, cool. |
23:01.37 | jhowardPA | Thanks for clearing that up. |
23:02.09 | verge | I am just trying to use G711. |
23:02.37 | verge | eKo1: do you have any suggestions? |
23:02.54 | jhowardPA | I figured it would be more like I'm used to, ala 'false ; echo $?' |
23:02.54 | FuriousGeorge | Hoagie: actually, i tried unloading and reloading the module after editing udev perms and rules, and now i get an error loading wcfxs |
23:02.56 | eKo1 | verge: Make sure both phones are set to use ulaw/alaw and that these codecs are allowed in *. |
23:03.06 | FuriousGeorge | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
23:03.37 | verge | eKo1: this was actually someone from the PSTN trying to call me. |
23:03.44 | eKo1 | jhowardPA: I understand. It was confusing for me at first also. |
23:03.46 | verge | I am connected to the PSTN via SIP as well. |
23:04.03 | eKo1 | Using what? |
23:04.12 | goldenear | tzanger, using only 1 udp port in indeed a benefit in many cases (like having only 1 port to forword if you want to host an * server behing a nat/firewall) |
23:04.36 | *** join/#asterisk bjohnson (~bjohnson@ip190-172.dsl.istop.com) |
23:04.39 | verge | eko1: I connect ATA---->*----->Telasip.com |
23:04.45 | verge | It's all sip |
23:04.46 | jhowardPA | eKo1: could you suggest where I might go to research my music problem? I can't identify a location which is failing to pipe the music into the SIP channel. |
23:05.33 | goldenear | tzanger, but this can also be a problem: CDR won't work after native briging of two endpoints ... neither call hold/forwarding :( |
23:05.35 | jhowardPA | I'm using WaitMusicOnHold(30) |
23:05.36 | eKo1 | jhowardPA: Look in the wiki for the musiconhold.conf page. |
23:05.57 | tzanger | goldenear: which is why I want to make a patch where after a native bridge a callback to the dorpped server is made so it can update its CDR |
23:06.04 | jhowardPA | I did, none of it seemed to relate to the symptoms I'm seeing - ie, the fixes I identified were already correct. |
23:06.42 | *** join/#asterisk fugitivo (~ajf@201.255.106.8) |
23:07.02 | FengShui | jhowardPA: What program are you sing for playing the MoH? |
23:07.03 | eKo1 | jhowardPA: Bottom line is, mpg123 sucks; it's discontinued and will be dropped from * soon (I hope). |
23:07.38 | goldenear | tzanger: how will this work ? this looks like a crappy work around ... |
23:07.40 | FengShui | yep. Madplay is much better. I've had stable MoH since I switched |
23:07.47 | tzanger | goldenear: nah |
23:07.53 | jhowardPA | I'm using mpg123 - 0.59r - Madplay's a better option? |
23:07.55 | tzanger | shouldn't be bad |
23:07.59 | jhowardPA | I'll try that. |
23:08.02 | jhowardPA | Thanks! |
23:08.20 | tzanger | A->B->C, A&C negotiate to drop B out, end of call A and C both issue an ACK'd IAX2 IE with CDR update |
23:08.20 | FengShui | jhowardPA: yeah, madplay doesn't spawn off all of those hung processes like mpg123 does. |
23:08.29 | goldenear | tzanger, also IAX has at the moment a pretty bad codec negocation capabilities |
23:08.39 | tzanger | goldenear: that too is being worked on |
23:08.46 | tzanger | in-call codec renegotiation |
23:09.16 | verge | eko1? do you need anymore info? |
23:09.59 | eKo1 | Make the calls and check which codecs are being used. |
23:10.36 | goldenear | tzanger and what about call hold/transfer not being possible during a native bridge call |
23:10.38 | goldenear | ? |
23:11.11 | tzanger | goldenear: haven't heard about that one |
23:11.41 | goldenear | did you try it ? |
23:12.31 | goldenear | call hold (with music)/transfer works only when * stay in the path... |
23:12.54 | nvrs | I cant get my sound card to work.. is it really required for anything important in z |
23:12.57 | nvrs | asterisk |
23:13.00 | tzanger | goldenear: well no shit |
23:13.05 | tzanger | where do you think the music comes from |
23:13.15 | tzanger | if A or C is playing the music then it should work just fine |
23:13.18 | Hogie | fairies |
23:13.21 | tzanger | but if B drops out then of course it ain't gonna work |
23:13.23 | Hogie | am I right tzanger? |
23:13.30 | jhowardPA | Damn, madplay didn't fix it. |
23:13.43 | jhowardPA | I'm just not getting anything out of the phone... |
23:14.43 | *** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) |
23:14.43 | jhowardPA | There it went with the "something is terribly wrong, goodbye" message. |
23:15.39 | jhowardPA | Is this the right syntax? It looks right: "/usr/bin/madplay --mono -R 8000 --output=raw:-" |
23:15.44 | goldenear | tzanger indeed, but it would be nice if B could be back in the path when need (eg music on hold). |
23:15.47 | Hogie | if I have my handset have 2 line instances on my phone, will it ring the 2nd instance when a 2nd call comes in, or the first instance? |
23:16.23 | FengShui | johwardPA: yeah, pretty muc. Here's mine: /usr/local/bin/madplay -Q -z --atten |
23:16.23 | FengShui | uate=-10 --mono -R 8000 --output=raw:- |
23:18.04 | jhowardPA | FengShui: Trying yours |
23:18.09 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
23:18.20 | jhowardPA | No dice. Still dead line. |
23:19.55 | *** join/#asterisk netofsickcoder (~netofsick@cpe-24-170-74-115.stx.res.rr.com) |
23:20.14 | FengShui | jhowardPA: Do you see the process running? |
23:20.48 | jhowardPA | I do, it looks proper. |
23:22.50 | habakuk | anyone using the manager originate function? |
23:23.00 | jhowardPA | strace output looks good - it appears to be reading the mp3, and pumping the output to stdout. |
23:23.31 | jhowardPA | For some reason, the SIP channel just isn't getting the feed. |
23:24.21 | habakuk | jhowardPA: do you have a zaptel driver ? |
23:24.35 | jhowardPA | I'm using ztdummy - should I not be? |
23:24.39 | habakuk | err kernel module loaded |
23:24.57 | jhowardPA | zaptel 174048 4 [ztdummy wcusb] |
23:25.49 | habakuk | jhowardPA: have you verified that otherthings like meetme work? |
23:26.13 | jhowardPA | habakuk: No, I haven't. How should I go about checking? |
23:26.30 | habakuk | setup a meetme room and see if that works |
23:26.39 | jhowardPA | Ok, trying... |
23:27.19 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
23:27.56 | jhowardPA | Yeah, meetme works. |
23:28.05 | jhowardPA | My conf room 300 is working fine. |
23:29.07 | habakuk | jhowardPA: ok so its got to be something else |
23:29.27 | jhowardPA | Is it something that'd be in my zapata.conf? |
23:30.20 | goldenear | tzanger, I wondering about you're cdr patch: what happens if one end doesn't terminate the call properly (network failure etc ...) ? |
23:32.50 | tzanger | I figure it would just take the update from one side |
23:33.03 | tzanger | I haven't written any code yet |
23:33.19 | tzanger | hmm |
23:33.24 | tzanger | I forgot onion in my salad |
23:33.45 | goldenear | bon appetit :) |
23:34.36 | Hogie | ztcfg -vvv shows what? |
23:34.43 | jhowardPA | Damn, still no music... I upped the volume, in case, but to no avail. |
23:35.02 | jhowardPA | 0 channels configured. |
23:35.06 | jhowardPA | Is that it? |
23:35.12 | Hogie | didn't mean to you howard |
23:35.14 | Hogie | I forgot to hit tab |
23:35.15 | Hogie | lol |
23:35.18 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
23:35.24 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
23:35.34 | denon | PTG123: well well well? :) |
23:35.54 | jhowardPA | Hogie: would that affect my music on hold? |
23:36.01 | Hogie | prob not |
23:36.19 | jhowardPA | Damn... |
23:36.20 | jhowardPA | :\ |
23:36.45 | Hogie | my colo facility (computer, not telco colo) wants $900/month for a pri + cross connect:( |
23:36.57 | denon | youch |
23:37.01 | PTG123 | :) |
23:37.10 | PTG123 | thats why i colo in a carrier neutral facility with no cross-connect fees |
23:37.26 | Hogie | I HATE HATE HATE the dallas info mart |
23:37.34 | Hogie | no way I'd colo there |
23:37.45 | denon | PTG123: get a chance to visit our friendly local government office? |
23:38.07 | PTG123 | hah my wife was suppose to do it today, when i see here tonight i'll get you the tracking # :) |
23:38.20 | denon | great :) |
23:38.34 | denon | im headin home, cya |
23:40.31 | FuriousGeorge | can anyone confirm when the status leds are supposed to go on, on the tdm400. when drivers are modprobed? or after ztcfg |
23:43.23 | goldenear | tzanger: what do you think about always maintening a link with * (only sig and info) even during native bridging with an other endpoint ? |
23:44.07 | tzanger | that would probably be a lot more convoluted but I'd have to investigate it |
23:44.27 | *** part/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
23:45.48 | tzanger | jbot, furiousgeroge is Furious George, the Furious little monkey... See him come, as furious as can be... |
23:45.49 | jbot | okay, tzanger |
23:45.54 | tzanger | oh |
23:45.56 | tzanger | he left |
23:46.01 | tzanger | I tough that was a join |
23:47.05 | goldenear | why a call park/transfer is not possible when * is not is the path |
23:47.16 | fugitivo | hey, I did a nmap to my linksys pap2-na, look at this |
23:47.17 | tzanger | uh |
23:47.19 | fugitivo | OS details: Sipura SPA-1000 or SPA-2000 POTS<->VOIP gateway |
23:47.25 | tzanger | because the fucking box isn't in the loop anymore? :-) |
23:47.47 | Qwell | fugitivo: doesn't mean much |
23:48.01 | fugitivo | Running: Sipura embedded |
23:48.15 | Qwell | It doesn't actually ASK the device what it is. It makes a logical guess. |
23:48.26 | tzanger | if you, me and fugitivo are talking and I piss off for a smoke, how can you ask me for the time? |
23:48.28 | fugitivo | Qwell: maybe yes, maybe not |
23:48.34 | Qwell | I'd say knowing that its an ATA is a pretty good guess |
23:48.49 | Qwell | tzanger: piss off for a smoke? Mind if I steal that one? |
23:49.05 | tzanger | goldenear: when B drops out, you lose access to all of B's services |
23:49.08 | tzanger | Qwell: heh |
23:49.10 | tzanger | go ahead |
23:49.14 | Qwell | excellent |
23:49.18 | goldenear | I'm test this with iaxcomm, and I can't tranfer/park the endpoint I'm talking to because of the native bridge :( |
23:50.01 | tzanger | goldenear: just say notransfer=yes in the iax.conf for the user/peer |
23:50.45 | goldenear | but why couldn't my endpoint ask the * box to reinvite the other end ? |
23:51.03 | tzanger | goldenear: feel free to add that functionality if it's that important |
23:51.15 | tzanger | I see IAX2 as a simple stack for simple connectivity |
23:51.27 | goldenear | and it is :) |
23:52.22 | goldenear | how does a tranfer works inside * ? |
23:53.04 | tzanger | goldenear: read the code |
23:53.40 | goldenear | but basicaly, isn't it like a SIP reinvite ? |
23:53.46 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
23:53.48 | *** mode/#asterisk [+o bkw_] by ChanServ |
23:54.00 | goldenear | (I'm not a skilled coder) |
23:54.24 | tzanger | nor am I :-) |
23:56.11 | TomL | ~seen manxpower |
23:56.15 | jbot | manxpower <~eric@adsl-35-236-60.msy.bellsouth.net> was last seen on IRC in channel #asterisk, 4h 43m 2s ago, saying: 'EGADS! I have a negative karma!'. |
23:56.15 | bugbot | seen manxpower is assigned nothing and reported nothing. |
23:56.38 | MikeJ[Jayden] | we take unskilled coders too ;) no time like the pres to learn |
23:56.58 | goldenear | tzanger, but you said you're going to write a patch for * :) |
23:57.07 | tzanger | I've written several patches for * |
23:57.11 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
23:57.30 | *** part/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
23:57.36 | *** join/#asterisk covici (covici@static-162-83-93-166.fred.east.verizon.net) |
23:59.56 | sean | when I do 'sip show peers' and the status column for my DID (sip proxy) show "unmonitored", is that normal? I want to be connected to the DID. |