00:03.47 | *** join/#asterisk popvoxdave (i=user@dave2.toad.net) |
00:08.20 | *** part/#asterisk Malthus (n=dumbdumb@port0005-adb-adsl.cwjamaica.com) |
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00:13.55 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
00:14.39 | redder86 | Does Asterisk always require a 60-second refresh? I'm using libiax and no matter what I tx in the registration request for a refresh I always get a response indicating a 60-second refresh. |
00:15.49 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-178.cybersurf.com) |
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00:51.41 | Rav1974 | not a lot of cativity today :) |
00:52.09 | fugitivo | it's friday |
00:52.36 | Rav1974 | :) I don't have a life |
00:53.19 | Rav1974 | when I installed a x100p after a TE110p aterisk dies |
00:57.04 | *** join/#asterisk used_ (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
01:03.28 | *** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
01:03.28 | shido6 | ? |
01:03.54 | Qwell | ! |
01:03.59 | MikeJ[Laptop] | * |
01:05.23 | Rav1974 | when I installed a x100p after a TE110p aterisk dies, any suggestions? |
01:05.44 | Qwell | dies how? and don't repeat |
01:06.01 | redder86 | anyone here that can assist with using libiax2 ? |
01:06.17 | Qwell | redder86: if you ask a question about it |
01:06.19 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
01:06.19 | Rav1974 | :) It says there is something wrong with tor or zap |
01:06.29 | Qwell | Rav1974: that help none |
01:07.11 | *** join/#asterisk NirS (n=Nir@84.94.52.50.cable.012.net.il) |
01:07.12 | Qwell | helps* |
01:07.21 | Rav1974 | Qwell: I know, but i'm a newbie, I am rebooting the server, give me a minute |
01:07.27 | Qwell | rebooting?! |
01:07.42 | Qwell | I hope you just compiled a new kernel or something |
01:07.57 | Rav1974 | Qwell:I'm from windows world, if it doesn't work reboot |
01:08.01 | Qwell | ... |
01:08.05 | Rav1974 | if it still doesn't work kick it |
01:08.16 | Rav1974 | if it STILL doesn't work, then it won't ever work |
01:08.24 | Rav1974 | chuck it |
01:08.26 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
01:08.36 | Qwell | Rav1974: let me send you my mailing address |
01:08.37 | Rav1974 | I think I know what is wrong, my zaptel or zapata config files |
01:08.56 | Qwell | I'll even pay for shipping |
01:08.58 | Rav1974 | Qwell: do you want some kicked hardware |
01:09.11 | Rav1974 | I have many pieces hope you have glue |
01:10.44 | redder86 | Qwell: I'm having trouble using the return value of iax_get_event. |
01:11.25 | redder86 | Qwell: for example, after getting the REGACK frame I want to reset an internal refresh value (so that I can refresh the registration in time). |
01:11.35 | *** join/#asterisk prav33n|home (n=praveen@59.144.38.160) |
01:11.41 | redder86 | Qwell: how do I use iax_get_event? |
01:11.46 | Qwell | got me |
01:11.47 | prav33n|home | Hello |
01:11.49 | Rav1974 | this is to be continued... Qwell: thanks for helping. I have some kids to play with |
01:12.06 | *** join/#asterisk used_ (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
01:12.09 | Rav1974 | i deeply appreciate all the helpful people who keep coming back to help us newbies |
01:12.16 | redder86 | Qwell: you don't use iax_get_event, then? What do you use instead? |
01:12.23 | Rav1974 | anyone in NYC, I'd buy you a beer |
01:12.27 | Qwell | redder86: I never said I did |
01:12.33 | prav33n|home | I have ordered Linksys WRT54GP2-NA to use with astreisk from VeriLAN.com |
01:12.48 | prav33n|home | Assuming that this model is not locked to anyone |
01:12.59 | prav33n|home | Can I use it fruitfully? |
01:13.22 | prav33n|home | I need some help here, please. |
01:13.53 | redder86 | Qwell: ah, so why'd you want me to ask the question if you don't know about libiax2? |
01:15.06 | *** join/#asterisk onhold (n=maxgluck@200.109.166.83) |
01:16.13 | Qwell | redder86: so that you wouldn't have to keep repeating yourself |
01:16.31 | Qwell | "Does anybody know how to fix this error?" |
01:16.35 | Qwell | WHAT error? ya know? |
01:16.46 | redder86 | Qwell: I specified that the error was with using libiax2 |
01:16.48 | Qwell | if somebody sees it, they might answer why |
01:17.02 | onhold | Hi, I need some advice please. I've configured musiconhold.conf and the calls are placed on hold, I have mp3 files, without tag in the /var/lib/asterisk/mohmp3 directory but no music is heard at all. Does asterisk installs mpg123 or I have to do it? |
01:17.08 | prav33n|home | Anyone here have some idea on Linksys WRT54GP2-NA |
01:17.20 | Qwell | redder86: see, what onhold said was an example of a good question |
01:17.41 | Qwell | onhold: from the asterisk source dir, do `make mpg123`. It'll download and compile it for you |
01:17.52 | sivana | tzanger: ping |
01:17.55 | *** join/#asterisk NirS (n=Nir@84.94.52.50.cable.012.net.il) |
01:18.05 | onhold | thanks Qwell, let me try that... =) |
01:18.09 | redder86 | Qwell: I asked whether anyone or not was listening that knew about libiax2 enough to answer some questions about its usage. The answer to that question was apparently no. |
01:18.10 | Qwell | actually, it might just download it, then you probably have to `cd mpg123 && make && make install` |
01:19.21 | onhold | yes, it created the directory mpg123-0.59r |
01:19.35 | Qwell | onhold: go in there, and `make && make install` |
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01:26.30 | onhold | Qwell, nothing... same happens. May it have something to do with the format not being configured? I'm entering moh classes show, I see the classes but only slin format. I really have searched over and over but can't find where to add mp3 format, if that is what's missing. |
01:27.42 | onhold | Qwell: moh files show doesn't display anything, but the files are there... |
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01:30.46 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
01:31.39 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
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01:37.35 | Qwell | onhold: doesn't show any for me either |
01:42.19 | *** join/#asterisk visik7_ (n=mierda@host70-140.pool8258.interbusiness.it) |
01:45.21 | L|NUX | how can i update my * server on DUNDi graph ? |
01:45.47 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
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01:50.38 | *** join/#asterisk musictomyears (n=maxgluck@200.109.166.83) |
01:51.48 | musictomyears | Qwell: thank you so much, there was just one thing left, the missing link! |
01:51.56 | Qwell | whats that? |
01:52.19 | musictomyears | ln -s /usr/local/bin/mpg123 /usr/bin/mpg123 |
01:52.29 | Qwell | hmm |
01:52.37 | Qwell | /usr/local/bin/ not in your path or something? |
01:53.01 | *** part/#asterisk hubbaba (n=me@71-12-171-64.dhcp.gsvl.ga.charter.com) |
01:54.23 | musictomyears | apparently not by default, it's a new system, but on the wiki it is commented |
01:55.02 | musictomyears | http://www.voip-info.org/tiki-index.php?page=Asterisk+config+musiconhold.conf |
01:55.44 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
01:58.15 | musictomyears | Thanks, have a great weekend, and a good night... |
01:58.19 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
01:58.45 | *** join/#asterisk iq (n=iq@209-180-97-92.omah.qwest.net) |
02:00.37 | *** join/#asterisk oej (n=oej@63.77.68.196) |
02:01.21 | Qwell | how is the polycom 301 for a cheap phone? |
02:01.30 | DaPrivateer | its nice Qwell |
02:01.46 | DaPrivateer | only problem is it doesnt re-register sometimes after a loss of network |
02:01.57 | DaPrivateer | you need to reboot the phone, and it takes about 4 minutes to load |
02:02.14 | Qwell | What else is fairly cheap? |
02:02.20 | Qwell | (besides GS) |
02:02.44 | DaPrivateer | *shrug* |
02:03.09 | DaPrivateer | if you are gonna go with Polycom i would recommend going for the 501 though. i believe its only a differnce of 30 dollars but there are a bunch more features |
02:03.27 | Katty | beep. |
02:04.52 | file | beep beep |
02:05.39 | MikeJ[Laptop] | tickling? what is that all about? |
02:05.45 | Katty | file: ... |
02:06.11 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:07.01 | MikeJ[Laptop] | does anybody in here have an audiocodes UA? |
02:07.06 | file | nope |
02:07.46 | MikeJ[Laptop] | file, do you have the ability to send dtmf notify via ser? |
02:08.29 | file | if you set it up to route that method, yeah |
02:08.32 | file | SER is just a proxy though |
02:08.45 | *** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st) |
02:08.48 | MikeJ[Laptop] | I need to be able to convert to it |
02:08.54 | MikeJ[Laptop] | I don't have anythign that can send it |
02:09.07 | file | then no. |
02:09.17 | MikeJ[Laptop] | hmmm do any of the sip test stacks support notify? |
02:09.20 | MikeJ[Laptop] | sipp or somthing |
02:09.26 | file | doubt it |
02:09.30 | MikeJ[Laptop] | dtmf notify specifically |
02:09.47 | MikeJ[Laptop] | damn, I just want to test this one damn bug and figure it out.. |
02:09.57 | visik7_ | does anyone can give me an extract of dialplan for personalized (I mean selectable extension for park and other ) supervised call transfer ? |
02:14.00 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:15.21 | *** join/#asterisk Assid (n=assid@203.115.64.60) |
02:15.22 | Assid | heya |
02:16.30 | Ariel_ | OK I am almost giving up on a system. Argh I hate to do this.... Why are there so many different distro's out there? Why? |
02:16.47 | MikeJ[Laptop] | Ariel_, windows? :D |
02:17.03 | Assid | hrmm.. tell me abt it.. my xwindows stop working.. and i frogot what made it do that |
02:17.07 | Ariel_ | In my view this is just as bad as windows. |
02:17.13 | Assid | and that too just in time since i got my KVM switch |
02:17.58 | Ariel_ | I have a SuSe setup that just does not want to run php4's correctly. not allowing me to get it to work with a mySQL database.... |
02:19.13 | iq | Hi, anyone know how to reset SPA-3000 ATA to factory default? |
02:19.31 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:20.03 | Assid | how do i come to know which version of CVS head i am using? |
02:20.21 | Ariel_ | show version |
02:20.33 | Assid | just says CVS HEAD |
02:20.59 | Qwell | Then its cvs head |
02:21.08 | Qwell | cvs isn't really versioned |
02:21.16 | MikeJ[Laptop] | show version should show a date\time as well |
02:21.32 | Qwell | MikeJ[Laptop]: thats compile date/time, right? |
02:21.36 | Qwell | show version files |
02:21.44 | *** join/#asterisk izo (n=izo@193.202.114.43) |
02:21.54 | Qwell | Thats as close as you'll get to the actual "version" |
02:22.00 | MikeJ[Laptop] | ummm.. I don't recall now |
02:22.07 | MikeJ[Laptop] | I do beleive so |
02:22.18 | MikeJ[Laptop] | ahhh.. gotta love CVS... bleh |
02:22.22 | Assid | Asterisk CVS-HEAD built by root@voip on a i686 running Linux on 2005-07-29 07:17:46 UTC |
02:22.29 | Assid | think its time to upgrade? |
02:22.38 | Qwell | daily |
02:22.50 | *** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net) |
02:22.56 | Assid | hehe.. nah.. actually that server is production |
02:23.00 | Qwell | I have a cron running every 45 minutes that downloads and compiles latest |
02:23.15 | Assid | i could do it on my test bed box |
02:23.18 | pr0m | umm.. overkill. ;-) |
02:23.27 | Qwell | I was kidding :p |
02:23.30 | MikeJ[Laptop] | Qwell, that;s kinda dangerous with a project on cvs... |
02:23.31 | MikeJ[Laptop] | heh |
02:23.31 | MikeJ[Laptop] | ok |
02:23.45 | Qwell | I'd do it if it was svn though :D |
02:23.45 | Assid | do i use the same cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
02:23.49 | MikeJ[Laptop] | bout to say, without atomic commits, you'd be getting half commited stuff all the time |
02:23.53 | Qwell | Assid: still got the old source? |
02:23.57 | Assid | yep |
02:23.57 | Qwell | Assid: if so, just cvs up |
02:24.22 | Assid | dunno :( |
02:25.11 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:25.59 | Assid | cant find a package called cvsup in debian |
02:26.31 | Qwell | Assid: cvs up |
02:26.34 | Qwell | not cvsup |
02:27.10 | Assid | do i have to set cvsroot etc ? |
02:27.23 | Qwell | if you're in the dir, it should just know |
02:28.31 | Assid | athlonxp:/usr/src# cvs up asterisk-sup |
02:28.32 | Assid | cvs update: cannot open CVS/Entries for reading: No such file or directory |
02:28.32 | Assid | cvs [update aborted]: no repository |
02:28.43 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:29.42 | MikeJ[Laptop] | asterisk-sup? |
02:30.18 | Katty | Hmmhesays: are you out drinking yet? |
02:30.34 | Assid | yeah.. thats my file i created |
02:31.04 | MikeJ[Laptop] | Assid, huh? |
02:32.13 | Assid | guess i will just re-download the whole thing |
02:33.15 | Assid | hrmm dont i need zapata as well.. for x100p ? |
02:33.21 | *** join/#asterisk apardo (n=apardo@23.Red-83-44-179.pooles.rima-tde.net) |
02:33.29 | Qwell | why not just type what I said? |
02:33.32 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
02:33.38 | Qwell | cd asterisk && cvs up |
02:34.03 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
02:34.33 | Bhaal | Anyone else here using FWD via iax2 and their asterisk server is constantly reconnecting to FWD? |
02:34.35 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:35.00 | Assid | yeah that seems to work |
02:36.01 | Assid | dont i need zapata fro x100p ? |
02:37.38 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
02:39.09 | Katty | god bless yoga pants. |
02:39.19 | file[laptop] | Katty: that's hot |
02:39.27 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
02:39.30 | Katty | file[laptop]: they're so /soft/ |
02:39.37 | Katty | file[laptop]: i love outfits that feel like pjs. |
02:41.01 | MikeJ[Laptop] | pants? |
02:41.24 | MikeJ[Laptop] | Assid, yes, x100p is zap |
02:42.18 | Assid | hrmm .. |
02:42.19 | Assid | make: *** No rule to make target `doc/ztcfg.sgml', needed by `doc/ztcfg.8.gz'. Stop. |
02:42.33 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:42.42 | Katty | MikeJ[Laptop]: pants for yoga. |
02:43.12 | Netgeeks_ | theus the term 'yoga pants' |
02:43.27 | Katty | MikeJ[Laptop]: http://www.campus-classics.com/images/Product/img1-15371.jpg <- not me, but she does have yoga pants. |
02:43.50 | *** join/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net) |
02:44.01 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
02:44.06 | Katty | those must have been some scary pants. |
02:44.10 | Katty | MikeJ[Laptop]: did you get url? |
02:44.42 | Netgeeks_ | he's too busy staring at the url |
02:44.53 | Katty | i wouldn't doubt it. |
02:44.56 | MikeJ[Laptop] | what url? |
02:44.56 | Katty | she's hot |
02:45.03 | Katty | 21:51 < Katty> MikeJ[Laptop]: |
02:45.03 | Katty | <PROTECTED> |
02:45.03 | Katty | <PROTECTED> |
02:45.23 | MikeJ[Laptop] | no.. I got bounced off.. |
02:45.42 | Netgeeks_ | she's 'get you in jail' young too |
02:45.51 | MikeJ[Laptop] | what makes those "yoga" pants |
02:46.05 | Katty | MikeJ[Laptop]: that's what the tag said. |
02:46.07 | Qwell | MikeJ[Laptop]: corporate branding |
02:47.04 | Netgeeks_ | actually it's probably just like martial art's pants, the design of the crotch area allows for much greater leg motion without stress or binding |
02:48.05 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
02:50.05 | Assid | umm.. what am i missing? i am still getting this error |
02:50.05 | Assid | make: *** No rule to make target `doc/ztcfg.sgml', needed by `doc/ztcfg.8.gz'. Stop. |
02:52.16 | *** part/#asterisk ericm_06 (n=ericm_06@c-24-9-147-88.hsd1.co.comcast.net) |
02:52.19 | fugitivo | asterisk compiled in 1.49 min on a dual opteron |
02:52.22 | Netgeeks_ | amazingly quiet here all of the sudden |
02:52.47 | Netgeeks_ | fugitivo: yeah, ran time make on a X2 4800+ the other day came in at 1m 12s |
02:53.14 | fugitivo | that's better |
02:53.23 | Assid | it compules for you? |
02:53.27 | Assid | how come im getting errors? |
02:53.28 | Netgeeks_ | and runs |
02:53.36 | fugitivo | Assid: i use gentoo |
02:53.37 | Assid | your using head? |
02:53.42 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
02:53.42 | Netgeeks_ | *shrug* my guess would be dependancies |
02:54.08 | Netgeeks_ | yes, I compiled 2005-08-28 head with 8 different patches from mantis |
02:54.13 | fugitivo | gentoo does all for you! |
02:54.14 | fugitivo | ;) |
02:54.31 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
02:54.37 | Netgeeks_ | and I compiled it on redhat FC4 x86_64 with a 'install everything' dist |
02:55.17 | Netgeeks_ | did you build zaptel first? |
02:55.57 | Assid | im tyring to |
02:56.02 | Assid | it keeps giving me a stupid error |
02:56.08 | Netgeeks_ | you are trying to build zaptel? |
02:56.20 | Ariel_ | Assid, your doc directory is not there. |
02:56.26 | Qwell | cvs up -d |
02:56.38 | Ariel_ | did up rm -fr zaptel directory before you download the new build? |
02:56.41 | Assid | found it.. docbook-utils wasnt installed |
02:56.49 | Assid | still doesnt help |
02:57.19 | Assid | there we go.. -d did the trick |
02:57.49 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:01.42 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:02.27 | Netgeeks_ | I was going to drop a couple 4 port cards into the X2 and loop them and see how it would run 192 channels of SIP-ZAP-SIP, but alas the motherboard locked solid after post with the cards installed |
03:02.45 | Bhaal | Is anyone else here experiencing problems with FWD's iax2 gateway? |
03:02.47 | Netgeeks_ | Never could get past the post screen |
03:05.26 | visik7_ | is there a better call parking around ? |
03:06.04 | *** join/#asterisk opus_ (n=opus@dahphish.org) |
03:06.28 | opus_ | Hello |
03:06.40 | opus_ | is anyone experiencing Realtime database problems with CVS HEAD? |
03:06.51 | opus_ | http://pastebin.ca/22016 |
03:07.01 | opus_ | I'm getting "Unkown database 'asterisk' |
03:07.09 | opus_ | however it succesffuly connects to mysql |
03:07.10 | opus_ | (?) |
03:07.21 | zippp | is there an asterisk db? |
03:07.25 | fugitivo | do you have a database called "asterisk" ? |
03:07.32 | zippp | mysql -u root |
03:07.33 | opus_ | i have it called 'asterisk_realtime' instead |
03:07.46 | fugitivo | then, change the name or change the config |
03:07.52 | zippp | that is the problem, it is looking for asterisk |
03:07.55 | Assid | wats realtime database? |
03:07.59 | opus_ | well, i'm more interested in why its not working |
03:08.04 | Netgeeks_ | ~wiki |
03:08.15 | Netgeeks_ | gah |
03:08.18 | Netgeeks_ | damn jbot |
03:08.27 | zippp | opus_, because there is no `asterisk` db, only a `asterisk_realtime` |
03:08.38 | Netgeeks_ | There is a good description of Asterisk realtime on www.voip-info.org, Assid |
03:08.45 | fugitivo | opus_: because it's look for a database called "asterisk" and you don't have one |
03:08.45 | opus_ | realtime database = database driven asterisk backend, for all .conf files queues/voicemail/sip/iax-channels |
03:09.19 | opus_ | zippp - but none of my configs have 'asterisk' anywhere only asterisk_realtime. :( |
03:09.25 | fugitivo | just, read the messages :) |
03:09.31 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:09.39 | Assid | yep,,, reading it |
03:10.17 | zippp | opus_, `grep -r asterisk /etc` |
03:10.23 | zippp | read each line and make sure |
03:10.36 | zippp | if you don't want to check _everything_ just grep in /etc/asterisk |
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03:11.02 | opus_ | zippp that was the first thing I did |
03:11.09 | Katty | hmm. |
03:11.12 | opus_ | http://pastebin.ca/22017 is my res_mysql.conf |
03:11.51 | zippp | try this |
03:11.55 | zippp | mysql -u asterisk -p |
03:13.06 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:13.10 | Assid | hrmm nice.. everything sql based |
03:13.19 | Assid | easier to make reporting/front end for this |
03:13.43 | opus_ | echo select \@\"hello world\"\;| mysql -uasterisk -pasterisk asterisk_realtime |
03:13.46 | opus_ | echo $? = 0 |
03:13.49 | Netgeeks_ | yeah, but careful, realtime isn't fully ready for primetime |
03:13.55 | Nugget | neither is mysql. :) |
03:14.05 | Assid | was referring to pgsql ;) |
03:14.07 | Netgeeks_ | same could be said for asterisk too |
03:14.09 | Nugget | yay :) |
03:14.21 | opus_ | same could be said about anything |
03:14.32 | opus_ | nobody's willing to help |
03:14.38 | opus_ | then it will stay that way:) |
03:14.40 | Netgeeks_ | yeah, I find it pretty impressive that yahoo's entire backend is... mysql |
03:14.49 | Nugget | it's friday night. I'm drunk. ask again during business hours. |
03:14.58 | Assid | really? i thought only yahoo finance |
03:15.27 | Netgeeks_ | nah, apparently in a recent interview thier head architect said the whole backend was mysql |
03:15.38 | opus_ | yeah right. |
03:15.42 | Assid | hrmm |
03:15.47 | opus_ | :) |
03:16.00 | opus_ | and AT&T runs asterisk realtime |
03:16.11 | Assid | isnt pgsql supposed to be faster for more complex queries? |
03:16.13 | Netgeeks_ | *shrug* if all you are doing is tons and tons of reads.... mysql would be fine |
03:16.34 | Qwell | google uses mysql |
03:16.44 | opus_ | insert into table1 select * from table2, swap 1/2 around and it will eventually corrupt your data. i love mysql |
03:16.54 | Nugget | I can't think of any reason to choose mysql over pgsql in any circumstance, unless you have software that only supports mysql. |
03:17.28 | opus_ | nugget thats exactly why i am using mysql. the least buggiest in my opinion with asterisk realtime from my experience.. |
03:17.35 | Netgeeks_ | I think nugget hit the nail on the head.... I wonder what yahoo's choices where when they started down the road with mysql |
03:17.54 | Assid | save money? |
03:17.57 | Qwell | does pgsql really scale? |
03:18.03 | Qwell | Assid: pgsql is truly free, afaik |
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03:22.34 | h3x | postgres rules |
03:22.52 | h3x | you know, informix is based on postgres |
03:23.09 | h3x | although postgres is way better now |
03:23.24 | Nugget | pgsql is freer than mysql. |
03:25.01 | *** join/#asterisk [2]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
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03:28.48 | fugitivo | postgresql is a real database |
03:28.49 | *** join/#asterisk zip_ (n=zip@68-118-126-104.dhcp.sprn.tx.charter.com) |
03:29.08 | Ariel_ | fugitivo, yes it is so is MYSQL |
03:29.15 | fugitivo | no way |
03:29.21 | fugitivo | mysql is not a real database |
03:30.18 | Nugget | http://www.livejournal.com/users/dirigibles/75781.html <-- unrelated to databases. |
03:30.29 | fugitivo | databases sux! |
03:31.46 | Ariel_ | fugitivo, I am just messing with you. I think all databases have there use and need |
03:31.55 | fugitivo | i agree |
03:32.27 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:33.02 | fugitivo | ms sql is a great database |
03:33.16 | fugitivo | in terms of functionality |
03:34.33 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
03:34.50 | fugitivo | Nugget: burgertime is great |
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03:39.33 | Nugget | yeah, mssql is solid. |
03:39.49 | *** join/#asterisk Grubs (n=Miranda@c220-239-96-230.eburwd5.vic.optusnet.com.au) |
03:40.08 | spackle | did I get here in the middle of a holy war? |
03:40.36 | Nugget | nah |
03:42.22 | Grubs | Q. For connecting 2 phones to asterisk - is it better to use a TDM400P with 2 x FXS ports and do it directly or use a Linksys PAP2 (half the price). Any pros or cons on either approach? |
03:42.29 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:43.37 | fugitivo | Grubs: i think i'll cheaper 2 ip phones instead of a tdm400 with 2 fxs ports |
03:43.47 | h3x | jesus h christ superstar |
03:43.49 | spackle | Grubs - I'd take the sipura. |
03:44.00 | h3x | voip-info could make a fortune selling clickthrus |
03:44.15 | h3x | im getting hammered with leads... |
03:44.15 | fugitivo | Grubs: i use the pap2-na, it works fine |
03:44.22 | Grubs | IP phones are expensive here. more than $100 each. PAP2 is $90. |
03:44.25 | h3x | screw google |
03:44.35 | fugitivo | Grubs: how much is a tdm400 with 2fxs there? |
03:44.38 | Nugget | $100 is expensive? |
03:44.46 | fugitivo | Grubs: around 350? |
03:44.50 | Grubs | no |
03:45.15 | fugitivo | Nugget: in some countries, it is :) |
03:45.20 | Grubs | any way you look at ...if you have the phone already a PAP2 is cheaper |
03:45.34 | fugitivo | Grubs: a pap2 is always cheaper |
03:45.36 | Nugget | I also suggest s sipura over a tdm400p. Zaptel can be a pain in the ass |
03:45.54 | Nugget | IP phones are still the better bet, though, unless you need wireless. |
03:46.04 | fugitivo | Grubs: i was saying, that 2 ip phones, are going to be cheaper than a tdm400 with 2 fxs |
03:46.04 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:46.33 | Grubs | I need the tdm400p because X100p's are mostly crap in AU because of the different line impedance... so I can get the TDM with 1FXO only and use a PAP2 - or I can get the TDM with 1FXO and 2 FXS. |
03:46.35 | fugitivo | Grubs: if you want, go with the pap2-na, it works ok |
03:47.19 | fugitivo | Grubs: if you don't want to spend too much money, go with the pap2 |
03:47.42 | fugitivo | if money is not a concern, buy ip phones |
03:48.13 | Grubs | sounds like there is no down side to the PAP2 then - thanks fugitivo. IP phones are great - but I love being cordless so I'll re-use my DECT phones. |
03:48.15 | spackle | but not grandstream ;-) |
03:48.22 | fugitivo | and not atcom |
03:48.27 | spackle | yeah |
03:49.15 | *** join/#asterisk TheCops (n=mdb@206-248-136-187.dsl.teksavvy.com) |
03:49.20 | fugitivo | Grubs: i'm using pap2-na with a siemens gigaset, it works fine, all the functions you'll need |
03:49.52 | fugitivo | Grubs: remember to get the NA |
03:49.53 | Grubs | Sipura SPA-841 is the cheapest IP phone here: $160 ... but there are not a lot of cordless IP phones around so PAP2 it is then. |
03:49.58 | Grubs | NA? |
03:50.05 | fugitivo | Grubs: the pap2-na is not locked |
03:50.17 | fugitivo | the pap2 is locked for providers |
03:50.24 | TheCops | I have a callwaiting option on my PSTN line, where can I start with asterisk for manage this feature ? |
03:50.29 | Grubs | oh - mine is not locked.. bought independed from a vendor |
03:50.34 | fugitivo | ok |
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03:51.34 | spackle | TheCops: what is your interface to the POTS? Zap? |
03:51.39 | Grubs | thanks for your help. Seems the PAP2 will work straight up.... my experience with X100P's here burned me a bit on the zaptel anyhow :) |
03:51.46 | TheCops | spackle, yup, FXO interface |
03:52.06 | fugitivo | Grubs: but, x100p is fxo, not fxs |
03:52.15 | spackle | Then check your settings in zaptel.conf. There are some available for call waiting. |
03:52.38 | Grubs | yes... I was thinking of the TDM400P with FXS modules using zaptel. |
03:53.26 | *** join/#asterisk onhold (n=maxgluck@200.109.166.83) |
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03:54.22 | *** join/#asterisk sycofly (n=syco@sycofly.com) |
03:54.38 | Grubs | Someone needs to make some DECT IP phones. |
03:54.53 | spackle | what is DECT phone? |
03:55.32 | TheCops | spackle, and what is doing the callwaiting options in the zaptel, how can I use it with a snom ? |
03:55.34 | Grubs | digital cordless |
03:55.57 | fugitivo | Grubs: there are wireless phones |
03:56.21 | Grubs | Wireless is great until the good wife turns on the microwave :) |
03:56.45 | spackle | TheCops: whether to pass the call waiting signal through to the phone while in use I believe. |
03:56.51 | Grubs | PAP2 + DECT is secure and perfect sound... so will do for now. |
03:57.19 | fugitivo | siemens have one that looks like a celphone |
03:57.21 | Grubs | DECT IP PHone!: http://www.gizmodo.com/gadgets/voip/index.php |
03:57.26 | spackle | theCops: is there a flash-hook button on your phone? |
03:57.40 | TheCops | spackle, no, its a snom 320, a lot of button, but not flash |
03:58.00 | Qwell | ip phones don't really have "call waiting" |
03:58.05 | Qwell | They have multiple line appearances |
03:58.17 | TheCops | Qwell, I can choose many line |
03:58.32 | fugitivo | Grubs: that's nice, it's a wireless phone and ata |
03:58.38 | TheCops | but when I have the special beep, I can't take the line |
03:59.05 | Qwell | TheCops: there is no button to activate a line? |
03:59.31 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
03:59.34 | TheCops | yeah but it dont take the other call on the call waiting line |
04:00.17 | TheCops | hrmm |
04:00.26 | TheCops | I guess it have a way to make Flash with a cmd from asterisk |
04:00.31 | Qwell | you should just have to press the button for the line |
04:00.45 | Qwell | it'll put the other line on hold, and switch the active line to that one |
04:00.48 | fugitivo | TheCops: what phone? |
04:00.54 | TheCops | snom 320 |
04:01.00 | fugitivo | is that the atcom? |
04:01.04 | fugitivo | yea, atcom 320 |
04:01.08 | fugitivo | i have the same issue |
04:01.11 | fugitivo | flash doesn't work |
04:01.14 | TheCops | atcom ? |
04:01.16 | TheCops | snom! |
04:01.20 | fugitivo | atcom |
04:01.21 | fugitivo | ! |
04:01.32 | h3x | snom rules!!!! |
04:01.52 | fugitivo | hmm, no, it's not the atcom |
04:02.08 | TheCops | Theres a way to make a extension command when youre in line on a zap channel |
04:02.14 | *** join/#asterisk nick125 (n=nick125@unaffiliated/nick125) |
04:03.04 | nick125 | hey, anyone know where i can get some basic configs (not the example ones, they are too large and hard to work with), with maybe one extension and etc? |
04:03.13 | spackle | apparently some phones have a link button too , http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Flash#comments |
04:04.07 | spackle | nick125, I think John Todd or someone like that has some good config file samples posted publicly. |
04:05.02 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:05.09 | nick125 | ok, the hard part: some example configs like that, except for asterisk 1.2 |
04:06.49 | opus_ | Grubs how well does that work? |
04:07.15 | spackle | and do you have link to the specs on it? |
04:09.01 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
04:09.26 | opus_ | 233mhz is way to slow for asterisk |
04:09.33 | opus_ | it would be great for an ATA however |
04:09.36 | opus_ | :) |
04:10.06 | spackle | opus: not much slower than soekris, is the same as my wrt54g with Asterisk running on it. |
04:10.08 | *** join/#asterisk {tasker} (i=sdfsddad@modemcable064.169-203-24.mc.videotron.ca) |
04:10.11 | Grubs | opus_ - seems to work well enough for my single line. Currently using iLBC which should max out the CPU but I am thinking that all the encoding is being done on my soft-phone client box and asterisk is just passing it through. A bit slow to boot and reconfigure though. Using Debian. |
04:10.35 | opus_ | spackle how well does the wrt54g work |
04:10.37 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:10.49 | Grubs | too slow? Its been working great for a few weeks now *shrug* |
04:10.54 | opus_ | Grubs hmmm. |
04:11.04 | opus_ | The wrt54g with asterisk, I ment. |
04:11.16 | spackle | can make two calls through it, no transcoding. does a basic job. |
04:11.25 | opus_ | why are you using iLBC? doesn't that codec suck? |
04:12.23 | opus_ | spackle - when you say no transcoding, do you mean canreinvite=yes or something to that effect? |
04:12.28 | Grubs | I'll do a little test and connect to asterisk using aLaw and then see if asterisk can talk to my VSP using iLBC :) |
04:12.34 | {tasker} | asterisk HEAD, 1.2.x and > 1.0.5 still don't pass DTMF to the terminating side |
04:12.37 | *** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net) |
04:12.51 | {tasker} | RFC2833 |
04:12.53 | opus_ | tasker -- You have that problem to |
04:12.58 | {tasker} | everyone does |
04:13.09 | {tasker} | there is even a bug opened and no progress since Aug 24 |
04:13.15 | Qwell | doesn't pass dtmf under what circumstances? |
04:13.29 | {tasker} | 1- dial into Asterisk |
04:13.35 | *** join/#asterisk musictomyears (n=maxgluck@200.109.166.83) |
04:13.36 | spackle | opus: no, I pass the calls through the wrtg54g to nufone or another set. not using canreinvite=yes |
04:13.38 | {tasker} | 2- transfer call out |
04:13.40 | opus_ | tasker - i've was told by bkw that its because the media gateway has fucked RFC2833 when I showed him the rtp dump |
04:13.56 | {tasker} | 3- punch some tones |
04:14.04 | {tasker} | terminating side receives no digits |
04:14.07 | {tasker} | rfc2833 |
04:14.10 | {tasker} | H.323 and SIP |
04:14.24 | opus_ | i have the problem as well, with calling in. asterisk will sporatically not register the DTMF digits for one call |
04:14.24 | spackle | so you have to use a cantain crunch whistle with CVS head. |
04:14.25 | {tasker} | it works the other way |
04:14.38 | {tasker} | oh |
04:14.38 | {tasker} | no |
04:14.42 | opus_ | or the next, then like 10 hours later will work. its the only thing stopping me from rolling asterisk out production wise |
04:14.42 | {tasker} | Asterisk receives the tones |
04:14.58 | {tasker} | if I dial into an Asterisk IVR app, it works fine |
04:15.04 | opus_ | I have it so asterisk receives RTP packets, but doesn't register them as DTMF |
04:15.07 | {tasker} | if the app redials out, DTMF does not pass |
04:15.31 | *** join/#asterisk stepcut (n=user@ip68-107-21-88.sd.sd.cox.net) |
04:15.38 | {tasker} | so, I dial into asterisk and the app transfers my call out to, say, my home voicemail or a PBX, I cannot dial any digits |
04:15.51 | {tasker} | however |
04:16.00 | {tasker} | i tested it by calling myself |
04:16.03 | {tasker} | on another phone |
04:16.09 | opus_ | tasker howabout updating the bug with the rtp debug of the call in question |
04:16.14 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:16.17 | Grubs | can people vote on the open bugs to elevate their priority? |
04:16.21 | {tasker} | it's already there |
04:16.27 | {tasker} | here's the thing |
04:16.28 | opus_ | url pls |
04:16.32 | {tasker} | it's 100% reproducible |
04:16.34 | opus_ | i think i know which one but not sure |
04:16.57 | {tasker} | http://bugs.digium.com/view.php?id=4989 |
04:17.04 | {tasker} | but check this |
04:17.07 | {tasker} | i call myself |
04:17.10 | {tasker} | through asterisk |
04:17.17 | {tasker} | phone 1 ----> asterisk -----> phone 2 |
04:17.23 | {tasker} | phone 2 can dial DTMF back to phone 1 |
04:17.29 | {tasker} | phone 1 cannot send DTMF to phone 2 |
04:17.33 | {tasker} | you hear nothing on phone 2 |
04:17.35 | opus_ | tasker is this only for h323? |
04:17.36 | {tasker} | rfc2833 |
04:17.40 | {tasker} | no |
04:17.41 | {tasker} | SIP as well |
04:17.53 | opus_ | OK i skipped it because it didn't say SIP big enough |
04:18.08 | {tasker} | it's an RTP issue |
04:18.28 | {tasker} | RTP debug receives the rfc2833 tone but does not send it to the other leg |
04:18.30 | {tasker} | it eats it |
04:18.44 | {tasker} | this started happening some time after 1.0.1 |
04:19.03 | {tasker} | if I compile an older version, it works without a problem |
04:19.34 | *** join/#asterisk dr123 (n=asdf@pcp09065496pcs.northw01.in.comcast.net) |
04:19.37 | {tasker} | i've tested it with several different types of endpoints, including Cisco AS53xx gateways |
04:19.40 | {tasker} | no joy |
04:20.01 | {tasker} | if i switch to inband & G711, it's working fine |
04:20.17 | {tasker} | because it's plain sound :) |
04:20.23 | {tasker} | but rfc2833 will not pass through |
04:20.28 | {tasker} | one way |
04:21.23 | {tasker} | i've tried to track the problem down but I cannot find the logic that eats the tone |
04:21.42 | dr123 | Is there a way to dial Via URL on Cisco 7960 via IAX2 url and have it pass it to asterisk for the call so it would not be a SIP to SIP call I still want it to go through the server |
04:21.47 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:22.07 | opus_ | tasker i added a comment saying it effects sip as well |
04:22.13 | {tasker} | ok |
04:22.22 | brc_ | ~seen j4m3s |
04:22.26 | jbot | j4m3s <n=debbie@digium.com> was last seen on IRC in channel #asterisk, 1d 7h 31m 26s ago, saying: 'brc_ kevin told me to talk to you about jira'. |
04:22.26 | {tasker} | do you have asterisk setup? |
04:22.29 | opus_ | i bookmarked it |
04:22.35 | opus_ | tasker yeah |
04:22.38 | nick125 | am i crazy for wanting to build a asterisk solution using mysql database (i want to link asterisk to the customer database, so, when a customer calls up, it can automatically lookup the info for the customer and such)? |
04:22.48 | opus_ | i will bug everyone to get that one looked at I think I am having the same problem |
04:22.59 | {tasker} | it's most likely the same issue |
04:23.02 | opus_ | nick125 yup |
04:23.07 | {tasker} | do you have an external IP endpoint you can dial out to? |
04:23.14 | opus_ | tasker I also have an issue is RTP and NAT |
04:23.16 | nick125 | i mean, doesnt asterisk support mysql or something like that? |
04:23.26 | Qwell | nick125: That actually won't be too difficult to do |
04:23.32 | opus_ | tasker yeah, but i can't touch it |
04:23.35 | Qwell | nick125: just a simple AGI, or a C app |
04:23.42 | {tasker} | opus: but you can dial out to it, right? |
04:23.47 | Qwell | app_customerlookup.c, or some such |
04:23.49 | {tasker} | is it connected to a PSTN circuit? |
04:23.58 | nick125 | hrm...c... |
04:24.00 | Qwell | (mysql is very easy to query up from C) |
04:24.03 | opus_ | tasker no I don't touch analog at all |
04:24.09 | nick125 | Qwell: any docs on AGI? |
04:24.17 | *** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au) |
04:24.21 | {tasker} | opus: what's the endpoint connected to? |
04:24.22 | Qwell | nick125: If you're going to use this with much volume, I wouldn't suggest agi |
04:24.33 | Qwell | agi scales poorly, afaik |
04:24.47 | opus_ | Qwell well, the problem with Queue how can you run a Macro before or when an agent picks up. the only idea i had was to loop the SIP channel back through asterisk with another channel that loops back to the real agent and runs a macro in the dial plan that way.. |
04:25.02 | *** join/#asterisk pauldy (n=pauldy@c-67-162-191-108.hsd1.co.comcast.net) |
04:25.04 | {tasker} | opus: what I was going to suggest is to call yourself through asterisk and dial some numbers on both phones. |
04:25.08 | opus_ | Macro(./send_user_desktop_popup.sh) |
04:25.22 | opus_ | tasker - that almost never works for me. |
04:25.26 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:25.26 | opus_ | tasker - or site-to-site |
04:25.38 | {tasker} | opus: dialing or DTMF? |
04:25.50 | opus_ | DTMF |
04:26.03 | opus_ | i can't even use my own company's voice tree 9 times out of 10 right now |
04:26.14 | {tasker} | might be the same issue |
04:26.23 | {tasker} | I have 100% failure |
04:26.24 | nick125 | well, the idea here is that i would build a app that would access a database. then, when a customer calls in and an agent picks up, asterisk would drop a record in the db, and the app would look at entry up by agent number, and then the app can find the customer number and look it up in the customer db. |
04:26.32 | opus_ | tasker I had problems dialing 1-800-CALL-ATT and entering digits |
04:26.39 | opus_ | tasker - and fixed it with modifying rtp.c |
04:26.44 | {tasker} | how? |
04:26.46 | {tasker} | where? |
04:26.51 | opus_ | but Sprint/Verizion/T-mobile voicemail box don't like it |
04:27.22 | {tasker} | where did you modify in rtp.c ? |
04:27.31 | opus_ | search for the comment '800ms' and change the value on the next line from '100ms'(?) or whatever to above 2000 in rtp.c |
04:28.02 | {tasker} | ok |
04:28.18 | opus_ | my co workers also go it working by hitting the dtmf pad extremely fast as well |
04:28.18 | {tasker} | checking now |
04:28.50 | opus_ | tasker -- i think that area of the code is what is fucked. there was a patch against that area somewhere between 1.0.5-1.0.7 |
04:29.56 | stepcut | is MoH working properly under FreeBSD these days (i am rather out of date...) |
04:30.04 | {tasker} | opus: that's when the problem started |
04:30.07 | {tasker} | > 1.0.5 |
04:30.09 | stepcut | (I don't have any zaptel hardware...) |
04:30.57 | opus_ | are you guys listening to the lousiana scanner shoutcast links, they're pretty hard to hear |
04:31.03 | {tasker} | this is what I found |
04:31.04 | {tasker} | /* Make duration 800 (100ms) */ |
04:31.04 | {tasker} | rtpheader[3] |= htonl((800)); |
04:31.07 | opus_ | yeah |
04:31.09 | PakiPenguin | anyone here uses linksys pap? |
04:31.10 | opus_ | change 800 to 5000 |
04:31.12 | {tasker} | ok |
04:31.20 | opus_ | which doesn't make sense, but worked for me. |
04:31.34 | opus_ | "hold down the fuckign dtmf key for 5 seconds you buggy piece of shit" |
04:31.52 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:33.51 | {tasker} | didn't work |
04:34.44 | {tasker} | I called myself and still couldn't hear DTMF tones in one direction |
04:34.49 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
04:34.52 | {tasker} | :( |
04:35.30 | opus_ | hmmm |
04:35.40 | opus_ | did you 'reload' or 'stop now' |
04:35.48 | {tasker} | I had to recompile, right? :) |
04:35.57 | opus_ | yea |
04:36.02 | {tasker} | lol |
04:36.14 | opus_ | i dunno, i tried to hot swap modules before |
04:36.15 | {tasker} | so reload or stop now are implied |
04:36.34 | {tasker} | ok |
04:36.57 | opus_ | but that area code is where its fucked, in my fuzzy logic brain here |
04:37.18 | opus_ | whoever patched it didn't even bother updating the comment |
04:37.26 | opus_ | wait nevermind |
04:37.29 | opus_ | i'm dum |
04:37.43 | {tasker} | lol |
04:37.47 | {tasker} | i'm dumber |
04:38.01 | {tasker} | because I didn't think of looking at that piece of code |
04:38.03 | *** join/#asterisk Moc_ (n=mochouin@229-198-0-72-ppp.3menatwork.com) |
04:38.44 | dr123 | has anyone tried 1.2 b1 is that out? |
04:38.55 | opus_ | what day was 1.0.5 released, i'll look at cvs diff |
04:39.12 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:39.14 | {tasker} | i don't remember off hand |
04:39.20 | {tasker} | dr123: yes, I have it running now |
04:39.27 | dr123 | anything new and interesting? |
04:39.45 | {tasker} | new, more reliable H.323 using up-to-date OH323 stack |
04:39.58 | {tasker} | some other new functionality |
04:40.00 | opus_ | < rtpheader[3] |= htonl((240)); |
04:40.17 | dr123 | I see |
04:40.19 | {tasker} | that's what it used to be? |
04:40.42 | dr123 | Is it possible w/ asterisk to use an IVR that is voice activated |
04:41.04 | *** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
04:41.25 | {tasker} | dr123: you mean have VAD in asterisk? |
04:41.35 | {tasker} | dr123: no, it's not implemented |
04:41.51 | Qwell | no, he means speach recognition |
04:41.55 | Qwell | (I think?) |
04:41.58 | Qwell | speech |
04:42.10 | PakiPenguin | anyone got this? |
04:42.12 | {tasker} | ah, ok |
04:42.13 | PakiPenguin | <PROTECTED> |
04:42.24 | Qwell | dr123: You can use sphinx, but it isn't great |
04:42.30 | dr123 | oh so that is called VAD i didnt realize it had its own acronymn |
04:42.30 | opus_ | <t3t:#interdictor-scanner2> ...sitting in front of the brigade. eta 3 sep. original (unt) capt. (unt) are in iraq? over. |
04:42.38 | dr123 | is that a feature that will be worked on |
04:42.38 | {tasker} | Palk: yes, you need to kill that |
04:42.43 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
04:42.52 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
04:43.17 | {tasker} | dr123: at the rate they're introducing bugs into asterisk, you'd better hope they don't |
04:43.23 | {tasker} | DTMF is broken |
04:43.24 | dr123 | hahahahaha |
04:43.27 | {tasker} | the most basic feature |
04:43.31 | dr123 | wow |
04:43.50 | dr123 | I am running the current stable on debian 3.0r1 and it is working great! |
04:43.53 | dr123 | no problems at all |
04:44.00 | {tasker} | Palk: in frame.c, that frame-dropping business on G729 needs to be disabled |
04:44.03 | dr123 | but I am not really using a lot of the fancy features |
04:44.25 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
04:44.30 | dr123 | Ok how about this: Has anyone heard a tentitive release date for the Cisco 7970 SIP image firmware? |
04:44.50 | {tasker} | dr123: try dialing into asterisk and transfer your call out to an external voicemail or dialing your cell-phone, then hit DTMF tones |
04:45.13 | JDLSpeedy | how do you setup a private caller? like someone calls me, if its a unknown under or what ever, it ask for its name, it would then contact me saying hey <person name> on the line, press 1 to accept the call, press 2 or hang up to deny the call |
04:45.40 | JDLSpeedy | unknown number* |
04:46.02 | {tasker} | JDL: check caller ID. if blank, jump to a voice prompt to record the user's name, dial out to you and play a prompt with an option |
04:46.22 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
04:46.43 | JDLSpeedy | ok {tasker}, I'll try that |
04:46.49 | JDLSpeedy | thx |
04:46.53 | {tasker} | np |
04:47.01 | {tasker} | it's a very simple script |
04:47.03 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:47.12 | *** join/#asterisk header (n=maxgluck@200.109.166.83) |
04:47.13 | opus_ | tasker. I don't think this will work, but I just check out asterisk 1.0.5's rtp.c into CVS HEAD and modified it to compile. |
04:47.16 | opus_ | want to try it? |
04:47.25 | {tasker} | sure |
04:47.39 | JDLSpeedy | you think its simple, LOL, im setting it up for the first time today |
04:47.51 | opus_ | or wait i'll just use pastebin |
04:48.20 | {tasker} | ok |
04:48.34 | {tasker} | JDL: you'll get the hang of it easily enough :) |
04:49.04 | header | what could be wrong if I receive only the header when calling from a 7940? If I call TO the 7940, the call gets through, but not the way around, I only see a header with 0 lines in asterisk...? |
04:49.10 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
04:49.16 | JDLSpeedy | I think asterisk is soooooooo cool |
04:49.41 | {tasker} | JDL: yeah, but it's riddled with bugs that don't get fixed because the developers don't use the bugged features |
04:49.44 | {tasker} | for example |
04:49.56 | {tasker} | dialing through asterisk to an external number that requires DTMF!! |
04:50.10 | {tasker} | caller -----> asterisk -----> company PBX |
04:50.15 | Qwell | {tasker}: May I rephrase your statement? |
04:50.18 | {tasker} | no joy sending DTMF tones |
04:50.23 | {tasker} | Qwell: please do |
04:50.26 | {tasker} | lol |
04:50.32 | Qwell | It's riddled with bugs that don't get fixed because the people who use the bugged features don't develop. |
04:50.40 | {tasker} | not true |
04:50.51 | opus_ | http://pastebin.ca/22025 |
04:51.02 | {tasker} | i've been busting my head on trying to find who fucked up the DTMF passthrough for a month now |
04:51.03 | opus_ | also run |
04:51.07 | opus_ | cvs update -D "01/23/2005" include/asterisk/channel_pvt.h |
04:51.09 | Grubs | Given there is a bugs database it does seem odd that some bugs dont get fixed |
04:51.22 | {tasker} | Grubs: agreed |
04:51.30 | {tasker} | opus: ok |
04:51.43 | opus_ | i don't think it will work, but it compiled |
04:51.43 | opus_ | hehe |
04:51.46 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
04:52.18 | opus_ | i just commented out the non-compiling parts. something i learned when porting ircII to dynix back in 1993 |
04:52.33 | {tasker} | Qwell: nobody knows asterisk better than the original developer and nobody has a better idea of where DTMF broke than the one who actually broke it |
04:52.34 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:52.47 | {tasker} | opus: hehe |
04:52.51 | {tasker} | opus: i'll give it a try |
04:52.54 | {tasker} | opus: hang on |
04:53.06 | JDLSpeedy | {tasker}: so your trying to have it set up so if you press a option, it would forward to a outside phone? |
04:53.21 | {tasker} | JDL: no |
04:53.30 | *** join/#asterisk shido6 (n=curtis@d57-87-253.home.cgocable.net) |
04:53.49 | {tasker} | JDL: if you use asterisk as a passthrough switch, where you dial in and it transfers your call to an outside line, you cannot send DTMF to that outside line |
04:54.01 | Jabroni | guys i got a lil' problem.. maybe someone can iluminate a bit my head... im using a A@H setup, created a IVR, and one of the options is to try to dial a number within the pbx, and link it up with another Zap Channel, but the problem is that channels never hang up :s |
04:54.26 | JDLSpeedy | DTMF? |
04:54.26 | Jabroni | exten => _.,1,Goto(from-internal,${EXTEN},1) |
04:54.29 | {tasker} | caller ----> asterisk -----> some PBX somewhere |
04:54.31 | Qwell | ~dtmf |
04:54.32 | jbot | DTMF: Dual Tone Multi-Frequency. The technical term describing Touch Tone dialing. Basically the combining of two tones, one low frequency and one high frequency. |
04:54.47 | opus_ | jabroni don't use h,hangup , make sure you actually have a hangup after you dial. don't use Dial(asdfas||r) |
04:54.49 | {tasker} | opus: no joy |
04:55.02 | {tasker} | opus: same problem |
04:55.16 | opus_ | well fuck |
04:55.22 | {tasker} | lol |
04:55.36 | {tasker} | it may have been broken somewhere related to catching tones |
04:55.42 | {tasker} | such as the '*' to hangup a call |
04:56.03 | {tasker} | you can disconnect an active call by hitting * during a call transfer |
04:56.29 | {tasker} | there were issues and they patched something at some point to be able to distinguish who pressed * |
04:56.42 | opus_ | haha |
04:56.51 | opus_ | yeah, i could just dial up a customer and hit * |
04:57.18 | {tasker} | shortly after that, DTMF stopped passing through from originating to terminating end |
04:57.18 | opus_ | tasker fuck |
04:57.21 | {tasker} | :( |
04:58.10 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
04:58.24 | opus_ | asterisk needs a unit tester |
04:58.35 | opus_ | that also does RTP |
04:58.36 | {tasker} | i have a whole carrier network at my disposal |
04:59.30 | opus_ | asterisk isn't disaster proof yet |
04:59.47 | {tasker} | it's fairly solid |
04:59.54 | {tasker} | we use it for prepaid card platforms |
05:00.02 | {tasker} | but that DTMF issue is a real ass-biter |
05:00.07 | opus_ | then why do i have so many problems. i guess its just my fucked up config |
05:00.13 | {tasker} | no, no |
05:00.18 | {tasker} | depends on the application |
05:00.23 | Qwell | You know, you could also pay to have a bug fixed. That jumps the priority pretty high |
05:00.31 | {tasker} | yeah |
05:00.33 | {tasker} | sure |
05:00.40 | {tasker} | people are buying asterisk licenses |
05:00.47 | {tasker} | i can't imagine they haven't the bug |
05:01.01 | {tasker} | because a fix would make its way into the tree |
05:01.04 | opus_ | if i had the money i would |
05:01.12 | {tasker} | most people are using asterisk as a PBX |
05:01.16 | {tasker} | not a passthrough switch |
05:01.20 | Qwell | asterisk is a pbx |
05:01.30 | {tasker} | asterisk is an IVR |
05:01.32 | {tasker} | not a PBX |
05:01.40 | {tasker} | it can be made to behave as a PBX |
05:01.52 | {tasker} | but it's nothing more than an engine that interprets instructions |
05:01.56 | Qwell | Asterisk? - the Open Source PBX! |
05:01.58 | Juggie | asterisk is a pbx |
05:02.05 | {tasker} | yes, yes, that's the name they gave it |
05:02.09 | {tasker} | but technically it is not |
05:02.10 | Qwell | because thats what it is |
05:02.21 | Qwell | I'm not even going to argue that point. |
05:02.21 | {tasker} | without extensions.conf instructing a PBX app, it does nothing |
05:02.26 | opus_ | what would be a better name description for it |
05:02.35 | {tasker} | Asterisk IVR |
05:02.39 | Qwell | {tasker}: it wouldn't be an IVR without extensions.conf |
05:02.41 | Juggie | i dont know of any ivr's that register peers, convert between sip/h323/iax/zaptel |
05:02.46 | {tasker} | not true |
05:02.48 | Juggie | yes |
05:02.54 | Juggie | no extensions.conf and its nothing |
05:02.56 | {tasker} | an IVR implies it can do whatever |
05:03.03 | {tasker} | a PBX is specific |
05:03.12 | {tasker} | an IVR can be made to do anything interactive |
05:03.14 | Qwell | This is a stupid argument |
05:03.15 | Juggie | its a pbx on drugs |
05:03.19 | {tasker} | lol |
05:03.24 | Juggie | its a pbx + ivr |
05:03.25 | {tasker} | im not arguing |
05:03.26 | Juggie | not a ivr |
05:03.33 | {tasker} | do you know what pbx means? |
05:03.35 | Qwell | a pbx is an ivr |
05:03.38 | Qwell | an ivr is not a pbx |
05:03.43 | {tasker} | public branch eXchange |
05:03.45 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
05:03.55 | Juggie | tasker, that term is null these days |
05:04.02 | {tasker} | an IVR is a blank system that accepts programmed instructions ;) |
05:04.03 | Juggie | since you can also have private branch eXchanges |
05:04.04 | Qwell | so, me picking up a phone and talking to you is "Interactive Voice Recording"? |
05:04.09 | Netgeeks_ | Interactive Voice Response doesn't describe asterisk any more than Private Branch Exchange does |
05:04.22 | {tasker} | i've been developing interactive voice response systems since Dialogic first started in the early 90's |
05:04.23 | {tasker} | lol |
05:04.23 | Qwell | is it recording, or response? |
05:04.26 | Qwell | ~ivr |
05:04.32 | Juggie | {tasker}, me too... |
05:04.35 | {tasker} | response |
05:04.48 | Juggie | i started writing dialogic in like 99 |
05:04.50 | {tasker} | an IVR is a very generic term |
05:04.56 | {tasker} | I started Dialogic in 93 |
05:05.08 | Juggie | like Qwell said, a pbx is an ivr but an ivr isnt a pbx |
05:05.08 | {tasker} | writing Dialogic apps, i mean |
05:05.13 | {tasker} | nope |
05:05.16 | {tasker} | the other way around |
05:05.24 | Juggie | no thats right |
05:05.24 | {tasker} | an IVR can be made to behave as a PBX |
05:05.26 | opus_ | er}> writing Dialogic apps, i mean |
05:05.32 | Juggie | no it cannot |
05:05.37 | Juggie | an ivr is not a switch |
05:05.46 | {tasker} | a PBX will not become a prepaid app, or an informational retrieval service, etc |
05:05.51 | {tasker} | no, but it can switch calls |
05:05.53 | opus_ | [H.323 0004989]: was just updated tasker, the guy saids he patched h323 and fixed it and will soon past a patch. |
05:06.03 | {tasker} | opus: KEWL! |
05:06.07 | Netgeeks_ | asterisk is currently more a softswitch than anything else |
05:06.07 | {tasker} | opus: thanks for the update |
05:06.08 | Juggie | {tasker}, have you looked at pbx's in the last 5 years? |
05:06.17 | Juggie | no one makes hardware based switches anymore |
05:06.20 | {tasker} | Netgeeks: that's a better description |
05:06.29 | {tasker} | Netgeeks: but a softswitch is not necessarily interactive :) |
05:06.47 | Juggie | tasker, when did you last put your hands on a mitel, cisco, nortel pbx |
05:06.51 | Juggie | they are all softswitches |
05:06.54 | {tasker} | every day |
05:06.56 | {tasker} | unfortunately |
05:06.57 | Juggie | well me too |
05:07.10 | Juggie | everything is soft these days... asterisk is a pbx like any other |
05:07.12 | {tasker} | we have Cisco AS5xxx series all over our network around the world |
05:07.13 | Netgeeks_ | but softswitch is a much wider description, much so that I think it applies more so than the other two |
05:07.18 | Juggie | hell, the original cisco crap ran windows |
05:07.23 | Juggie | until they ported it to linux |
05:07.25 | {tasker} | just the callmanager |
05:07.34 | Juggie | yah, the original call manager ran windows |
05:07.36 | Juggie | now its linux |
05:07.43 | Juggie | mitel released a softpbx running windows |
05:07.50 | {tasker} | blech |
05:07.50 | opus_ | do you mean they don't use any DSP's |
05:07.51 | opus_ | ? |
05:08.08 | Juggie | they do, i know mitel has some dsp's on their cards |
05:08.12 | Juggie | not sure about the cisco call manager gear |
05:08.13 | opus_ | i recently installed a shoretel switch and it was all FPGAs/DSPs |
05:08.30 | Juggie | opus, was it an ip switch? |
05:08.36 | opus_ | yeah |
05:08.38 | Juggie | or digital phones/analog |
05:08.51 | Juggie | what i see around here is all in like 2U racks |
05:08.56 | opus_ | of course. it uses the polycom IP-500 rebadged as the "shorepoint 100" |
05:09.10 | Juggie | it has dedicated hardware, but alot of it is software as well |
05:09.20 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
05:09.22 | opus_ | windows 2000 box to do its voicemail |
05:09.34 | opus_ | however you can reboot that box and no lines will go down |
05:09.48 | {tasker} | kind of like Excel |
05:09.55 | {tasker} | the call manager runs off a separate box |
05:09.59 | {tasker} | be it under windows or whatever |
05:10.14 | {tasker} | *sigh* |
05:10.15 | opus_ | when you say 'call manager' what do you mean |
05:10.26 | {tasker} | call setup / teardown |
05:10.29 | {tasker} | switching |
05:10.29 | opus_ | shoretel world = call manager is desktop CAD software |
05:10.32 | {tasker} | prompts |
05:10.32 | {tasker} | etc |
05:10.50 | Grubs | dudes - you could have had that bug fixed by now if your were not bickering over semantics ;) |
05:10.51 | opus_ | cisco world = call manager is IP-PBX? |
05:11.02 | Juggie | yes callmanager = ip pbx |
05:11.09 | {tasker} | opus: nooooooo |
05:11.10 | {tasker} | lol |
05:11.21 | {tasker} | ok, ok, have it your way |
05:11.47 | opus_ | shoretel is nice, has computer desktop helpers etc |
05:11.53 | {tasker} | in Excel, you can create custom apps and send instructions to the switch via TCP/IP |
05:11.59 | opus_ | .asp admin interface |
05:12.04 | {tasker} | they're hex code commands |
05:12.18 | opus_ | haha. |
05:12.26 | {tasker} | Cisco uses TCL that it directly interprets for custom apps |
05:12.34 | {tasker} | frankly, that's a butt ugly language |
05:12.41 | {tasker} | but, hey, their choice |
05:12.41 | Juggie | tcl is gross |
05:12.42 | {tasker} | lol |
05:12.49 | Juggie | * can use any scripting or programming language |
05:12.52 | Juggie | anything you choose |
05:13.04 | Juggie | i personally use php :P |
05:13.06 | {tasker} | yes, but the mechanics is ugly |
05:13.08 | {tasker} | example |
05:13.15 | opus_ | when I look at the local yellow pages under 'phone equipment' everyone is advertising IP-based phones, what the heck are the adverting |
05:13.19 | {tasker} | Dial() locks your app until it's complete |
05:13.25 | {tasker} | but |
05:13.35 | {tasker} | I tried throwing Dial() into a thread so my app can continue |
05:13.45 | opus_ | Like 3com VOIP |
05:13.48 | {tasker} | and I would check DIALSTATUS to get the line status |
05:13.54 | shido6 | thats what the manager is for |
05:13.55 | {tasker} | and it blocks there :) |
05:13.55 | {tasker} | lol |
05:14.01 | {tasker} | until Dial is done |
05:14.07 | opus_ | its not really voice over an IP stack. I call it VoCAT5 |
05:14.09 | Juggie | then feel free to fix it :P |
05:14.16 | {tasker} | I tried |
05:14.28 | {tasker} | unfortunately, the whole pbx_xxxx interface is sequential |
05:14.32 | Juggie | why would you want to move before a dial was done |
05:14.36 | {tasker} | it needs to much of a rewrite |
05:14.43 | shido6 | use the manager. |
05:14.44 | Juggie | what would be the scenario where that was required |
05:14.51 | opus_ | what technology does Nortel use for VOIP? |
05:14.56 | {tasker} | if you needed your app to perform other tasks during an active call |
05:15.05 | {tasker} | i.e. account updates, etc |
05:15.08 | shido6 | then use the manager, taskbar |
05:15.16 | shido6 | tasker |
05:15.28 | {tasker} | there's something new for me |
05:15.32 | {tasker} | how? |
05:15.53 | *** join/#asterisk twisted[home] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
05:15.53 | *** mode/#asterisk [+o twisted[home]] by ChanServ |
05:15.54 | {tasker} | wouldn't I have to retrieve the channel name and open a new STDIN? |
05:16.01 | twisted[home] | wheee |
05:16.03 | twisted[home] | there are 3 of me |
05:16.05 | {tasker} | doesn't that become resource intensive? |
05:16.10 | Juggie | tasker, there are ways you could do that, however i'm not sure what you would want to do that you coudnt do before or after a call |
05:16.23 | {tasker} | furthermore, there's an open bug about having too many manager sessions open |
05:16.32 | twisted[home] | hmm |
05:16.35 | twisted[home] | bkw_, you around? |
05:16.38 | opus_ | tasker use the manager proxy |
05:16.58 | {tasker} | sequentially process requests? |
05:17.02 | {tasker} | that would work |
05:17.04 | opus_ | how many manager api connections until it crashes? |
05:17.15 | {tasker} | don't remember |
05:17.21 | {tasker} | but it wasn't many |
05:17.25 | Juggie | tasker, one solution that comes to mind, use php, do spawn your thread to do what you want, then when your dial completes, signal the thread that its time to move on. |
05:17.26 | opus_ | hmm. that could be aproblem for me shit thanks for letting me know |
05:17.33 | shido6 | using Asterisk::Manager |
05:17.38 | shido6 | perl |
05:17.44 | {tasker} | kewl |
05:18.00 | {tasker} | so I can throw Dial() into a thread and use Manager to retrieve channel status? |
05:18.04 | shido6 | use strict; |
05:18.08 | shido6 | use CGI' |
05:18.15 | shido6 | use Asterisk::Manager; |
05:18.35 | shido6 | sendcommand |
05:18.49 | {tasker} | ok |
05:19.01 | {tasker} | i'll try it |
05:19.14 | Juggie | tasker, do the dial in the main thread, and use a worker thread to do what you want |
05:19.20 | Juggie | do it in the reverse of how your thinking |
05:19.27 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
05:19.32 | {tasker} | Juggie: that's also a good idea |
05:19.42 | {tasker} | Juggie: then have a flag to indicate call completion |
05:19.47 | {tasker} | awesome |
05:19.49 | {tasker} | thanks |
05:19.52 | Juggie | yes, signal your thread |
05:20.01 | Juggie | to let it know if the call is setup or not, etc. |
05:20.03 | {tasker} | my brain is too fried from all the projects and the lack of vacation |
05:20.12 | {tasker} | I haven't gone on vacation since 2002 |
05:20.15 | {tasker} | I need a break |
05:20.16 | Juggie | hah :P |
05:20.16 | {tasker} | *sigh* |
05:20.20 | Juggie | welcome to my world |
05:20.23 | {tasker} | lol |
05:20.24 | Juggie | my only vacation is visiting family |
05:20.29 | Juggie | not really a vacation |
05:20.33 | {tasker} | no kidding |
05:20.39 | {tasker} | visiting family != vacation |
05:20.40 | Juggie | what language are you going to attempt this in? |
05:20.44 | {tasker} | Perl |
05:20.48 | Juggie | k |
05:20.59 | Juggie | i was gonna say with php you'll need a custom compile most likely to enable thread control |
05:20.59 | {tasker} | it's a whole lot faster to prototype |
05:21.14 | {tasker} | i thought of PHP but the threading is the issue |
05:21.21 | Juggie | here is also a res_php |
05:21.23 | Juggie | fyi |
05:21.32 | Juggie | which means no using stdin/out io etc |
05:21.33 | {tasker} | kewl |
05:21.41 | Juggie | no using a socket for chatter |
05:21.49 | Juggie | and a res_perl |
05:21.51 | {tasker} | i looked at res_perl but it looks unstable |
05:21.51 | Juggie | for that matter |
05:21.56 | Juggie | it should be ok |
05:22.00 | Juggie | check www.pbxfreeware.org |
05:22.12 | Juggie | bug bkw_ if there are problems with res_perl |
05:22.24 | {tasker} | k |
05:22.31 | {tasker} | bk = burger king |
05:22.34 | {tasker} | i'm hungry |
05:22.44 | *** join/#asterisk knoppix (n=knoppix@cm52.theta17.maxonline.com.sg) |
05:22.47 | {tasker} | i think i'm gonna bug my wife and take her out for a midnight snack |
05:22.54 | Juggie | hah |
05:22.57 | Juggie | burger king |
05:22.58 | Juggie | eugh |
05:22.58 | {tasker} | lol |
05:23.00 | Juggie | gross |
05:23.01 | {tasker} | no, no |
05:23.03 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
05:23.04 | {tasker} | the burger idea |
05:23.07 | Juggie | mcdonalds is better then buger king |
05:23.09 | {tasker} | Carl's Junior is my fav place |
05:23.17 | {tasker} | but there aren't any here |
05:23.26 | Juggie | rotten rons |
05:23.34 | {tasker} | BK used to be good when they flame broiled their patties |
05:23.37 | {tasker} | now they boil them |
05:23.43 | {tasker} | in that stupid drawer |
05:23.53 | {tasker} | who the hell boils their burgers? |
05:23.59 | {tasker} | and still claims flame broiled |
05:24.08 | {tasker} | BK does |
05:24.26 | {tasker} | I haven't seen a grill mark on their burgers in nearly half a decade |
05:24.28 | Juggie | http://www.pbxfreeware.org/res_perl.tgz |
05:24.34 | Juggie | thats the latest res perl |
05:24.34 | {tasker} | thanks, J |
05:24.35 | Juggie | check it out |
05:24.53 | {tasker} | so am I |
05:24.56 | Juggie | i think i'll get a ham samwhich |
05:25.00 | Juggie | blackforest ham |
05:25.00 | Juggie | umm |
05:25.08 | {tasker} | yum |
05:25.11 | {tasker} | that sounds good, too |
05:25.14 | Juggie | yah |
05:25.21 | Juggie | i have this spicey mustard |
05:25.23 | Juggie | thats really good |
05:25.36 | {tasker} | man, i'm coming to your place |
05:25.46 | {tasker} | i have nothing in the fridge |
05:25.53 | {tasker} | we eat out too much |
05:26.10 | {tasker} | and never fill that fridge |
05:26.29 | Juggie | hah |
05:26.36 | Juggie | i filled it because i am trying to save money |
05:26.45 | {tasker} | that's the right idea |
05:26.48 | {tasker} | and save your health |
05:26.57 | {tasker} | have you seen Supersize Me? |
05:26.59 | {tasker} | lol |
05:27.00 | Juggie | no |
05:27.00 | Grubs | Can asterisk make any use of dual CPU's? |
05:27.06 | Juggie | but i did have rotten rons today |
05:27.13 | {tasker} | how is that? |
05:27.13 | Juggie | grubs, yes |
05:27.20 | {tasker} | good burgers? |
05:27.21 | Juggie | rotten rons = mcdonalds |
05:27.24 | {tasker} | lol |
05:27.26 | Juggie | Rotten Ronnies |
05:27.28 | Juggie | get it :P |
05:27.36 | {tasker} | LOL |
05:27.41 | Juggie | just like |
05:27.42 | Juggie | KFC |
05:27.44 | Juggie | = |
05:27.47 | Grubs | thinking of a nice slow dual rackmount server for *. Dual 500 MHz ample? |
05:27.48 | Juggie | The dirty bird |
05:27.52 | {tasker} | lol |
05:28.06 | shido6 | grubs |
05:28.11 | {tasker} | Grubs: as long as you don't do a lot of transcoding |
05:28.19 | Juggie | grubs, depends on call volume and codecs |
05:28.25 | Juggie | if you are using like 1 t1 card or less |
05:28.29 | Juggie | you should be fine |
05:28.29 | {tasker} | Grubs: on our all software IVR setups, we even prerecord prompts in G729 |
05:28.40 | Grubs | nice idea |
05:28.51 | Grubs | I prefer iLBC :) |
05:28.55 | {tasker} | im able to get 1800 simultaneous passthrough calls on a dual Xeon |
05:29.01 | Juggie | not many hard phones support ilbc |
05:29.10 | Juggie | if any |
05:29.13 | Juggie | i've never seen one |
05:29.17 | {tasker} | speex is a nice codec. I wish it would gain mainstream support |
05:29.25 | Grubs | only 1-3 simultaneous calls here though. |
05:29.32 | Juggie | then you are best kind |
05:29.46 | {tasker} | ITU unfortunately has too much invested in making royalties |
05:29.50 | opus_ | i perfer ultrawideband :) |
05:30.20 | {tasker} | sounds like a heavy man's underwear |
05:30.21 | Grubs | I am sure I saw an ATA or two with iLBC support. |
05:30.27 | Juggie | has anyone been paying attention to the blog from neworleans |
05:30.38 | Juggie | the NOC thats running in new orleans and has been running the entire time |
05:30.39 | shido6 | which blog ? |
05:30.43 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
05:30.45 | {tasker} | wow |
05:30.45 | Qwell | directnic |
05:30.55 | shido6 | url? |
05:31.00 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
05:31.08 | opus_ | i'm cohosting there |
05:31.16 | Qwell | http://www.livejournal.com/users/interdictor/ |
05:31.18 | Juggie | i bet your servers up ;P |
05:31.18 | opus_ | i wonder how much :)_ |
05:31.31 | Juggie | i must say i am truly impressed |
05:31.34 | Juggie | the city is underwater |
05:31.39 | opus_ | are you listening to the scanner? |
05:31.39 | Juggie | and these guys are on irc chatting |
05:31.41 | Juggie | yes |
05:31.52 | opus_ | theres live transcripts on #interdictor-scanner1 and scanner2 |
05:32.00 | Qwell | opus_: freenode? |
05:32.03 | opus_ | yeah |
05:32.08 | *** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
05:32.13 | opus_ | <nop_:#interdictor-scanner2> [better than "only fix wing aircraft allowed inbound" if you ask me] |
05:32.22 | *** join/#asterisk tabo23 (n=acura0@S010600119503044b.vc.shawcable.net) |
05:33.02 | Juggie | they are on a quest to get deisel now |
05:33.05 | Juggie | and get another oc3 up |
05:33.09 | Juggie | they are down to one peer |
05:36.30 | *** join/#asterisk cianhughes (n=cian@cian.ws) |
05:43.35 | {tasker} | OMG! |
05:43.41 | {tasker} | I fixed the DTMF problem |
05:43.47 | {tasker} | lol |
05:43.51 | {tasker} | how stupidly trivial |
05:44.23 | {tasker} | opus: you there? |
05:46.07 | hugo-v6 | ok asterisk-phones rock the hood. |
05:46.53 | {tasker} | if anyone is interested in the fix to the DTMF passthrough problem, go here: |
05:46.54 | {tasker} | http://pastebin.com/353386 |
05:49.37 | *** join/#asterisk Rowters (n=SilverDr@dsl-201-129-89-96.prod-infinitum.com.mx) |
05:51.08 | {tasker} | 'nite all |
05:51.14 | {tasker} | c y'all next time |
05:51.16 | *** part/#asterisk {tasker} (i=sdfsddad@modemcable064.169-203-24.mc.videotron.ca) |
05:52.05 | *** part/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
05:55.19 | *** join/#asterisk azrishahril (n=azrishah@60.50.193.179) |
05:59.08 | hugo-v6 | did isay asterisk-phones? well im still sleepy, meant elmeg-phones rock da hood. |
05:59.36 | hugo-v6 | and btw gd morning *yawn* |
06:00.03 | opus_ | hmmm |
06:00.13 | opus_ | instead of rebuilding all analog lines in NO, maybe they could use VOIP |
06:00.15 | opus_ | :) |
06:00.47 | hugo-v6 | opus_: i wont use analog in the house anymore. i deplay voip-phones :> |
06:00.55 | hugo-v6 | roxx. |
06:01.21 | hugo-v6 | s/. i/. i will/ |
06:02.00 | *** join/#asterisk coppice (n=chatzill@167.203.17.210.dyn.pacific.net.hk) |
06:02.33 | hugo-v6 | is there a way to save the "regexten" entry in a phone-section in sip.conf? |
06:04.17 | *** join/#asterisk [hC] (n=hardcore@c-24-127-192-210.hsd1.fl.comcast.net) |
06:04.56 | hugo-v6 | well found something, but i cant call i.e. "myuser" as in example on voip-info *think* at at least not from an isdn-phone. |
06:06.18 | *** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk) |
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06:14.08 | konrads | Hello |
06:14.13 | gordonjcp | morning |
06:14.18 | konrads | I am having problems with SIP->H323 cals |
06:14.25 | konrads | NOTICE[9301] src/chan_h323.c: Don't know how to deal with mode 0 |
06:14.26 | konrads | x40 (slin) |
06:14.39 | konrads | it attempts some sort of native bridge between stip and h323 |
06:14.48 | konrads | i guess it tries to send same rtp payload |
06:14.56 | konrads | how do I force re-encoding? |
06:20.35 | *** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net) |
06:23.04 | *** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net) |
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06:34.11 | azrishahril | konrads: |
06:35.15 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
06:36.54 | hugo-v6 | puh. now going to get a box of tobacco. l8r |
06:37.53 | X-Rob | 'box'? |
06:37.57 | X-Rob | you 'mericans are wierd. |
06:38.39 | glm2k | X-Rob: tobacco does come in boxes |
06:38.51 | glm2k | er, do |
06:43.04 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
06:45.58 | hugo-v6 | X-Rob: .de ;) not .us. well i roll from hand. so u can buy a bag or a box ;) |
06:51.29 | X-Rob | what size is a 'box' of tobacco? |
06:51.38 | X-Rob | when you say 'box' I would assume 'carton' |
06:51.44 | X-Rob | eg, 10kg or so |
06:53.57 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
06:53.57 | hugo-v6 | x-rob: look after the ranslation ;) |
06:54.02 | hugo-v6 | in my case it means 120g |
06:54.05 | hugo-v6 | grains |
06:54.48 | hugo-v6 | grains? am i stupid? |
06:55.08 | hugo-v6 | 120 gram which are about 4 ounces |
06:55.35 | hugo-v6 | so no on a walk with my dog. l8r. |
06:58.35 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
06:59.27 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
07:01.07 | X-Rob | I must say that this song I've been listening to (on repeat) all day is damn good. |
07:01.24 | PakiPenguin | X-Rob, which one? |
07:01.29 | X-Rob | 'Cobrastyle' by Teddybears Sthlm |
07:01.35 | X-Rob | Was just typing the same |
07:01.36 | X-Rob | name |
07:01.49 | X-Rob | Definately 'Sthlm' |
07:02.09 | X-Rob | If you've seen the heineken advertisment where they're all breakdancing, it's the background music to that |
07:02.20 | X-Rob | 'pop and lock' I've seen it described as |
07:02.48 | X-Rob | and if you haven't seen that |
07:02.51 | X-Rob | I'll upload it somewhere 8) |
07:03.34 | X-Rob | If you have limewire running, it's worth getting 8) |
07:17.08 | *** join/#asterisk coppice (n=chatzill@167.203.17.210.dyn.pacific.net.hk) |
07:23.14 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
07:24.13 | *** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il) |
07:27.36 | *** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
07:34.18 | *** join/#asterisk criptos (n=criptos@201.135.121.4) |
07:34.53 | criptos | hello everyone! :) |
07:34.55 | *** join/#asterisk file[lap1op] (n=jcolp@mctnnbsa30w-156034035106.nb.aliant.net) |
07:38.35 | *** join/#asterisk pbxbart__ (n=pbxbart@p54B00652.dip0.t-ipconnect.de) |
07:38.40 | *** part/#asterisk pbxbart__ (n=pbxbart@p54B00652.dip0.t-ipconnect.de) |
07:39.09 | *** join/#asterisk Pkunk (n=Pkunkage@mbbs.munnabhai.info) |
07:39.48 | PakiPenguin | Pkunk: awesome domain |
07:41.03 | X-Rob | What's the relevance of munnabhai.info ? |
07:41.37 | PakiPenguin | munnabhai mbbs is a movie :) |
07:41.39 | PakiPenguin | haha |
07:41.50 | *** join/#asterisk samborambo (n=samboram@219-89-7-115.dialup.xtra.co.nz) |
07:41.51 | glm2k | another bollywood production? |
07:42.00 | PakiPenguin | yeah |
07:42.04 | X-Rob | aaah |
07:42.16 | PakiPenguin | anyone here has seen proof the movie? |
07:42.18 | samborambo | hello all :) |
07:42.20 | glm2k | hmm, i should try to find it then, might be fun |
07:43.35 | samborambo | could someone please help me with a problem I'm having with Amp? |
07:43.48 | X-Rob | I was just going through all the old PC's here, and I found a P4 2.4ghz with a 40g HDD and 256M |
07:43.53 | X-Rob | IT seems to work too |
07:44.15 | X-Rob | *woo* |
07:44.19 | glm2k | nice |
07:45.00 | PakiPenguin | what |
07:45.03 | PakiPenguin | :o |
07:45.06 | PakiPenguin | X-Rob, sendo it to me :p |
07:45.08 | PakiPenguin | haha |
07:45.12 | PakiPenguin | you dont need it anyways |
07:45.16 | *** part/#asterisk criptos (n=criptos@201.135.121.4) |
07:45.43 | samborambo | From what I understand, Apache2 needs to run as user "asterisk", however, asterisk is not the only thing running on that server, and so those other services need to be www-data |
07:46.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
07:46.37 | samborambo | anyone got any ideas? |
07:47.26 | Byte | X-Rob: that's an "old" pc? |
07:50.32 | X-Rob | Byte - it was sitting in the 'old' pile, and it had a rusted back panel thingo (around the parrallel port, keyboard, etc, the little metal clip in) |
07:50.49 | X-Rob | I'm going through 'em all, stripping good bits out of 'em, and binning the rest. |
07:51.08 | X-Rob | samborambo - Are you talking about AMP? |
07:53.46 | gordonjcp | X-Rob: cool |
07:53.54 | X-Rob | It's got two passes of memtest. Woo. I think it's time for the old Piii/500 gateway to be upgraded! |
07:54.05 | zedkatuf | samborambo: afaik, AMP uses apache1.3 rather than apache2....is there any way you could get the other services to run under apache2 at the same time? |
07:54.08 | gordonjcp | samborambo: can't it be in the www-data group? |
07:54.53 | X-Rob | Ah, he _is_ talking about AMP (reads the top of the screen) |
07:55.39 | X-Rob | zedkatuf - there's no _real_ requirement for 1.3 |
07:55.53 | *** join/#asterisk GaryH (n=ghawkins@gromit.garysoft.co.uk) |
07:56.02 | X-Rob | We've just committed some patches to make it work with php5 on FC4, and I think FC4 uses apache 2.0, doesn't it? |
07:58.29 | zedkatuf | X-Rob: I agree..my suggestion was just a workaround...not convinced it's the best way to do it though! |
07:58.44 | X-Rob | It can be worked around. |
07:58.56 | X-Rob | When we release AMP2 I'll have to figure out a way to make the installation a lot easier |
07:59.04 | X-Rob | I really don't like having to run apache as asterisk |
08:03.00 | *** join/#asterisk MikeJ__ (n=ircatjer@adsl-68-74-8-29.dsl.sfldmi.ameritech.net) |
08:04.06 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
08:05.06 | samborambo | sorry X-Rob.....was away looking for more answers on the web. Yeah, I'm talking about AMP |
08:05.24 | *** join/#asterisk GaryH (n=ghawkins@gromit.garysoft.co.uk) |
08:07.03 | samborambo | gordonjcp: you think run apache as www-data group and asterisk user? |
08:07.16 | *** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim) |
08:08.20 | samborambo | I thought there might be a workaround by running a virtual server in apache just for amp |
08:08.39 | samborambo | though, I'm no apache expert |
08:08.47 | GaryH | Hi, has anyone tested chan_capi with asterisk 1.2.0beta1 yet? |
08:11.23 | *** join/#asterisk VoIPMasta (n=John@dsl-200-95-42-39.prod-infinitum.com.mx) |
08:11.43 | VoIPMasta | Hi, I'm having some problems with a dialplan... is someone available and willing to help? |
08:11.54 | samborambo | X-Rob: is there any documentation on this? |
08:12.30 | *** join/#asterisk zoo (i=nobody@ip-172-16.travedsl.de) |
08:17.09 | VoIPMasta | When I try to forward a call it just rings once and then is cut off |
08:17.49 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
08:17.52 | *** join/#asterisk mastix (n=mastix@ip-85-160-13-68.eurotel.cz) |
08:18.09 | mastix | hi |
08:18.14 | VoIPMasta | if anyone can help, the logs and confs are in http://pastebin.com/353419 |
08:18.39 | *** join/#asterisk atmel (n=vlad@ip68-4-101-199.oc.oc.cox.net) |
08:21.04 | mastix | hi does anyone know where can i find some help for configuration of h.323 |
08:21.33 | mastix | would be great if some manual exists |
08:22.31 | VoIPMasta | mastix: pm me, I might be able to help you |
08:23.58 | mastix | sorry what does it mean "pm me"< |
08:24.01 | mastix | ? |
08:24.10 | VoIPMasta | send a query/private message/msg/etc |
08:34.25 | VoIPMasta | anyone here has experience forwarding calls from a DID provider to a SIP extension? |
08:36.25 | *** join/#asterisk Tili (n=Tili@202-133-67-168-dialup.sat.net.pk) |
08:37.46 | PakiPenguin | hey Tili |
08:37.58 | PakiPenguin | VoIPMasta, i've tried it and it works |
08:38.00 | Tili | hi PakiPenguin |
08:38.03 | PakiPenguin | what do you exactly want to do? |
08:38.06 | VoIPMasta | great :) |
08:38.16 | PakiPenguin | i mean DID -> SIP :) |
08:38.21 | VoIPMasta | ohh |
08:38.26 | VoIPMasta | can you help me a little bit here? |
08:38.30 | PakiPenguin | yeah sure why not |
08:38.35 | VoIPMasta | I thought that you were talking about your h323 issue |
08:38.39 | PakiPenguin | what are youe xactly trying to do |
08:38.43 | PakiPenguin | haha i am still compiling pwlib |
08:38.57 | VoIPMasta | I just purchased a DID from a DID provider, they deliver the DID to a regular softphone |
08:39.12 | PakiPenguin | okay |
08:39.15 | VoIPMasta | so I added a register => line in my sip.conf, so that my asterisk box acts as a client |
08:39.20 | PakiPenguin | fine |
08:39.35 | VoIPMasta | my asterisk server registers with the provider's gateway |
08:39.42 | PakiPenguin | so when you call your did , what happens? |
08:39.52 | VoIPMasta | now, when I dial the DID from my cellphone, it just rings once and then hangs up |
08:40.03 | VoIPMasta | but my cellphone keeps displaying the call as active |
08:40.27 | VoIPMasta | http://pastebin.com/353419 <-- that will give you an idea of what's going on |
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08:49.34 | zedkatuf | offtopic Q: If I want to flush all iptables rules, I simply do iptables -F ..is this correct? |
08:50.26 | zedkatuf | (my problem atm is that I can't access my asterisk box..ie no pinging possible...I'm wondering if it's as a result as of installing shorewall on my * box :P) |
08:51.01 | zedkatuf | (so I've stopped shorewall & done iptables -F but still no access..the phyiscal eth connectors are ok) |
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09:04.00 | [Airwolf] | Is it also possible to somehow script my sip configuration ? So that I don't have to make the same entry 30 times ? |
09:04.27 | Byte | ? |
09:07.53 | [Airwolf] | I have a few sip configurations now, but they are all the same. So I just want one configurations and then just specify the number and password |
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09:14.21 | newl | [Airwolf]: you can use a macro. |
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09:23.12 | [Airwolf] | newl, just the same as I use in my extensions.conf ? |
09:23.21 | [Airwolf] | I didn't knew that was possible, tnx |
09:32.47 | newl | [Airwolf]: yep. I use a macro for dial actions, makes things much cleaner with larger dial plans. |
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09:37.02 | Assid | okay .. i have this funny issue.. my format in cdr_custom.conf but.. same thing in cdr.conf doesnt get updated (in the database) |
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09:58.13 | PakiPenguin | hmm |
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10:06.25 | Assid | okay i wanna change the format of cdr_pgsql |
10:06.34 | Assid | do i just add a format => line ? |
10:06.45 | Assid | or does it go into another conf file? |
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10:08.49 | r0m | good morning! |
10:13.04 | *** part/#asterisk zoo (i=nobody@ip-172-16.travedsl.de) |
10:21.43 | *** join/#asterisk phoe (n=phoe@82.171.44.25) |
10:22.38 | phoe | good afternoon |
10:32.25 | coppice | good evening |
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10:35.10 | hat | hi, i am new to asterisk but we need to deploy a not too small system support web/sms callback for 5000 users. Who can suggest the hardware configuration to me? |
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10:54.23 | jeffik | anybody know how to use meet-me? |
10:54.53 | wasim | jeffik: how hard can it be? |
10:55.22 | jeffik | wasim: i have never done it and have been reading and can not figure it out |
10:55.35 | phoe | wasim: hard if you dont know what meet-me is :) |
10:56.03 | jeffik | wasim: as i understand it's a conference bridge users can dial in to |
10:56.09 | wasim | jeffik: it is indeed |
10:57.02 | jeffik | ok, so may i ask you how to set up a room? for example my extension is 1500 as i understand i should be a blre to dial 81500 and create a conferene room |
10:57.49 | wasim | in meetme.conf put conf => 1500 |
10:57.59 | wasim | in extensions.conf put exten => 1500,1,Meetme(1500) |
10:58.01 | wasim | dial 1500 |
10:58.56 | wasim | don't forge to reload, do a show application meetme for more options, and read the wiki |
10:59.27 | wasim | oh, AND you need a timing interface like a zapcard or ztdummy |
11:00.06 | jeffik | ok i'll read more thanks |
11:00.38 | phoe | Asterisk looks very nice, but it's all very complicated to me :P |
11:00.48 | phoe | gonna start using is this monday |
11:00.54 | wasim | why wait? |
11:01.18 | phoe | lol, tbh I did a little bit at my home pc |
11:01.26 | phoe | it's for work experience, and that starts monday |
11:01.51 | phoe | never done anything before with VoIP or Linux. it's all very new to me :P |
11:02.00 | phoe | only managed to install it so far hehe |
11:02.45 | newl | heh wicked, Meetme worked. That was simple. I've been wondering the same for ages (knew what it was and where to look for config, but never did anything about it until now). |
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11:08.41 | Alcee | 19h30 ? |
11:08.54 | Alcee | sorry .... bad window |
11:11.56 | visik7_ | is there a way to customize the call parking commands? like change the # or the behaviour of the app ? |
11:20.15 | wasim | visik7_: features.conf |
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11:42.26 | queuetue | I have a broadvoice account inbound to Asterisk. BV gives me a second number to dial into that same account - is there any way to makse * act differently when that call comes in on the alternate number? |
11:42.37 | konrads | queuetue: yeah |
11:42.59 | konrads | queuetue: if the broadvoice signals on what number it is incoming (that a sip/voip sutff?) |
11:43.08 | konrads | you can do a [broadvoice-in] |
11:43.14 | konrads | and exten => number1,rules... |
11:43.24 | konrads | exten => number2,rules... |
11:43.44 | queuetue | konrads: It is sip. I'm not sure what "signals" means, though. |
11:43.55 | konrads | queuetue: in your case it is fine :) |
11:44.12 | konrads | queuetue: just do as I said, write two extensions |
11:44.34 | konrads | exten => SipNr1,Dial(CAPI/contr1/b123) |
11:44.44 | konrads | exten => SipNr21,PlayMpr |
11:44.46 | konrads | mp3 |
11:44.51 | queuetue | I'm looking at the reports that amportal provides and the source on both numbers seems identical. |
11:45.08 | queuetue | Does that not matter? |
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11:47.03 | konrads | queuetue: when broadvoice gives you a call, it gives you a SIP INVITE in form extenssion@yourhost.whatever.net |
11:47.14 | konrads | s/ss/s/ |
11:47.40 | konrads | i would exptect, that your two numbers would have different extensions |
11:47.41 | queuetue | And how do I "see" that invite? sip debug? |
11:47.51 | konrads | queuetue: asterisk console should show it |
11:47.58 | konrads | but sip debug will definetl;ey so |
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11:50.09 | queuetue | Both show SIP/Phone1-random phone2 does not show. |
11:50.48 | konrads | can you paste those two strings? |
11:50.55 | konrads | You should look for a To: string iirc |
11:53.22 | queuetue | FROM_DID ? |
11:53.51 | konrads | look for INVITE envelope |
11:53.57 | konrads | with sip debug turned on |
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12:01.35 | queuetue | On with BV support... |
12:01.49 | visik7_ | wasim is it configurable only in 1.2.x branch or olso in 1.0.x |
12:01.52 | visik7_ | ? |
12:02.05 | queuetue | does anyone know an incoming-only company that I could deal with instead of BV? |
12:02.25 | queuetue | I'm paying for outgoing BV service s I don't need.. |
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12:02.45 | *** join/#asterisk podzap (n=podzap@roswell.pp.saunalahti.fi) |
12:02.49 | podzap | hi |
12:03.10 | podzap | i am having a really hard time getting * setup. i need some help. |
12:03.36 | podzap | i tried asteriskathome and it seems like an overcomplicated, buggy mess. |
12:04.04 | podzap | i just need to get a bare-bones * setup, please, who can help by sharing some config files, etc? |
12:05.48 | queuetue | podzap: A@H is remarkably simple, and probably th very best way to get a bare bones setup... What specifically is causing a problem ( A@H or otherwise?) |
12:06.53 | podzap | queuetue: A@H , I can't get it to connect to FWD, although I followed the example in the A@H handbook verbatim. |
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12:07.32 | podzap | queuetue: i have a sipura-2002. i configured the other line with a second fwd account, and could connect directly, no problem. but not with A@H. |
12:08.05 | queuetue | podzap: that seems pretty strange - did debugging shed any light as to why? |
12:08.09 | podzap | queuetue: i looked at the config files, A@H leaves include lines commented out, but those commented out files are the ones which are edited via the GUI |
12:09.16 | podzap | queuetue: it would immediately drop the IAX2 connection after establishing it |
12:09.50 | queuetue | Did it actually establish and drp, or did it not successfully establish? |
12:09.54 | podzap | there are something like 50 configuration files |
12:10.05 | podzap | it established and dropped |
12:10.51 | podzap | my brain can not wrap itself around 50 configuration files. i want to get an extremely simple 3-4 file setup working, then expand from there. |
12:11.01 | queuetue | Yes, you are not supposed to be digging in A@H' config files. If you have t hand-tweak then install asterisk and do it yourself... A@H is not meant to be tweaked - definitely not by a newbie... |
12:11.14 | podzap | well, it does not work |
12:11.15 | drray | or by anyone |
12:11.44 | podzap | not according to the examples in the handbook |
12:12.22 | queuetue | Idf you're doing config changes by hand, then i'm not surprised it it not working. I guess install asterisk , and then try one of the many explanations of how to get pulver up and going on asterisk. Come on back with specific problems... |
12:12.44 | podzap | i didn't try to change anything until after it did not work. |
12:13.06 | queuetue | Ok. My advice doesn't change. :) |
12:13.31 | podzap | i can put the original config files back, and show you, it still does not work. |
12:15.07 | queuetue | Give em the url for the FWD setup on A@H. |
12:15.35 | podzap | just a second |
12:19.02 | podzap | the box is really slow right at the moment, gotta wait til it calms down |
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12:20.09 | queuetue | podzap: give me (in here or pmsg) your pulver number so I can test the setup. |
12:22.21 | podzap | pm |
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12:35.15 | *** join/#asterisk djspeck (n=djspeck@ewersbach.net) |
12:35.32 | djspeck | HELLO everybody |
12:36.17 | djspeck | I have a problem: when i run asterisk as root, everything is fine, but when i try to run as asterisk i don't hear anything... |
12:36.23 | djspeck | asterisk is in group audio |
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12:37.19 | simprix | What is a good speed box for a 25 user system with a pri |
12:37.45 | konrads | a new PC? |
12:37.59 | simprix | yea |
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12:40.44 | konrads | i meant a new PC is enough |
12:41.13 | konrads | basically, CPU power is required when you translate from codec to codec |
12:41.21 | konrads | which is sometimes the VoIP case |
12:41.45 | konrads | otherwise I/O is the limiting factor |
12:42.00 | konrads | if you have active ISDN card, a recent server class stuff should work |
12:42.12 | konrads | 1CPU P4 2.4Ghz will be more than sufficient |
12:43.04 | visik7 | someone should split essential config files from auxiliary ones |
12:43.43 | *** join/#asterisk asteriskn00b (i=user65@adsl-68-91-7-225.dsl.tulsok.swbell.net) |
12:44.10 | asteriskn00b | morning all |
12:44.41 | cochi | maybe OT. but is there some tool which emulates CAPI by accessing TAPI? got some app which just supports showing callerid using CAPI :| |
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12:49.07 | asteriskn00b | get anyones opinions on the best hosted pbx solution out there? |
12:56.17 | djspeck | hello everybody.. i have a problem with starting asterisk: |
12:56.27 | djspeck | when i start it as root everything works fine |
12:56.42 | djspeck | when i start asterisk -U asterisk everything works fine |
12:56.55 | djspeck | but when i try to start it from init.d |
12:57.10 | djspeck | i don't hear anything |
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13:06.53 | cochi | mh is there any way to dial using another channel if the first group's full? like CAPI ports full, switch to SIP for dialout? |
13:08.35 | wasim | cochi: see which priority it goes to if the CAPI ports are full, and put your SIP on that, should be +1 |
13:09.43 | cochi | well but isn't dial supposed to go to next prio if other party doesnt answer, too? |
13:10.22 | wasim | you can do a chanisavail as well |
13:10.33 | cochi | oh wasnt aware of that command. thanks :) |
13:13.45 | Ariel_ | cochi, you can use a macro to create your dial checks before you dial out. You can also count how many calls are on a channel with setgroup. Take a look on the wiki. |
13:13.50 | Ariel_ | ~docs |
13:13.52 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
13:14.09 | cochi | already doing this via macro. just had overlooked the ChanIsAvail command :) |
13:14.09 | Katty | beep. |
13:14.14 | cochi | thanks for your help :) |
13:14.28 | Ariel_ | Katty, morning... |
13:14.43 | Katty | Ariel_: (= |
13:15.04 | Ariel_ | Katty, I was under the impression you had a 3 day weekend and were going on a trip? |
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13:15.29 | Katty | Ariel_: just to mamma's. |
13:15.46 | Ariel_ | ahh |
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13:27.52 | podzap | hi, where can i find a simple howto: asterisk, FWD, with sipura-2000? |
13:32.20 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
13:34.27 | Nugget | ~docs |
13:34.28 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
13:36.38 | phoe | what does wiki mean? |
13:38.24 | Ariel_ | phoe, the wiki is like a live reference doc's that if you have access too you can edit and post your info there as well. |
13:38.38 | Nugget | http://justfuckinggoogleit.com/?q=what+is+wiki |
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13:39.40 | Nivex | Ah yes. Google is the answer! I don't care what the question is! :) |
13:39.49 | *** join/#asterisk Rav1974 (n=r@ool-18bd2d63.dyn.optonline.net) |
13:39.57 | Nugget | It's certainly the answer to that question. |
13:39.59 | Ariel_ | Nugget, would it not have been easyer to just do this: |
13:40.05 | Ariel_ | ~google what is wiki |
13:40.37 | Nugget | that perhaps would not have made the point quite as well. especially in light of your lack of success just now. :) |
13:41.02 | Ariel_ | looks like jbot is not going out to the google today |
13:41.11 | Ariel_ | ~weather KTMB |
13:41.15 | PakiPenguin | jbot's been drinking |
13:41.25 | Ariel_ | well it's kinda working |
13:41.59 | Rav1974 | google is so great, don't know what I did without it |
13:42.45 | Ariel_ | Rav1974, have you yahoo.... they post they have more but they lie |
13:43.19 | Ariel_ | before google.... I used to use altavista allot back then. |
13:43.26 | Nugget | before google I altavista'd. before altavista I yahoo'd. |
13:43.31 | gordonjcp | altavista was good |
13:43.48 | Nugget | before yahoo, everyone had a big list of cool links on their home page. :) |
13:43.51 | Rav1974 | yesterday, 2 printers started spewing out this foul smell, changed drum & toner but it was still smelling. Called brother for any known issues, they had none. Looked on google & then found the answer!!! |
13:44.22 | Ariel_ | Rav1974, what you found you needed to change the ozone filter |
13:44.25 | Rav1974 | it was the paper. It was the most aweful smell. Like vomit. The receiptionist threw up twice. |
13:44.50 | Rav1974 | Ariel_: we just used a different brand of paper and it was fine |
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13:45.22 | Rav1974 | Thank god for google |
13:45.35 | Rav1974 | if they made it a paid service, I'd subscribe |
13:45.50 | Ariel_ | don't evey think of that |
13:46.07 | Rav1974 | :) |
13:46.12 | Ariel_ | argh paid service what will we do... Oh it's already paid service it has ads |
13:46.49 | Rav1974 | true |
13:47.24 | Ariel_ | in fact on there site the first 2 or 3 are paid service links... kinda selling out in my view |
13:48.16 | Rav1974 | at least it works, searching with other companies doesn't work as well |
13:48.28 | Ariel_ | ok I have an error with a php page on Suse. Anyone here good with php and suse? |
13:48.59 | Rav1974 | all the tech dudes are sleeping |
13:49.55 | Ariel_ | well I think this is more of something missing in suse's apt installer. |
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13:51.38 | Rav1974 | phoe: those tech guys are going to be grogy so you better summon some women to wake them up |
13:52.07 | Ariel_ | Rav1974, sometimes it's not a matter of women but beer |
13:52.37 | phoe | my skill is too low to summon women :( |
13:52.59 | Ariel_ | beer then |
13:53.30 | phoe | when was the first release of Asterisk? |
13:53.40 | Rav1974 | :) |
13:53.53 | Ariel_ | last year version 1 around the first astercom |
13:54.02 | phoe | astercom? |
13:54.07 | Rav1974 | zapatatelephony.org was the starting point |
13:55.07 | Ariel_ | astricon |
13:55.09 | Ariel_ | sorry |
13:55.25 | phoe | lol, nm. I dunno anything about it ;) |
13:55.48 | mog_home | asterisk is aprox 5 years old |
13:55.48 | Ariel_ | Rav1974, yes it was I was using .5 version a little over 3 years ago. It was very different back then. No wiki yet |
13:55.53 | mog_home | in the current form 4 years old |
13:56.05 | mog_home | a non threaded version was made for lss back in the day |
13:56.19 | mog_home | mark rewrote his code for a threaded version 10 -12 months later |
13:56.25 | mog_home | and is the same base we use now |
13:56.32 | Ariel_ | mog_home, great info |
13:56.41 | *** join/#asterisk Corydon76-home (i=orange@pdpc/supporter/sustaining/Corydon76-home) |
13:56.54 | Ariel_ | perks hummmmm |
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14:12.38 | natted | Good morning everyone... I'm having some problems with NAT, for example, an X-Lite receiving calls that were dialed to an ATA-186, both behind the same address... The Asterisk Server is outside the NAT, any suggestions please? |
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14:18.29 | clyrrad | does anyone know if you can pattern match using gotoif something like exten => s,1,GotoIf($["${ARG1}" = "_XXXX"]?16:2) |
14:19.24 | Ariel_ | yes but that matches everything |
14:19.32 | Ariel_ | take the _ off it |
14:22.55 | clyrrad | exten => s,1,GotoIf($["${ARG1}" = "2XXX"]?16:2) |
14:23.01 | clyrrad | didnt seem to make much difference |
14:23.25 | clyrrad | that "should" matching all 4 digit extensions starting with 2 |
14:23.51 | Ariel_ | wait |
14:24.45 | PakiPenguin | anyone here uses linksys pap? |
14:24.49 | *** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl) |
14:25.41 | Zaw | PakiPenguin: i have one, but it sucks and goes down all the time. i'm guessing it's just my particular one going bad, but i dunno. |
14:27.42 | PakiPenguin | Zaw, mine is acting strange , sometimes it calls , sometimes it just gives me a fast busy |
14:28.27 | Zaw | yep |
14:28.33 | Zaw | sometimes it doesn't have a dial tone |
14:28.38 | Zaw | that's what mine does too |
14:29.50 | Ariel_ | clyrrad, exten => s,1,GotoIf($[${ARG1} = "2XXX"]?16:2) |
14:30.32 | Ariel_ | PakiPenguin, is it behind a nat? |
14:31.28 | clyrrad | Ariel_ that yields the same results, * skips right over it... It does not seem to evaluate the pattern expression... |
14:31.52 | Ariel_ | your getting this from another macro |
14:32.07 | clyrrad | ? |
14:32.25 | clyrrad | I am in a macro where I run that line you just pasted... if thats your question ;) |
14:32.48 | Ariel_ | what is your dial string that goes to this macro |
14:33.28 | clyrrad | the only thing that goes to this macro is an extension number |
14:33.42 | clyrrad | a 4 digit extension thats held in ARG1 |
14:34.26 | PakiPenguin | Ariel_: yeah |
14:34.28 | PakiPenguin | it is |
14:34.37 | PakiPenguin | and i cant seem to find a place to add a stun server |
14:35.30 | Ariel_ | PakiPenguin, your firewall/nat is not keeping the port open need to set it to register less then every minute. This might be the issue |
14:35.42 | Ariel_ | qaulify=yes might help with this |
14:36.12 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
14:36.18 | clyrrad | Ariel_ is it because I am in a macro that gets it all screwed up? |
14:36.42 | PakiPenguin | hmmjs testing |
14:36.43 | Ariel_ | clyrrad, it's suppose to work in a macro |
14:36.59 | clyrrad | hrm.... where are we going wrong then? lol |
14:38.09 | PakiPenguin | i think thats the issue! |
14:38.14 | PakiPenguin | thanks Ariel_ |
14:38.22 | Ariel_ | PakiPenguin, np |
14:39.09 | *** join/#asterisk |cleric| (n=dacleric@p5482A5D6.dip0.t-ipconnect.de) |
14:41.26 | Ariel_ | clyrrad, put a exten => s,1,NoOp(${ARG1} exten => s,2,GotoIf($["${ARG1}" = "2XXX"]?16:3) |
14:41.38 | Ariel_ | just to see what you have for ARG1 |
14:41.51 | Ariel_ | clyrrad, put a exten => s,1,NoOp(${ARG1}) exten => s,2,GotoIf($["${ARG1}" = "2XXX"]?16:3) |
14:41.59 | Ariel_ | sorry forgot the ) |
14:42.42 | clyrrad | lol thats ok i cought that :) |
14:43.24 | clyrrad | arg1 = 2000 |
14:44.15 | konrads | i thikn that asterisk extensions are the most scary part |
14:44.21 | konrads | the rest either works or not |
14:44.27 | konrads | like my capi stuff :( |
14:44.38 | konrads | or h323->sip bridging |
14:47.20 | clyrrad | Ariel_ it works like this exten => s,2,GotoIf($["${ARG1}" = "2000"]?16:3) |
14:47.38 | clyrrad | only problem is I would need to do that for EACH extension.... that would be a real pain |
14:50.18 | PakiPenguin | konrads: capi always works for me |
14:53.51 | *** join/#asterisk fataia`falamu (i=cLeRiSY@CPE-61-9-197-32.nsw.bigpond.net.au) |
14:54.37 | *** join/#asterisk kwoot (n=kvirc@i2rs-son.xs4all.nl) |
14:54.55 | konrads | PakiPenguin: did you use misdn with capi? |
14:55.53 | kwoot | hello. Is there anybody willing to help me out configuring zaphfc to an BRI? I am really stuck now. |
14:56.11 | *** join/#asterisk juice (n=juice@mo-69-69-116-98.sta.sprint-hsd.net) |
14:57.32 | *** join/#asterisk patrick_ (n=patrick@pc-205-21.scpe.quickclic.net) |
14:58.54 | clyrrad | Ariel GotoIf(Condition?label1:label2) is from the manual, perhaps we cant do pater matching, and can only use a LABEL or Static value? |
14:59.18 | kwoot | when starting * it does say it sees 1 bri interface with 2 channels but when I want to call it says no channel avail. Why? |
15:02.28 | *** join/#asterisk rbd (n=robbyd@24-181-194-064.dhcp.hckr.nc.charter.com) |
15:07.30 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
15:07.40 | kwoot | hello? anybody? |
15:07.52 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
15:08.20 | Wonka | http://data.4chan.org/b/src/1125712295576.jpg |
15:09.03 | hugo-v6 | Wonka: lol |
15:09.14 | phoe | hi kwoot |
15:09.17 | Wonka | not nearly pc, but funny :) |
15:09.37 | *** join/#asterisk Byte (i=byte@2001:4bd0:1000:0:202:44ff:fe47:d3ee) |
15:11.29 | kwoot | Hy phoe: okay. I see somebody is awake :-) very good. Can you help me with the question I stated earlier? |
15:12.11 | phoe | no sorry :), I don't know anything about that stuff |
15:12.31 | kwoot | thats too bad. thanks at least for reading. |
15:13.05 | *** join/#asterisk Byte (i=byte@proxima.arlott.org.uk) |
15:13.06 | *** part/#asterisk ChrisN (i=[v838MAk@166.84.1.5) |
15:17.56 | *** join/#asterisk file[laptop] (n=jcolp@156.34.35.106) |
15:20.19 | [Airwolf] | I'm wondering. I'm using macro's in my dailplan. But is it possible to use a macro in de sip configuration. Because administration of sip.conf will be more difficult when I have more phones. |
15:20.27 | [Airwolf] | Now I have something like this: |
15:20.28 | [Airwolf] | restrictcid |
15:20.31 | [Airwolf] | aarg |
15:20.34 | [Airwolf] | this: |
15:20.37 | [Airwolf] | [498] |
15:20.37 | [Airwolf] | ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)! |
15:20.37 | [Airwolf] | ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed |
15:20.37 | [Airwolf] | type=friend |
15:20.37 | [Airwolf] | regexten=498 ; When they register, create extension 1234 |
15:20.37 | [Airwolf] | username=joost |
15:20.39 | [Airwolf] | callerid="Joost" <498> |
15:20.42 | [Airwolf] | host=dynamic |
15:20.43 | [Airwolf] | ;nat=yes ; X-Lite is behind a NAT router |
15:20.45 | [Airwolf] | ;canreinvite=no ; Typically set to NO if behind NAT |
15:20.49 | [Airwolf] | disallow=all |
15:20.51 | [Airwolf] | allow=gsm ; GSM consumes far less bandwidth than ulaw |
15:20.53 | [Airwolf] | allow=ulaw |
15:20.55 | [Airwolf] | allow=alaw |
15:20.57 | [Airwolf] | mailbox=498@default |
15:20.59 | [Airwolf] | context=putgraaf |
15:21.01 | [Airwolf] | And I would like to have just one configuration and then just one line for every user to configure it |
15:21.06 | [Airwolf] | Like I have in my dial plan |
15:21.53 | *** part/#asterisk greekman (n=alex@host254.209.113.199.conversent.net) |
15:22.16 | newl | afaik, that isn't possible. |
15:22.54 | drumkilla | actually, yes you can |
15:23.07 | drumkilla | in cvs head, there is a way to do inheritance in configuration ... |
15:23.36 | newl | from AEL to ASL? lol |
15:23.52 | drumkilla | If you read doc/README.configuration in the source directory, it will tell you about how it works |
15:24.48 | drumkilla | it's not the same as a macro, but ... it cuts down on repeated config |
15:25.10 | drumkilla | ~thwack file[laptop] |
15:25.13 | jbot | ACTION smacks file[laptop] on the forehead with a toaster |
15:25.17 | drumkilla | ! |
15:25.25 | newl | flee! |
15:25.57 | file[laptop] | drumkilla: what'cha doing? |
15:26.22 | *** join/#asterisk hat (n=hat@cm157.epsilon173.maxonline.com.sg) |
15:26.54 | clyrrad | does anyone know how to pattern match with GoToIF? Seems * does not.... exten => s,2,GotoIf($["{ARG1}"= "2XXX"]?16:3) |
15:27.20 | [Airwolf] | drumkilla, thanks |
15:27.36 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
15:27.48 | [Airwolf] | But I'm not using the cvs-head |
15:28.24 | [Airwolf] | But isn't it possible to use variables in de sip.conf. That would atleast make it a littlebit easier. |
15:28.39 | *** join/#asterisk Samoied (n=Samoied@201.3.210.150) |
15:28.49 | *** join/#asterisk chet (n=chet@cpe-065-190-056-004.triad.res.rr.com) |
15:29.01 | drumkilla | no, it's not. |
15:29.05 | hat | hi, we need to setup a asterisk web callback system. The buget is about USD$6,000(including all hardware). What configurations of these hardware ? |
15:29.06 | clyrrad | Airwolf.... I was not able to get variables to work in SIP.conf or IAX.conf, only in EXTENSIONS.conf |
15:29.10 | drumkilla | file[laptop]: about to get my party on :D |
15:29.20 | *** join/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
15:29.35 | [Airwolf] | clyrrad, ok then I just have to copy & paste |
15:29.35 | [Airwolf] | :P |
15:29.51 | clyrrad | yup, thats what I had to do.... kinda sucks but at least it works |
15:29.55 | file[laptop] | drumkilla: excellent |
15:30.23 | drumkilla | file[laptop]: first football game is today, so this town is going crazy |
15:30.39 | file[laptop] | not as crazy as here, the rolling stones play tonight |
15:30.45 | file[laptop] | so it's a wee bit insane |
15:31.36 | kwoot | can someone tell me how to debug the dial application to find out why * doesn´t use a channel? |
15:32.10 | kwoot | o, sorry. Addon: please. :-) |
15:32.27 | file[laptop] | kwoot: that made no sense to me |
15:32.34 | *** join/#asterisk Laureano (n=tomas@OL155-33.fibertel.com.ar) |
15:33.05 | kwoot | well, i seem to have a working zaphfc, but I still can not dial out. Dunno why. |
15:35.14 | hat | which digium card i should get to support maximum numbers of concurrent calls and less CPU usage? |
15:35.20 | kwoot | eh, hello? |
15:35.37 | kwoot | <hat> an expensive one? :-) |
15:35.40 | konrads | kwoot: error messages? |
15:35.44 | file[laptop] | my world doesn't revolve around IRC you know |
15:36.03 | [Airwolf] | file[laptop], is there any other world then ? |
15:36.10 | hat | my budget is about 6000 USD$. Of course, it must be stable |
15:36.19 | file[laptop] | [Airwolf]: indeed, the real world |
15:36.26 | kwoot | konrads: unable to create channel of type zap |
15:36.32 | chet | hat, here are some products- http://www.digium.com/index.php?menu=product_category&category=hardware |
15:36.33 | hat | kwoot, including a server |
15:36.35 | [Airwolf] | file[laptop], aahhh, I have to check it out one time ;) |
15:36.45 | hat | chet, thanks & let me see |
15:38.18 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
15:38.29 | file[laptop] | hahahahahahaha |
15:38.29 | *** part/#asterisk Laureano (n=tomas@OL155-33.fibertel.com.ar) |
15:38.35 | file[laptop] | my parents went to PEI |
15:38.51 | chet | which linux distro do you guys recommend for *? cent, debian, etc.. |
15:39.02 | chet | or freebsd? |
15:39.18 | [Airwolf] | My personal favorites are debian and gentoo |
15:39.30 | [Airwolf] | But that's different for everybody |
15:39.33 | chet | yeah |
15:39.39 | Nugget | asterisk is the same no matter what unix you use. |
15:39.44 | chet | just wondering what some people have had success with |
15:39.51 | Nugget | zaptel is a total pain in the ass unless you use linux. |
15:39.51 | file[laptop] | PARTY! |
15:39.57 | chet | well, hasnt there been some issues with freebsd? |
15:40.04 | Nugget | only with zaptel |
15:40.07 | chet | ok |
15:40.27 | hat | chet, what is 3.3 volt PCI and 5.0 volt PCI ? DOes it mean i need to choose my server first? and what is the difference between echo cancellation card and non echo-cancellation-card? |
15:40.29 | Nugget | I run three asterisk servers. One in Linux, one in FreeBSD, and one in OSX. |
15:40.42 | chet | yes hat, based on motherboard specifications |
15:41.03 | chet | hat- you may want to find asterisk consultants on voip-info.org and ask for recommendations |
15:41.24 | chet | Nugget, for recreation or production? |
15:41.44 | Nugget | both |
15:42.35 | hat | chet, i hope i can get help from irc :) |
15:42.43 | Nugget | My home machine is linux because I wanted a zaptel card. My "production" server is freebsd because I didn't need zaptel. My powerbook is osx because osx rocks. |
15:42.53 | file[laptop] | Nugget: yes, yes it does |
15:43.20 | chet | very nice |
15:43.22 | Nugget | I run asterisk on the powerbook because I've had much better luck running x-lite connecting to a local asterisk install and then talking iax to my "real" server. |
15:43.37 | Nugget | sip over hotel ethernet with its nat hell is often impossible |
15:44.23 | chet | yeah, i read they were looking at making or changing sip for that reason? and other business reasons |
15:44.25 | *** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl) |
15:44.33 | Nugget | "they"? |
15:44.47 | chet | well, some people were proposing rfc additions |
15:44.57 | chet | i think i read it in networkworld |
15:45.00 | Netgeeks | hat: just check the manual on the motherboard your computer will use, it will tell you whether or not your PCI slots are 3.3 or 5.0 volts |
15:45.26 | file[laptop] | SIP has changed so much, it's not funny |
15:45.46 | Netgeeks | hat: as for echo cancelling versus lack of echo cancelling in hardware. Once the hardware and drivers are stable, the chard with echo-cancelling will perform better both in CPU utilization as well as in echo cancelling quality |
15:45.49 | *** join/#asterisk DrJolo (n=chatzill@host-ip58-113.crowley.pl) |
15:45.54 | file[laptop] | "Let's add this, let's change this - oh one method of putting a call on hold? nooo - let's add another!!!" |
15:46.08 | chet | file[laptop]- has that changed interoperability? or has everyone pretty much kept up? |
15:46.16 | Netgeeks | hat: right now, if I understand it correctly people are having stability issues with the cards that have the on-board canceller |
15:46.23 | file[laptop] | chet: for the most part stuff works, as implementation of basic things are fine |
15:46.26 | hat | thanks Netgeeks |
15:46.28 | file[laptop] | and most stuff retains backwards compatibility |
15:46.32 | chet | cool |
15:46.35 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
15:46.52 | file[laptop] | there's not many ways you can screw up a basic SIP call |
15:46.52 | Netgeeks | I'm writing an extension to the rfc to allow SIP to flush my toilet |
15:47.08 | file[laptop] | Netgeeks: INVITE to the toilet, then send BYE :P |
15:47.13 | chet | http://www.networkworld.com/techinsider/2005/061305ti-sip-b.html |
15:47.19 | drumkilla | what does the SDP look like? |
15:47.19 | chet | article on sip-b |
15:47.41 | Netgeeks | file: not so simple, due to the whole water conservation thing, we have to check the status before the flush |
15:47.42 | hat | Netgeeks, do you mean current echo-cancellation-card is not stable? |
15:47.58 | Netgeeks | if the bowl is fresh, no need to flush... |
15:48.02 | file[laptop] | Netgeeks: ah so you get a status code back from the toilet |
15:48.18 | chet | if=yellow then=mellow |
15:48.21 | *** join/#asterisk bmg505 (n=leon@rndf-146-11-247.telkomadsl.co.za) |
15:49.19 | Netgeeks | But the toilet hardware is incapable of determining bowl status (it only knows full of water or empty) so I was thinking about attaching a web cam and using some code to determine cleanliness via the some visual recognition system.... ;) |
15:49.31 | chet | nice |
15:49.50 | Netgeeks | hat: the mix of hardware/drivers... I'm not sure, I haven't played with the echo can cards yet |
15:50.27 | Netgeeks | As for the current software echo-can in asterisk, it doesn't crash asterisk, so you could call it stable. |
15:50.48 | hat | thanks. Netgeeks. |
15:50.52 | Netgeeks | Heya drumkilla: did you by chance get my email with updates on some work I'm looking to get done? |
15:51.17 | file[laptop] | Netgeeks: drumkilla is very very silly |
15:51.53 | Netgeeks | and that makes him different from the rest of us, how? ;) |
15:52.05 | Netgeeks | besides timecop of course, he's just mean |
15:52.06 | file[laptop] | we're only very silly |
15:53.26 | Netgeeks | well, for any asterisk devs looking for a little income on the side, I've got some small patches I need done in the next 10 days.... |
15:53.44 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
15:53.44 | *** mode/#asterisk [+o bkw__] by ChanServ |
15:53.48 | file[laptop] | Netgeeks: what is it? |
15:53.59 | file[laptop] | which reminds me, I need to hunt down Leif and submit this to the bug tracker |
15:54.02 | Netgeeks | hrm, bunch of small stuff.... let see |
15:54.19 | Netgeeks | add setvar= to zaptel, function just like sip and iax setvar |
15:54.30 | *** join/#asterisk huslage (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
15:54.31 | *** join/#asterisk nain (n=nain@137.101.144.131) |
15:54.40 | nain | HI |
15:54.43 | file[laptop] | a likely story. |
15:55.24 | file[laptop] | hey Mr. DJ, in case you forgot |
15:55.36 | file[laptop] | I came to get down so you better make it hot, cause I can't jump around when I hear... |
15:55.37 | Netgeeks | add a global system_id, which is a field in all realtime db's, so that when a system loads config info from realtime you can have one database for numerous systems and identify the entries either as general (for all) or for a specific system_id |
15:57.07 | Netgeeks | and a few more little patches that I can provide in email if you are interested. However I'm on a deadline... need them by 9/12 for testing, and if issues are found, fixed by 9/19 |
15:57.10 | nain | Some one would like to tell me that how to apply patch rtcp_patch_20050817B.patch.txt to asterisk avaialble at http://bugs.digium.com/view.php?id=2863 ? |
15:58.40 | Netgeeks | well, I'm going back to sleep.... |
15:59.20 | file[laptop] | sleep is overrated |
15:59.32 | Netgeeks | as is growing old! |
15:59.33 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
15:59.56 | Netgeeks | unfortunately they are directly related, older = need more sleep |
16:00.16 | file[laptop] | not... OLD! |
16:00.49 | Netgeeks | was up til 3 am beating my head against some html/python/sql stuff and had to get up 7 am to help the wife set up her farmer's market stand |
16:00.54 | Netgeeks | it's nap time! |
16:01.13 | Netgeeks | but only after watching yesterday's Battlestar Galactica episode |
16:02.05 | Netgeeks | if you are interested in doing any of those patches, or want info on the others, let me know file... compensation is included.... chris at netgeeks dot net |
16:03.49 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
16:04.44 | *** join/#asterisk YoYo (i=YoYo@pool-151-199-13-222.roa.east.verizon.net) |
16:05.40 | nain | file[laptop]: Hi |
16:06.16 | file[laptop] | ...hello |
16:07.05 | nain | i would like to patch asterisk 1.0.9 against this patch rtcp_patch_20050817B.patch.txt, at http://bugs.digium.com/view.php?id=2863. But i don't know how to apply patch to asterisk ? can u help plz ..... |
16:07.29 | file[laptop] | download the patch to your asterisk directory |
16:07.36 | file[laptop] | and type patch -p0 < filename |
16:08.02 | nain | OK, Only needed to patch or i have to recompile asterisk again after patching ? |
16:08.15 | file[laptop] | you have to recompile asterisk |
16:08.24 | file[laptop] | or else your asterisk won't be patched |
16:09.18 | nain | ok, it should be like this? patch -p0 < /path-to-path/patch rtcp_patch_20050817B.patch.txt or i have to define the file name which is going to be patch ? |
16:09.29 | file[laptop] | nope. |
16:09.34 | file[laptop] | just do like I said. |
16:09.41 | nain | Ok |
16:09.58 | file[laptop] | probably won't apply totally, so have fun! |
16:10.01 | DannyF | morning folks |
16:10.04 | DannyF | lo file |
16:10.08 | file[laptop] | hi hi |
16:10.42 | Ariel_ | nain, that patch is for CVS Head you said you have stable 1.0.9 |
16:11.03 | nain | yes i have 1.0.9 |
16:11.05 | file[laptop] | ah yes, I didn't even see that - yeah you need CVS head... |
16:11.08 | nain | so i can't apply it ? |
16:11.20 | file[laptop] | and that patch is for August 17th, SOOO it may not apply cleanly to today's CVS |
16:11.26 | file[laptop] | nain: correct, it won't work on 1.0.9 |
16:11.36 | nain | hmmmmm, |
16:11.57 | nain | Actually i am getting this error and calls going to drop. what does this error means "Unable to write fd 60 (32, Broken Pipe)" this cause to hangup calls |
16:12.04 | Ariel_ | there have been lots of changes to the chan_sip |
16:12.30 | *** join/#asterisk darkskiez (n=darkskie@host86-132-169-66.range86-132.btcentralplus.com) |
16:12.51 | file[laptop] | nain: it means something very wrong happened |
16:13.12 | file[laptop] | because I know what that error means, but I've never seen anybody have it happen, 'nor do I know what would cause it |
16:13.25 | nain | file[laptop]: yes this is very serious error and i m in very trouble i didn't find any help regarding this error. |
16:13.41 | nain | I don't konw what's wrong with my setup |
16:14.31 | nain | file[laptop]: would you explain error plz ? what does it mean... |
16:14.49 | file[laptop] | it means the file descriptor where it was trying to write to returned an error |
16:15.29 | file[laptop] | what are you using for a distro? or dare I ask... OS |
16:15.43 | nain | so it is the operating system error causing dropping of calls ? |
16:15.45 | nain | sure |
16:15.52 | nain | I am using Fedora core 3 |
16:15.58 | *** join/#asterisk natted (n=maxgluck@200.109.166.83) |
16:16.00 | file[laptop] | that should work fine |
16:16.17 | file[laptop] | I think my devel box is FC3, I forget... |
16:16.19 | nain | I have asterisk 1.0.9 with asterisk-oh323.0.6.6 running |
16:16.46 | file[laptop] | try 1.2.0 beta1 |
16:16.53 | natted | hi, does insecure = very really exist? because I can only find explanations for invite and port on the wiki |
16:16.53 | DannyF | &%¤%# |
16:17.06 | file[laptop] | natted: yes, it's both invite and port |
16:17.10 | DannyF | is everyone on sixtel on vacation? |
16:17.17 | *** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
16:17.19 | file[laptop] | DannyF: always. |
16:17.21 | nain | i tried it but failed to compile so i continue to asterisk 1.0.9 and waiting for stable release of 1.0.2 |
16:17.29 | natted | hmmm, ok. |
16:17.47 | DannyF | heh tried one of their DID's and it permanently busy... aint even reaching me.... |
16:18.02 | DannyF | ah well, guess i'll us someone else |
16:18.03 | nain | sorry asterisk 1.2 not 1.0.2 |
16:18.05 | DannyF | use* |
16:18.36 | file[laptop] | nain: then the logical course is to figure out why 1.2.0 didn't compile |
16:18.50 | *** join/#asterisk rob314 (n=rob314@cpe-65-185-169-238.neo.res.rr.com) |
16:19.18 | *** join/#asterisk jpayne (n=jpayne@baconhouse.sackheads.org) |
16:19.24 | file[laptop] | I wonder when my food will be here |
16:20.19 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
16:21.15 | nain | file[laptop]: i didn't try hard to compile 1.2.0. Actually asterisk 1.0.9 and previous stable release also tested but all have same error with my setup. |
16:21.50 | file[laptop] | oh food is here |
16:22.04 | Ariel_ | nain, what is the error your getting and what is your system? |
16:22.18 | file[laptop] | Ariel_: file descriptor go kaboom |
16:22.35 | *** join/#asterisk RoyK (n=roy@100.80-203-27.nextgentel.com) |
16:22.36 | Ariel_ | nain, I see it's fedora Core 3 |
16:22.49 | nain | Ariel_: OS: FC3, asterisk1.0.9, asterisk -oh323-0.6.6 and getting "Unable to write fd 60 (32, Broken Pipe error)" |
16:23.14 | *** join/#asterisk werkenbass (i=werkenba@pc-14-79-86-200.cm.vtr.net) |
16:23.15 | RoyK | ~seen kpfleming |
16:23.17 | jbot | kpfleming <~kpfleming@207.111.174.1> was last seen on IRC in channel #asterisk, 156d 10h 27m 4s ago, saying: 'no, there is no specific plan at this time'. |
16:23.21 | nain | the value of fd xx is random some time it is 43, 60, 5x etc |
16:23.26 | werkenbass | hey motherfuckers :D |
16:24.33 | werkenbass | :( |
16:24.40 | *** part/#asterisk werkenbass (i=werkenba@pc-14-79-86-200.cm.vtr.net) |
16:25.06 | nain | Ariel_: Any idea on error ? |
16:25.32 | Ariel_ | nain, what hard ware do you have in the system? |
16:27.43 | nain | Ariel_: I am not using any digium or other card, SYstem works as h323 to SIP GW, I took call in h323 and forward to sip provider |
16:27.55 | *** part/#asterisk rob314[laptop] (n=rob314@cpe-65-185-169-238.neo.res.rr.com) |
16:28.29 | nain | Ariel_: and regarding system specification, it's intel Piv machine |
16:30.54 | *** join/#asterisk BurnedOutGeek (n=mark@66-191-172-049.dhcp.gnvl.sc.charter.com) |
16:31.45 | Ariel_ | nain, well that should work fine. P4 turn off Hyper threading and what MOH program are you using? mpg123.59r |
16:32.05 | *** join/#asterisk kamathln (i=foobar@220.226.14.205) |
16:32.41 | nain | Ariel_: MOH program? well i m not using voicemail feature so no mpg123xxxx usage |
16:33.12 | Ariel_ | nain, is it installed? |
16:33.40 | nain | Ariel_: How can i check i did default compilation of asterisk ? |
16:33.53 | *** join/#asterisk speck (n=root@ewersbach.net) |
16:34.46 | Ariel_ | at the prompt #mpg123 will give you info if there is one installed FC3 comes with there own mpg321 which is not compatible with asterisk |
16:35.18 | speck | hi everybody i have a problem with dialing in asterisk, when i place a call to sip, it alerts, but only 20 seconds, then there is a message: I IND: TIMEOUT and it hangs up. CAN SO HELP ME?? |
16:35.56 | speck | I didn't define a timeout in DIAL command |
16:35.59 | Ariel_ | speck, it alerts like rings the device |
16:36.07 | speck | jes.. |
16:36.07 | nain | Ariel_: server*CLI> mpg123 |
16:36.08 | nain | No such command 'mpg123' (type 'help' for help) |
16:36.18 | Ariel_ | ok |
16:36.36 | speck | nain: try to install mpg123 but not this from debian package |
16:37.01 | clyrrad | Ariel_ did you have any other thought on the pattern matching? |
16:37.09 | Ariel_ | speck, what are your dial string |
16:37.13 | speck | mom |
16:37.38 | nain | Ariel_: how does mpg123 relates to File descriptor error ? and should i install it and from where if not from debian ? |
16:37.40 | Ariel_ | clyrrad, not really. |
16:37.58 | clyrrad | Ariel_ maybe it cant be done? Have you ever done it? |
16:38.02 | Ariel_ | nain it relates allot to broken pipe |
16:38.12 | speck | Ariel_: Dial(SIP/${EXTEN}@sip.1und1.de,,m) |
16:38.32 | Ariel_ | clyrrad, I was looking for a mcro I had with it setup like that. But could not find it. |
16:38.43 | speck | when i try ...de,1000,m) its the same problem |
16:38.47 | nain | Ariel_: strange, so from where should i installed it and which one is compatible with asterisk ? |
16:38.52 | Ariel_ | speck why m |
16:38.54 | clyrrad | Ariel_ so then you had it working? Means there is a way to do it..... |
16:39.04 | speck | want to have music while dialing |
16:39.39 | Ariel_ | speck, do you have canreinvite=no in the sip.conf |
16:39.47 | speck | yes |
16:40.52 | speck | Ariel_: it's not only with SIP, also with mISDN |
16:41.07 | SkramX | hello all. |
16:42.22 | speck | I wonder why it is I IND: TIMEOUT, that normaly means it receives timeout from ISDN Bus |
16:44.42 | speck | And i have the error message all the time: dev_del_timer: timer(8153f14) not running |
16:45.29 | Ariel_ | speck, don't know the mISDN device you setup but I know most systems do need a timing device |
16:45.45 | Ariel_ | do you have the timing set correctly for the device? |
16:45.46 | *** join/#asterisk azrishahril (n=azrishah@60.50.193.179) |
16:46.02 | speck | Ariel_ where can i set the timing?? |
16:46.18 | speck | and what is dev_del_timer?? |
16:47.12 | Damin | Woot.. |
16:48.28 | file[laptop] | Damin: YOU! |
16:49.59 | JDLSpeedy | anyone use teliax |
16:50.00 | JDLSpeedy | ? |
16:50.45 | RoyK | teliax?? |
16:50.56 | JDLSpeedy | voip service |
16:51.26 | RoyK | ah |
16:51.30 | RoyK | ulaw voip |
16:51.37 | JDLSpeedy | allows u to make phone calls |
16:51.39 | JDLSpeedy | ya |
16:51.44 | RoyK | OH |
16:51.50 | RoyK | YOU MUST BE KIDDING?????? |
16:51.59 | RoyK | a voip service allowing THAT????? |
16:53.16 | bkw__ | haha |
16:53.19 | bkw__ | smartass |
16:53.31 | JDLSpeedy | Im just trying to figure out if they send u mail to ur home address |
16:53.52 | speck | mISDN: I IND :TIMEOUT when dialing afer 10 secs of no answer. |
16:53.57 | speck | HELP! |
16:54.46 | RoyK | we should have separate channels for mISDN and h.323 |
16:56.19 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
16:57.40 | *** join/#asterisk Dogzz (n=bob@66.148.168.234.nw.nuvox.net) |
16:57.49 | *** join/#asterisk doolph (i=doolph@201.226.146.178) |
17:00.00 | *** join/#asterisk knoppix (n=knoppix@cm52.theta17.maxonline.com.sg) |
17:03.18 | *** join/#asterisk bsd3 (n=bsd@203.134.192.202) |
17:04.02 | *** join/#asterisk pauldy (n=pauldy@c-67-162-255-229.hsd1.tx.comcast.net) |
17:07.56 | *** join/#asterisk jake1932 (n=jake1932@pool-70-16-130-29.phil.east.verizon.net) |
17:08.03 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
17:08.06 | netsurfer | hi ppl |
17:08.34 | netsurfer | has anyone here got any docs on app_groupcount ? it's missing on the wiki |
17:08.50 | netsurfer | ~docs |
17:08.51 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
17:09.06 | jake1932 | is there a way to ge the "user" variable from sip.conf inside the dialplan? |
17:10.03 | Ariel_ | Netgeeks, http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SetGroup |
17:10.10 | Ariel_ | it's part of setgroup |
17:11.03 | jake1932 | i'm trying to get the username automatically for voicemail - but the callerid i want to use for the pstn |
17:11.07 | *** join/#asterisk f_meehan (n=fmeehan@whoami8.cedval.org) |
17:11.16 | netsurfer | Ariel_ - I guess u meant netsurfer ;) |
17:11.26 | netsurfer | thx.. i'd seen that page.. but wanted more info |
17:11.28 | Ariel_ | netsurfer, yes |
17:11.37 | Ariel_ | netsurfer, like? |
17:12.07 | Ariel_ | jake1932, username automaticly to voicemail? |
17:12.42 | jake1932 | right - i.e. i press the voicemail key and do a Voicemail Main with the username |
17:12.47 | netsurfer | Ariel_ - i've overlooked the main section of that page heh.. think it covers my questions.. i'll try a reconfig now and see how it goes |
17:12.57 | jake1932 | but only value i could find was the callerid |
17:13.29 | jake1932 | callerid != username |
17:13.55 | Ariel_ | callerID="Name One"<200> |
17:14.24 | jake1932 | my callerid is 215XXXXXXX |
17:14.32 | Ariel_ | then you have ${CALLERIDNUM} and ${CALLERIDNAME} |
17:14.49 | jake1932 | right - but I'm looking for the username (not callerid) |
17:15.01 | jake1932 | userid is 2000 |
17:16.32 | jake1932 | looks like hack to do it: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+channels - but wanted to know if there's any variable |
17:18.21 | Qwell | jake1932: sip? |
17:18.26 | jake1932 | yep |
17:18.26 | Qwell | mailbox= |
17:18.56 | jake1932 | i have that - but that seems to be only for MWI |
17:19.55 | Qwell | dunno, I hit my voicemail button, and it takes me to my voicemail box |
17:19.58 | Ariel_ | well looks like your going to have to use the cut unless your username is the same as the exten number. |
17:20.33 | jake1932 | Ariel_: yep - just wanted to make sure |
17:20.36 | Qwell | nowhere in my cisco config does it have my voicemail box, and in extensions.conf, all I have is Voicemailmain |
17:21.00 | jake1932 | Qwell: but, your callerid is probably your userid? |
17:21.05 | Qwell | I think, anyhow |
17:21.10 | Qwell | no |
17:21.22 | Qwell | hmm |
17:21.44 | Qwell | there should be a var |
17:22.22 | jake1932 | i call it - VoicemailMain(s${CALLERIDNUM}) |
17:22.27 | *** part/#asterisk bsd3 (n=bsd@203.134.192.202) |
17:23.15 | Ariel_ | jake1932, that goes in using your callerIDnumber and by passes the password. |
17:23.50 | jake1932 | Ariel_: right - which would work if my callerid was a valid mailbox |
17:24.19 | podzap | hi, is anybody here running * behind NAT, connected to FWD? |
17:24.51 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
17:24.53 | Ariel_ | podzap, yes |
17:25.11 | Qwell | Why isn't the value of mailbox= in a variable? |
17:25.18 | Qwell | seems like that would be incredibly useful |
17:25.22 | jake1932 | hehe |
17:26.25 | jake1932 | Qwell: i wish I knew C better - I'd do it |
17:26.59 | jake1932 | proabaly not that difficult |
17:27.04 | podzap | Ariel_: do you use a sipura? |
17:27.16 | Ariel_ | I have do. |
17:27.38 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
17:27.49 | podzap | Ariel_: i am having a hard time getting connected with * to fwd, behind nat, using iax. it works from sipura to fwd directly, though. |
17:27.52 | fugitivo | morning |
17:28.02 | podzap | Ariel_: could you possibly share some configuration with me? |
17:28.45 | Ariel_ | podzap, did you go into the fwd web and asked for your account to be iax? |
17:28.53 | *** join/#asterisk mrverbose (i=YQ0flwgF@i53874332.versanet.de) |
17:28.53 | podzap | Ariel_: yes |
17:28.54 | mrverbose | hi |
17:29.01 | Ariel_ | the samples form there web site works fine. |
17:29.08 | fugitivo | hello |
17:29.12 | Ariel_ | fugitivo, buena tarde |
17:29.21 | fugitivo | buenas tardes |
17:29.22 | podzap | Ariel_: i can not get it to work |
17:29.32 | mrverbose | is it possible to make asterisk gateway for voip telephones into ordinary isdn phone net? |
17:29.36 | fugitivo | you're right, it's 2:30pm here |
17:29.48 | podzap | Ariel_: could you share some config files with me? |
17:29.56 | fugitivo | mrverbose: yes, with the correct hardware |
17:29.57 | Ariel_ | fugitivo, it's 1:30 pm here |
17:30.19 | mrverbose | fugitivo: i have 2 hfcs-usb isdn cards and a voip phone |
17:30.51 | fugitivo | podzap: i think there's a problem with fwd |
17:30.59 | nain | Ariel_: Hi, I installed mpg123 but still problem is same |
17:31.12 | fugitivo | podzap: mine isn't registering, some time ago it was working |
17:31.27 | fugitivo | Ariel_: fwd is working for you? |
17:31.28 | podzap | fugitivo: i can get it registered, but the call won't work |
17:31.43 | fugitivo | hmm, i need to check my account status then |
17:31.45 | podzap | fugitivo: it always either gives all circuits busy |
17:31.57 | podzap | from sipura to fwd directly, it works fine |
17:32.16 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
17:32.18 | podzap | why don't more people post working examples of * setups? |
17:32.32 | Qwell | podzap: there are hundreds of examples out there |
17:32.45 | fugitivo | podzap: the example that fwd gives you works fine |
17:33.05 | podzap | Qwell: there is not one complete example for sipura -> * -> NAT -> FWD |
17:33.09 | Ariel_ | podzap, http://pastebin.ca/22046 |
17:33.09 | Qwell | podzap: if you're able to call fwd from your sipura, without going through asterisk, your account is setup for SIP |
17:33.19 | Ariel_ | it was yesterday I have not tried it today. |
17:33.31 | podzap | Qwell: fwd accounts can be setup for multiple protocols |
17:33.43 | Qwell | you sure about that? |
17:33.49 | nain | Ariel_: Hi, i m back after testing mpg123 with asterisk but still i m getting the same fd error ? |
17:33.53 | podzap | Qwell: i read it from their website |
17:33.58 | DaPrivateer | i am setup for IAX on FWD and I usually can login with X-Lite |
17:34.07 | Qwell | multiple protocols simultaneously? |
17:34.07 | fugitivo | podzap: check, that your iax account is not the same as your sip account |
17:34.30 | DaPrivateer | but as far as i can tell FWD is still down |
17:34.44 | podzap | Qwell: of course not simultaneously |
17:35.10 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
17:35.11 | podzap | fugitivo: i signed up for one iax account and one sip only account |
17:36.28 | podzap | Ariel_: can you give me your entire /etc/asterisk directory, minus your passwords, of course... |
17:36.38 | podzap | i am tired of this not working. |
17:36.39 | Ariel_ | podzap, no |
17:36.49 | fugitivo | podzap: that's being lazy |
17:37.02 | podzap | lazy, i've been fucking with this thing for 4 days now |
17:37.22 | Ariel_ | podzap, I have too many custom settings I would have to take at least 2 hours to get it to where you can make sence of it. |
17:37.34 | fugitivo | podzap: http://www.freeworlddialup.com/content/view/full/1501/ |
17:37.37 | Qwell | podzap: how about you post YOUR configs? |
17:37.54 | podzap | Qwell: well, OK, it's A@H, so there is about 50 config files. |
17:37.58 | Qwell | ... |
17:38.03 | *** join/#asterisk audela (n=audela@82-33-115-145.cable.ubr08.azte.blueyonder.co.uk) |
17:38.07 | Qwell | theres your problem |
17:38.08 | podzap | Qwell: but I'd be glad to do it, if you would care to analyze them. |
17:38.09 | fugitivo | nice |
17:38.10 | fugitivo | lol |
17:38.25 | podzap | I am trying to use A@H, what's wrong with that? |
17:38.36 | podzap | is it a piece of shit, or what? |
17:38.55 | Qwell | are you using the GUIs to make the changes to the configs? |
17:38.59 | fugitivo | no |
17:38.59 | Ariel_ | podzap, no it's not it works fine on it |
17:39.04 | podzap | I am using the GUI |
17:39.07 | Ariel_ | I have it working on a@H as well |
17:39.10 | fugitivo | it's just difficult to help somebody that uses aah |
17:39.29 | Ariel_ | fugitivo, no it's not. |
17:39.34 | podzap | fugitivo: OK, I could use the normal * if I had a set of config files to start with then. |
17:39.43 | *** join/#asterisk zoo (i=nobody@ip-172-16.travedsl.de) |
17:39.46 | Qwell | * comes with sample configs |
17:40.00 | Ariel_ | podzap, on the web site for asterisk@home there is a doc that has how to setup fwd via iax setting for amp |
17:40.13 | podzap | I followed the asteriskathome handbook to the letter to setup with fwd, and it DOES NOT WORK |
17:40.17 | podzap | :-) |
17:40.46 | Ariel_ | podzap, the iax server of there is an experiment and they don't always have it working correctly |
17:41.03 | podzap | Ariel_: is yours working now, yes or no? |
17:41.36 | Ariel_ | podzap, I have had it registered with them for over 8 months it could be that new settings are not up there yet for yours. |
17:42.03 | podzap | Ariel_: I am willing to bet that this is a * configuration problem. |
17:42.10 | fugitivo | mine is not working |
17:43.07 | podzap | i get the idea that people are trying to make a lot of money with * "consulting" |
17:43.09 | Ariel_ | podzap, what is your number? let me call it to see what I get |
17:43.19 | Ariel_ | podzap, really |
17:43.45 | podzap | Ariel_: you mean the one that is direct sipura -> fwd, or the * number? |
17:44.11 | Ariel_ | podzap, the one to asterisk. sipura is an different number? |
17:44.34 | fugitivo | service numbers are not answering for me |
17:44.37 | sivana | is it safe to use the lastest SpanDSP version? |
17:44.47 | sivana | ie: pre20 |
17:44.51 | podzap | i have two lines on the sipura, one to * and one directly to fwd. it takes me a minute to enable the * again. |
17:44.55 | Ariel_ | sivana, I have pre18 |
17:45.12 | Ariel_ | podzap, never mind it's not working for other then 800 numbers for mine. |
17:45.26 | Ariel_ | They must be having issue's with the iax server |
17:45.34 | fugitivo | Ariel_: can you receive calls? |
17:45.52 | Ariel_ | don't know 65342 |
17:46.50 | fugitivo | nothing? |
17:46.59 | Ariel_ | I guess there down |
17:47.03 | fugitivo | yep |
17:47.16 | Ariel_ | podzap, it's not your settings it's FWD |
17:47.37 | podzap | Ariel_: do you know somebody else i could test with, then? |
17:48.14 | Ariel_ | for iax free account or for sip? |
17:48.34 | podzap | either one i could get working; I am behind NAT. |
17:48.38 | *** join/#asterisk huslage (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
17:48.38 | fugitivo | podzap: stanaphone gives you a free new york number, but it's sip |
17:48.39 | Ariel_ | podzap, you can get a free sip account from stanaphone and sipphone |
17:49.18 | podzap | Ariel_: well, my one of my sipura lines does sip to fwd if i enable the nat settings and outbound proxy. |
17:49.35 | podzap | can you enable nat and outbound proxy with * and fwd? |
17:49.40 | doolph | what's the latest zaptel version |
17:49.50 | Ariel_ | sip behind nat you need to put in the sip.conf externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.0 |
17:50.23 | podzap | i can't edit sip.conf with asteriskathome, or I will go to hell and die |
17:50.31 | Ariel_ | doolph, I am using the one from cvs head for zaptel and then putting in the echo cansule from KB1 works great with stable |
17:50.34 | podzap | according to some in this channel... |
17:50.55 | Ariel_ | podzap, yes you can and it will not die |
17:51.01 | doolph | zaptel.c:4083: error: dereferencing pointer to incomplete type |
17:51.36 | Ariel_ | A@H only changes with the gui sip_additional.conf |
17:51.55 | podzap | can I just remove all those A@H files, and just put some normal * files there and will it still work? |
17:52.03 | doolph | why my debian doesnt want to compile zaptel |
17:52.04 | podzap | i am tired of this gui thing |
17:52.11 | Ariel_ | podzap, yes |
17:52.13 | doolph | it's because I dont have zaptel hardware? |
17:52.26 | podzap | Ariel_: where can i get a working set of files? |
17:52.30 | Ariel_ | did you comment out the #ztdummy |
17:52.53 | *** join/#asterisk omarc55 (n=omarc55@200.63.200.121) |
17:52.54 | Ariel_ | podzap, just copy the ones out of there and then go to /usr/src/asterisk and do make samples |
17:53.21 | podzap | Ariel_: OK |
17:53.35 | doolph | nop |
17:53.48 | doolph | no |
17:53.49 | doolph | oh yeah |
17:53.53 | podzap | that A@H shit is waaaaay to complicated for me. i want like 4 simple files, that's it. |
17:54.36 | Ariel_ | wow a gui is too complicated hummm sounds like either a dos person or a lunix command prompt guy to me. |
17:54.44 | omarc55 | hi all, does anybody know how to use iptables? I have 2 internet connections and I want to pass IAX2 calls through one connection and data through the rest, does anybody know how to do this? |
17:54.51 | podzap | Ariel_: yes, that's me. |
17:55.34 | Ariel_ | iax2 is udp port 4569 |
17:55.45 | fugitivo | omarc55: there're a lot of examples on the net |
17:56.11 | omarc55 | yeah, I know, I've tried a few but can't get it to work, and I am using udp port 4569 |
17:56.32 | podzap | Ariel_: that A@H, if you look at e.g. iax.conf, the include iax_additional.conf is commented out, but that is what the GUI writes to. How the hell does * use it, when it's commented out? |
17:56.36 | fugitivo | iptables -A interface0_in -p udp -m udp --dport 4569 -j ACCEPT |
17:56.51 | fugitivo | iptables -A interface0_in -p udp -m udp --dport 5036 -j ACCEPT |
17:56.56 | Ariel_ | the # is not a commenting it out |
17:57.05 | podzap | AHA |
17:57.13 | podzap | what is it, then? |
17:57.24 | Ariel_ | #include =>/external/file |
17:57.31 | Ariel_ | the ; will comment it out |
17:57.33 | podzap | like a C include line |
17:57.47 | omarc55 | fugitivo: isn't that for port forwarding? I am trying to use 1 internet connection just for IAX2 |
17:58.03 | fugitivo | omarc55: no, it's not |
17:58.27 | fugitivo | omarc55: replace interface0_in for whatever your rule is called |
17:58.50 | podzap | Ariel_: ok, i did `make samples'. I got 41 new files into /etc/asterisk. How many of those can I remove? What is the bare minimum which I need to keep? |
17:59.10 | Ariel_ | ~docs |
17:59.11 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
17:59.53 | omarc55 | fugitivo: not sure what my rule is called, I have the whole bunch of rules I am trying to get to work, I can put it in a paste bin... |
18:00.14 | Ariel_ | podzap, don't know what you want to work with. But if you don't want to use mgcp.conf then noload them in the modules.conf file |
18:00.31 | Ariel_ | podzap, there are more then 4 files you will need to use. |
18:01.03 | podzap | Ariel_: I only want to send and recieve calls to fwd |
18:01.09 | podzap | from one phone |
18:01.44 | omarc55 | http://www.ninethsense.com/praveen/pastebin/index.php?p=245 |
18:01.55 | podzap | i want the smallest configuration which I can get. |
18:02.16 | *** join/#asterisk pauldy (n=pauldy@c-67-162-255-229.hsd1.tx.comcast.net) |
18:02.42 | Ariel_ | podzap, then why not just use the sipura then |
18:03.39 | podzap | Ariel_: starting to sound like a good idea |
18:04.29 | podzap | i mean how in the hell is a new person supposed to learn how to setup a software which has 41 configuration files? |
18:04.47 | zip_ | rtfm |
18:04.49 | zip_ | like the rest |
18:04.51 | Ariel_ | podzap, by using it and reading |
18:05.18 | *** join/#asterisk Roldyx (n=Roldyx3@201.255.105.250) |
18:05.20 | Roldyx | hi |
18:05.26 | Ariel_ | funny thing when I started with asterisk over 3 years ago there was not wiki. or at least it was just starting. It was allot harder then. |
18:05.37 | Roldyx | but no rule 't' in context 'default' |
18:05.55 | Roldyx | ""but no rule 't' in context 'default'"" <-- what means this? |
18:06.13 | fugitivo | Roldyx: you didn't set a t rule in your context default |
18:06.18 | Ariel_ | Roldyx, it means there is no time out rule in your default context |
18:06.35 | Roldyx | fugitivo, !! |
18:06.55 | Roldyx | fugitivo, en español |
18:07.00 | [Airwolf] | podzap, start slowly. First make one thing work, then another and so one. After a few things, look at how it can be done better and then move on. |
18:07.18 | Hmmhesays | wham bam another successful asterisk install |
18:07.26 | fugitivo | Roldyx: exten => t,1,Hangup() for example |
18:07.34 | Hmmhesays | and i've paid for my drinking tonight |
18:07.37 | Hmmhesays | and gas to the lakes |
18:07.49 | [Airwolf] | Hmmhesays, did that last week |
18:07.52 | [Airwolf] | :P |
18:07.57 | fugitivo | Roldyx: es la regla que buscara asterisk cuando se produzca un timeout |
18:08.30 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-42-166.dsl.irvnca.pacbell.net) |
18:08.33 | Roldyx | fugitivo, esa regla la debo poner en extensions? |
18:08.42 | fugitivo | Roldyx: si |
18:08.46 | tainted_ | anyone use 'screen' to keep asterisk console up |
18:09.06 | Ariel_ | tainted_, why do you need to do that use safe_asterisk |
18:09.07 | faa_ | tainted_ whaT? |
18:09.09 | Roldyx | fugitivo, una sola linea es? |
18:09.18 | tainted_ | what is safe_asterisk |
18:09.22 | fugitivo | Roldyx: podes utilizar la cantidad que quieras |
18:09.28 | tainted_ | is it in 1.0.7? |
18:09.30 | tainted_ | i'm using .7 |
18:09.36 | fugitivo | Roldyx: t,1 t,2 t,3 etc |
18:09.37 | Roldyx | fugitivo, pero si pongo esa linea ya lo soluciono? |
18:09.41 | Ariel_ | tainted_, ahh stop asterisk then start it with safe_asterisk then connect to it via asterisk -r |
18:09.55 | fugitivo | Roldyx: si, pero usa una linea que sirva a tu proposito |
18:10.23 | Roldyx | fugitivo, el tema es que aso le aparece a mi amigo cuando yo lo llamo |
18:10.33 | fugitivo | Roldyx: tiene un ivr tu amigo? |
18:10.35 | Roldyx | fugitivo, a mi no me pasa |
18:10.52 | Roldyx | fugitivo, solo instalamos el asterisk |
18:11.22 | *** join/#asterisk roulduke_ (i=kw374vri@p508D0ADF.dip0.t-ipconnect.de) |
18:11.32 | fugitivo | Roldyx: eso no afecta en nada el funcionamiento, pero es mejor si lo tiene |
18:11.35 | tainted_ | Ariel_ oh wow |
18:11.49 | Roldyx | fugitivo, por que podria dar timeout? |
18:11.55 | *** join/#asterisk Legend` (n=legend@24.244.142.133) |
18:12.13 | Ariel_ | tainted_, what OS do you have? |
18:12.22 | fugitivo | Roldyx: por ejemplo si tiene un IVR, y el que hace el llamado no ingreso ningun DTMF, el asterisk salta al timeout para saber que tiene que hacer |
18:12.50 | *** part/#asterisk zoo (i=nobody@ip-172-16.travedsl.de) |
18:13.24 | omarc55 | fugitivo: were you able to make sense of what I sent? |
18:13.32 | Roldyx | fugitivo, IVR? |
18:13.42 | Roldyx | me desconciertas |
18:13.48 | fugitivo | Roldyx: interactive voice response, un menu interactivo |
18:14.06 | tainted_ | Ariel_ linux |
18:14.15 | Roldyx | fugitivo, y como le digo que lo saque? |
18:14.16 | fugitivo | Roldyx: cuando llamas y la operadora te dice, presione 1 para soporte, 2 para tal cosa, etfc |
18:14.22 | Roldyx | aha |
18:14.25 | fugitivo | Roldyx: eso es un IVR |
18:14.26 | Roldyx | entendi de primera |
18:14.31 | Ariel_ | tainted_, yes which distro ... OS |
18:14.42 | Roldyx | fugitivo, como puedo sacarlo? |
18:14.45 | fugitivo | Roldyx: entonces si el que llama, no ingresa ningun numero en un tiempo, asterisk va a saltar a la regla de timeout |
18:14.47 | Roldyx | saco el modulo? |
18:14.54 | fugitivo | sacar que? |
18:15.02 | Roldyx | para que no conteste |
18:15.19 | Roldyx | o sea para que atienda directamente el asterisk |
18:15.19 | fugitivo | el ivr? |
18:15.22 | Roldyx | sip |
18:15.29 | tainted_ | Ariel_ fedora |
18:15.44 | Roldyx | fugitivo, mi amigo no quiere usar IVR |
18:15.47 | fugitivo | busca exten => s,1 eso es lo que hace atender al asterisk reemplazalo por lo que quieras |
18:16.02 | Roldyx | mira |
18:16.17 | Ariel_ | tainted_, then you can go into the /usr/src/asterisk and do ... make config that way when you reboot it will load it automaticlly for you. |
18:16.35 | fugitivo | Roldyx: podes poner exten => s,1,Dial(SIP/1234,30,t) para que cuando el asterisk atienda, llame al interno 1234 |
18:16.48 | doolph | ......... |
18:16.52 | Roldyx | fugitivo, excelente! |
18:16.57 | Roldyx | fugitivo, ya entendi |
18:17.49 | Roldyx | :D |
18:17.54 | Roldyx | gracias again |
18:17.57 | Ariel_ | fugitivo, your a better man for that exchange then I. |
18:18.04 | fugitivo | :) |
18:18.12 | Roldyx | fugitivo, por que usan SIP en lugar de IAX? |
18:18.24 | tainted_ | Ariel_ does it require sendmail for NOTIFY=email@address.com to work? |
18:18.35 | Ariel_ | tainted_, yes |
18:19.04 | fugitivo | Roldyx: depende de lo que utilices, si utilizas telefonos IAX, entonces seria Dial(IAX2/1234,40,t) |
18:19.33 | Roldyx | fugitivo, el tema es que nosotros no usamos telefonos |
18:19.40 | Roldyx | fugitivo, NO TENEMOS PLACAS |
18:19.48 | Roldyx | sorry por las mayusculas |
18:19.50 | tainted_ | Ariel_ so how do u get asterisk to run on startup? |
18:20.12 | Qwell | tainted_: there are init scripts in the contrib/ dir |
18:20.16 | Ariel_ | did you read what I posted to you earler? |
18:20.36 | tainted_ | yea make config |
18:20.38 | Ariel_ | tainted_, just go to the /usr/src/asterisk directory then make config |
18:20.39 | fugitivo | Roldyx: utilizan softphones? |
18:20.44 | Roldyx | nop |
18:20.49 | Roldyx | fugitivo, CLI |
18:21.06 | fugitivo | ok, entonces reemplaza el comando por el que desees usar |
18:21.11 | fugitivo | yo te di un ejemplo |
18:21.12 | Roldyx | fugitivo, no encuentro un softphone como la gente |
18:21.16 | Ariel_ | tainted_, if you have RH/CentOS/Fedora it will make the script for you that on boot up will load it all for. |
18:21.29 | fugitivo | Roldyx: para windows o linux? |
18:21.34 | Roldyx | linux |
18:21.47 | Ariel_ | oh boy using asterisk as a soft phone.... |
18:21.53 | fugitivo | Roldyx: xlite |
18:22.34 | Roldyx | fugitivo, qt o gtk? |
18:22.43 | Ariel_ | strange they were dialing via iax to each others box... neat.. |
18:22.44 | tainted_ | Ariel_ what script would that be? |
18:22.51 | fugitivo | Roldyx: no lo utilice, no es opensource, pero si gratuito |
18:22.54 | tainted_ | Ariel_ the filename.. maybe i already have it |
18:22.58 | fugitivo | Roldyx: creo que es gtk |
18:23.28 | Ariel_ | tainted_, even if you already have it. It will not hurt just in case. |
18:23.55 | tainted_ | found it.. /etc/rc.d/init.d/asterisk |
18:24.11 | Ariel_ | yep |
18:27.11 | tainted_ | mmm.. i have 15 asterisk processes now |
18:27.16 | tainted_ | after i rebooted |
18:27.16 | tainted_ | lol |
18:28.24 | razu | anyone have experience with linksys PAP2 echo problem ? |
18:28.35 | fugitivo | razu: i don't have echo problem with the pap2 |
18:29.24 | razu | hmm |
18:29.35 | fugitivo | razu: how do you know that it's a pap2 problem? |
18:29.50 | razu | cause everybody with pap2 is wining :P |
18:30.02 | Qwell | wining? |
18:30.03 | bkw__ | tainted_, thats 15 threads boi |
18:30.12 | shido6 | erf? |
18:30.20 | shido6 | my pap2 sounds better than my 7960 |
18:30.29 | fugitivo | mine works great |
18:30.39 | doolph | where did you get that pap2 |
18:30.47 | doolph | everywhere is solt out |
18:30.56 | shido6 | I have 9k in stock |
18:30.57 | razu | strange ... |
18:31.02 | fugitivo | shido6: how much? |
18:31.03 | doolph | really |
18:31.07 | doolph | I want 2 |
18:31.10 | shido6 | $70 |
18:31.15 | doolph | bah |
18:31.17 | doolph | too expensive |
18:31.24 | fugitivo | razu: how are you using the pap2? zap calls? or 100% sip calls? |
18:31.26 | Qwell | shido6: Do I want to know why you have 9k pap2's? |
18:31.27 | PakiPenguin | shido6, i have a pap2 , how do i put in a stun in there? |
18:31.29 | fugitivo | shido6: expensive |
18:31.29 | shido6 | then order from teh sold out |
18:31.38 | doolph | 2100 for $64.95 |
18:31.56 | fugitivo | shido6: how much for 100 ? |
18:31.59 | Qwell | $70 for a pap2-na is pretty damn good |
18:32.01 | razu | i'm testing here right now and pap2 doesn't have an echo really ... but the other number where i called gets an echo :S |
18:32.02 | Ariel_ | doolph, get the 2002 instead |
18:32.11 | razu | fugitivo : 100% sip calls |
18:32.26 | fugitivo | razu: what codec? |
18:32.26 | doolph | Ariel_ what's the difference 2000 with 2002 |
18:32.28 | shido6 | go to the admin interface, PakiPenguin |
18:32.46 | doolph | 2002 is $5 more expensive |
18:32.54 | PakiPenguin | okay , am in admin interface |
18:32.55 | Ariel_ | doolph, the 2002 has a updated dps and can do 2 g729 calls. the 2100 has a wan port that sucks |
18:33.08 | doolph | ohhh |
18:33.10 | razu | fugitivo : G711a |
18:33.12 | doolph | but i dont care |
18:33.15 | doolph | i will use g723 |
18:33.16 | doolph | heh |
18:33.23 | fugitivo | razu: is your bandwitdh ok? |
18:33.23 | Ariel_ | doolph, same thing |
18:33.29 | razu | fugitivo : yep |
18:33.40 | Ariel_ | 2002 will do two not one. the 2000 only can do one |
18:33.47 | doolph | that's why |
18:33.55 | doolph | 5 more expensive |
18:34.22 | fugitivo | razu: try a softphone and check if there's echo |
18:34.36 | razu | ok |
18:35.27 | fugitivo | anybody downloaded the free linspire? :) |
18:37.35 | mog_home | debian...? |
18:37.53 | Qwell | fugitivo: I try to only use real Linux |
18:39.38 | fugitivo | i was thinking on my mother for linspire |
18:39.53 | mog_home | ubuntu |
18:41.27 | Ariel_ | I have not seen linspire but for desktops I like ubuntu for gnome or Mepis for KDE setup. |
18:41.39 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
18:42.12 | doolph | i hate X |
18:42.16 | doolph | I prefere windows |
18:42.17 | doolph | heh |
18:42.19 | [Airwolf] | ubuntu rules for desktop |
18:42.35 | fugitivo | what's different about ubuntu? |
18:42.52 | tainted_ | it's spelled differently from debian |
18:42.59 | [Airwolf] | gehe |
18:43.28 | Ariel_ | actually Mepis and ubuntu are both based on Debian |
18:43.45 | fugitivo | i can make my gentoo a great desktop too |
18:43.55 | Ariel_ | fugitivo, then do that |
18:43.57 | Roldyx | fugitivo, |
18:44.23 | fugitivo | but linspire seems to be nice for people that doesn't use linux |
18:44.24 | Ariel_ | fugitivo, you made a new friend. |
18:44.44 | tainted_ | why are there 15 asterisk threads |
18:45.32 | bkw__ | different things require threads |
18:45.54 | doolph | i am getting error compiling asterisk 1.2 duh |
18:46.11 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
18:46.22 | tainted_ | etc/init.d/asterisk generates 15 threads |
18:46.27 | tainted_ | just wanted to make sure that is kosher |
18:46.33 | bkw__ | yes |
18:46.41 | bkw__ | thats normal when your ps output shows each thread |
18:46.48 | bkw__ | load the box up with calls |
18:46.52 | bkw__ | you'll see even more |
18:47.08 | bkw__ | I only see one thread on mine |
18:47.09 | bkw__ | :P |
18:47.15 | robin73 | hi, there |
18:47.15 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0dh.dialup.mindspring.com) |
18:47.16 | doolph | yeah me too |
18:47.20 | doolph | lol |
18:47.22 | tainted_ | why does it only show one when i 'asterisk -vvvvgc' from command prompt |
18:47.23 | bkw__ | because we use NPTL threads |
18:47.37 | bkw__ | tainted_, it should still show them all |
18:47.50 | tainted_ | only one! |
18:47.55 | tainted_ | weird |
18:47.58 | doolph | bkw_ |
18:48.16 | robin73 | i'am working on a web admin interface for asterisk 1.2b. is there already good opensource web admin UI ? |
18:48.34 | bkw__ | robin73, If I had 10 bucks for everytime I have heard someone say that |
18:48.36 | bkw__ | I would be RICH |
18:48.45 | doolph | heh |
18:48.46 | Ariel_ | robin73, I like amp |
18:48.55 | tainted_ | lol |
18:48.57 | *** join/#asterisk iswm (i=iswm@unaffiliated/iswm) |
18:49.01 | doolph | pbx_dundi.c:31:18: zlib.h: No such file or directory |
18:49.07 | robin73 | yes amp is good, but not complete |
18:49.14 | doolph | asterisk:/usr/src/asterisk-1.2.0-beta1# locate zlib.h |
18:49.14 | bkw__ | doolph, install zlib-devel |
18:49.15 | tainted_ | i've got some pretty sweet billing written in .NET |
18:49.16 | robin73 | i'am using asterisk@home |
18:49.23 | bkw__ | you needs zlib headers boi |
18:49.24 | doolph | /usr/include/linux/zlib.h |
18:49.29 | bkw__ | not the same |
18:49.35 | doolph | isnt that one? |
18:49.38 | bkw__ | no |
18:49.38 | tainted_ | getting asterisk to talk to MS SQL was a bitch |
18:49.41 | bkw__ | thats in the kerenl boi |
18:49.48 | bkw__ | tainted_, no its not.. its really easy |
18:49.53 | robin73 | any other suggestion for a good existing WEN UI ? |
18:50.08 | bkw__ | tainted_, but I was also the first person to bring ODBC to asterisk :P |
18:50.10 | tainted_ | bkw_ through AGI i mean |
18:50.17 | bkw__ | oh thats simple too |
18:50.27 | bkw__ | I can do that in three lines of perl |
18:50.28 | doolph | thnx bkw_ |
18:50.31 | tainted_ | i've resorted to freetds & DBD::Sybase |
18:50.46 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
18:50.47 | bkw__ | yes but you must use freetds 0.62 I think |
18:51.03 | bkw__ | or is it 0.63 but they made chnages it no longer works correctly |
18:51.05 | tainted_ | freetds-0.63/ |
18:51.12 | bkw__ | any higher version will not work |
18:51.14 | tainted_ | that's what im using |
18:51.17 | doolph | bkw_ there's anyway to see a calltimer in a call |
18:51.18 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
18:51.41 | tainted_ | bkw_ where did u stick $SYBASE env var |
18:51.41 | bkw__ | I want realtime cdr events every X seconds |
18:51.43 | PakiPenguin | tainted_, is your .net app opensource? |
18:51.58 | doolph | bkw_ does it exist? |
18:52.01 | bkw__ | tainted_, its all wrapped in this perl module we wrote |
18:52.14 | file[laptop] | it's beautiful |
18:52.16 | tainted_ | see.. not even close to three lines |
18:52.19 | tainted_ | heh |
18:52.30 | bkw__ | tainted_, yes in my perl scripts I do use SQLSERVER; |
18:52.34 | tainted_ | i got pissed at mysql and kicked that bitch to the curb |
18:52.49 | bkw__ | my $db = init_isp(); |
18:52.55 | tainted_ | lol |
18:52.58 | doolph | i dont need mpg123 for asterisk 1.2 eh |
18:53.00 | tainted_ | cheater |
18:53.03 | bkw__ | $db->do("blah"); |
18:53.08 | bkw__ | done |
18:53.10 | bkw__ | NEXT!!! |
18:53.11 | bkw__ | haha |
18:53.21 | bkw__ | or |
18:53.24 | tainted_ | sigh |
18:53.32 | bkw__ | my $sql = "some sql query"; |
18:53.34 | doolph | bkw_ how you can see the calltime in a call |
18:53.36 | bkw__ | foreach my $obj (@{$db->sql2allhash($sql)}) { do something } |
18:53.36 | tainted_ | i tore my hair out getting that shit to work |
18:53.43 | mog_home | bkw_ |
18:53.44 | faa_ | do anyone use cisco call manager? or cisco call manager express? |
18:53.47 | mog_home | i will pay |
18:53.48 | mog_home | you |
18:53.53 | fugitivo | faa_: #cisco |
18:53.54 | mog_home | to go to mongolia |
18:53.54 | bkw__ | for? |
18:53.55 | mog_home | find a monk |
18:53.58 | mog_home | and learn |
18:54.01 | mog_home | PATIENCE |
18:54.02 | mog_home | ^_^ |
18:54.05 | tainted_ | lol |
18:54.08 | bkw__ | why? |
18:54.12 | mog_home | no reason |
18:54.12 | PakiPenguin | lol |
18:54.12 | file[laptop] | patience is overrated |
18:54.14 | mog_home | ^_^ |
18:54.19 | bkw__ | don't make me bitch smack you |
18:54.24 | tainted_ | patience is for the weak |
18:54.32 | tainted_ | we need more progress |
18:54.33 | mog_home | patience is for the strong |
18:54.40 | mog_home | like tolerance |
18:54.48 | mog_home | the weak can be intolerant and impatient |
18:55.42 | tainted_ | patience is waiting for spilled water to evaporate rather than wiping it up |
18:55.49 | mog_home | lol |
18:55.51 | mog_home | no |
18:56.00 | mog_home | patience is when someone spills toxic waste |
18:56.05 | Nivex | no I believe that is called laziness |
18:56.06 | Qwell | tainted_: no, thats just lazy :p |
18:56.08 | mog_home | and you call the guy to get it lcened |
18:56.13 | mog_home | cleaned* |
18:56.21 | mog_home | and you wait and make sure no one steps in it |
18:56.37 | Qwell | patience is waiting for the cable guy to come "sometime between 9 and 4" |
18:56.43 | Qwell | and then punching him in the face when he gets there |
18:57.04 | mog_home | lol |
18:57.05 | tainted_ | punching is barbaric |
18:57.11 | Qwell | okay, stabbing |
18:57.21 | tainted_ | beating him down with tire iron is more civilized |
18:57.24 | sivana | bkw__: it seems the sangoma cards can't handle data very well |
18:57.39 | bkw__ | sivana, what do you mean? |
18:57.41 | bkw__ | the TDM boards? |
18:57.47 | sivana | I went back to the TE405 and I was able to fax 2 pages |
18:57.53 | sivana | without hangup |
18:57.55 | sivana | ya |
18:58.00 | bkw__ | its your clocking |
18:58.10 | mog_home | digium is awesome ^_^ |
18:58.11 | bkw__ | we have a sangoma board for testing works great |
18:58.15 | sivana | well... the clocking is better with the 405 then |
18:58.21 | mog_home | but its all about clock |
18:58.28 | bkw__ | sivana, chances are you had the clock set wrong |
18:58.33 | mog_home | if clock is bad there will be no faxing |
18:58.35 | bkw__ | its not trivial to set that |
18:58.40 | tainted_ | shit |
18:58.41 | sivana | bkw__: you're faxing with the sangoma? |
18:58.43 | bkw__ | yes |
18:58.46 | tainted_ | forgot to register for astricon |
18:58.51 | doolph | Now I am getting compiling zaptel |
18:58.53 | mog_home | me thinks digium card is better |
18:58.56 | tainted_ | bkw_ what are your rates for writing customer stuff for asterisk |
18:58.57 | sivana | even the dial up pool was all fubared with the sangoma |
18:58.59 | bkw__ | we aren't even using zaptel with ours |
18:59.00 | mog_home | but they are very similar |
18:59.01 | tainted_ | custom |
18:59.11 | bkw__ | its not even setup on asterisk |
18:59.12 | bkw__ | :P |
18:59.15 | sivana | bkw__: fax machine -> FXS -> * -> PRI ? |
18:59.27 | doolph | what do i need to install zaptel |
18:59.33 | mog_home | me needs money for making asterisk code |
18:59.35 | bkw__ | sivana, you were using a TDM board? |
18:59.43 | bkw__ | TDM400P? |
18:59.46 | sivana | FXS = CAC Channel Bank |
18:59.52 | sivana | with the A104 |
19:00.02 | bkw__ | i bet you the clock was wrong |
19:00.03 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
19:00.08 | mog_home | doubt it |
19:00.14 | bkw__ | no I bet it was |
19:00.16 | mog_home | as he didnt change it when he got card |
19:00.19 | mog_home | if he did |
19:00.23 | mog_home | then he should have known |
19:00.26 | sivana | possibly... span 1 was getting clock from PRI and the other spans were getting it from 1 |
19:00.26 | bkw__ | no in the wanpipe driver you ninny |
19:00.28 | bkw__ | it was wrong |
19:00.42 | bkw__ | like I said that is an easy one to get wrong in there |
19:00.49 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
19:00.54 | sivana | bkw__: I have the span 2 set the clock reference to 1 |
19:01.12 | bkw__ | chances are you didn't need it like that |
19:01.14 | mog_home | taking timing |
19:01.21 | sivana | I dunno... didn't friggin work right though |
19:01.42 | mog_home | well thats awesome |
19:01.45 | sivana | my customers did notice increase in echo since going back to TE405 though |
19:02.02 | mog_home | whats your echocan set at sivana |
19:02.07 | sivana | KB1 now |
19:02.15 | mog_home | in zapata |
19:02.17 | tainted_ | bkw_ so u dumped mysql as well? |
19:02.20 | mog_home | echocan=? |
19:02.22 | sivana | yes |
19:02.26 | mog_home | bump it |
19:02.26 | sivana | echocan = yes |
19:02.27 | mog_home | up |
19:02.29 | mog_home | 128 |
19:02.30 | Hmmhesays | anyone know how to set no callerid for voipjet? |
19:02.32 | mog_home | or 256 |
19:02.44 | mog_home | or get one of those new sexy 406s ^_^ |
19:02.54 | sivana | I'll try next week... I just changed the type to KB1, so maybe that'll be better |
19:02.55 | mog_home | but bumping value up will fix it i bet |
19:03.36 | sivana | I need to wait until Monday now for the business volume to pick back up:) |
19:04.19 | sivana | bkw__: it certainly sounds like timing, like after 6 mins, things would go haywire |
19:04.36 | bkw__ | sivana, ya |
19:04.41 | sivana | bkw__: but I don't know enought that shit... I'm programmer not an engineer dammit :) |
19:04.58 | doolph | do I need kernel header to install zaptel? |
19:05.04 | JerJer | yes |
19:05.17 | mog_home | or complete kernel source ^_^ |
19:05.36 | sivana | I haven't tried faxing > 2 pages yet though, so maybe it's not 100% solved |
19:05.39 | *** join/#asterisk tessier (n=treed@wsip-68-224-172-77.sd.sd.cox.net) |
19:06.18 | sivana | but with the sangoma card, I was lucky to get 1 page out before comm error |
19:06.28 | doolph | mmm |
19:06.33 | doolph | that might be the problem |
19:06.33 | doolph | then |
19:07.13 | sivana | could have just been configuration issue though with the sangoma... I sent them an email about it and got no reply |
19:07.17 | doolph | mmm same errors |
19:07.28 | tainted_ | anyone going to astricon? |
19:07.45 | Damin | Yeah.. I'm supposed to be doing a talk on something, but I'm not sure what yet.. |
19:07.57 | doolph | /usr/include/linux/stat.h:68: error: storage size of `atime' isn't known |
19:08.00 | doolph | /usr/include/linux/stat.h:70: error: storage size of `ctime' isn't known |
19:08.02 | mog_home | and speaking i hope |
19:08.03 | tainted_ | Damin are you serious lol |
19:08.09 | doolph | /usr/include/linux/stat.h:69: error: storage size of `mtime' isn't known |
19:08.15 | sivana | anyone here from Manx yet? |
19:08.16 | doolph | any idea |
19:08.24 | sivana | s/here/hear |
19:08.26 | tainted_ | the speaker list doesn't seem too strong right now |
19:08.39 | sivana | ~seen manxpower |
19:08.43 | jbot | manxpower <i=eric@1Cust2920.an7.dfw28.da.uu.net> was last seen on IRC in channel #asterisk, 1d 10h 19m 50s ago, saying: 'Um, actually responsetimeout might be more what you are looking for.'. |
19:08.43 | mog_home | yeah he needs to update it |
19:08.47 | bkw__ | sivana, whats your email? |
19:09.01 | tainted_ | mog_home who is organizing |
19:09.15 | sivana | bkw__: richard@aspworld.com |
19:09.27 | *** join/#asterisk L|NUX (n=linux@202.5.145.14) |
19:09.33 | mog_home | ole and sokol |
19:10.02 | Damin | tainted_: Yeah.. I've got a couple of different topics into Olle and Steve, but I probably won't know which one they'll pick until I get there.. :) |
19:10.04 | Ariel_ | mog_home, an 5ess ATT switch type is a normal T1 for e&m not a pri system is it? |
19:10.19 | mog_home | yeah |
19:10.23 | tainted_ | Allison Smith: the voice of asterisk is a speaker!?!?!?! |
19:10.27 | mog_home | yeah |
19:10.31 | mog_home | she comes to all of them |
19:10.50 | tainted_ | Damin which topics? |
19:12.36 | Damin | tainted_: Either intro to asterisk tutorial or Deploying Asterisk for the SOHO market.. |
19:13.02 | Damin | tainted_: Or I could just get up there and spew profanity like Zoa did in Atlanta.. ;) |
19:13.17 | mog_home | i think i am gonna be part of a q and a session and hopefully talk about jabber and asterisk |
19:13.28 | tainted_ | Damin what was he swearing about |
19:15.02 | bkw__ | Damin, the soho one is great I think |
19:15.19 | bkw__ | mog_home, jabber and asterisk.. thats pretty funny |
19:15.42 | bkw__ | mog_home, what does res_jabber do now? |
19:15.43 | file[laptop] | \ |
19:15.48 | file[laptop] | very funny |
19:16.10 | mog_home | res_jabber does a lot |
19:16.27 | Damin | tainted_: Don't know.. I couldn't understand a whole lot of what he said other than "F**K" alot. ;) |
19:16.44 | mog_home | ill release it when it makes me smile |
19:16.48 | mog_home | and doesnt crash my box |
19:17.02 | file[laptop] | that would be a good thing |
19:17.06 | file[laptop] | crashes are usually frowned upon |
19:17.09 | tainted_ | astricon 2004 was pretty strong |
19:17.17 | Damin | file[laptop]: Unless you are a crash test dummy.. |
19:17.21 | mog_home | heh |
19:17.43 | Damin | tainted_: I'm looking forward to Anaheim.. |
19:17.45 | file[laptop] | nope, I'm not a crash test dummy |
19:17.49 | doolph | anyone can help me out with zaptel compiling |
19:17.49 | tainted_ | i'm going just to see the faces behind the characters |
19:17.50 | tainted_ | hahaha |
19:17.51 | bkw__ | mog_home, I could accomplish the same thing via AGI using Net::Jabber |
19:17.57 | doolph | I am using debian |
19:18.02 | tainted_ | bkw_ well put |
19:18.05 | mog_home | you could crash it? |
19:18.11 | bkw__ | no make it not crash |
19:18.16 | mog_home | and /me haates agi |
19:18.16 | bkw__ | via agi and net::jabber |
19:18.21 | bkw__ | agi has its place |
19:18.23 | bkw__ | even I hate it |
19:18.25 | mog_home | well it doesnt seem to go |
19:18.26 | bkw__ | but it does have its place |
19:18.33 | file[laptop] | omg this movie is evil |
19:18.37 | file[laptop] | if she dies I swear I will kill someone |
19:18.41 | tainted_ | it'd be sweet to stdout < NET::jabber |
19:18.42 | tainted_ | haha |
19:18.42 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
19:18.44 | bkw__ | file what are you watching? |
19:18.52 | file[laptop] | City of Angels |
19:18.56 | file[laptop] | on TBS |
19:19.08 | bkw__ | everyone stay away from file |
19:19.14 | file[laptop] | she's dead. |
19:19.18 | mog_home | agi is not the place for something this important... |
19:19.25 | mog_home | and this big i think |
19:19.31 | mog_home | or as big as i hope for it to be |
19:19.31 | glm2k | file[laptop]: i was just about to tell you that >:) |
19:19.31 | bkw__ | mog_home, yes it is.. its all in how you impliment it |
19:19.37 | tainted_ | mog javver is mission critical? |
19:19.37 | file[laptop] | bah |
19:19.38 | bkw__ | you can even do it via the manager |
19:19.40 | doolph | hello? |
19:19.46 | mog_home | lol |
19:19.47 | bkw__ | creating a custom manager event from the dialplan.. with an external jabber proxy |
19:19.50 | mog_home | yeah you could |
19:19.55 | mog_home | its how the other people are doing it |
19:19.59 | bkw__ | thats more safe than trying to stick it inside of asterisk |
19:20.11 | bkw__ | just because you can doesn't mean you should |
19:20.14 | bkw__ | unless you do it correctly |
19:20.23 | tainted_ | grasshoppa |
19:20.24 | mog_home | well youll see |
19:20.26 | mog_home | one day |
19:20.41 | mog_home | unless you go aeferion on us bkw |
19:21.03 | bkw__ | and i'm shocked you can even fucking spell that |
19:21.05 | bkw__ | I never can |
19:21.31 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
19:21.35 | mog_home | which is why no one ever gets to the site... |
19:21.45 | bkw__ | haha |
19:21.49 | bkw__ | I just said that today also |
19:21.57 | mog_home | they could use a better name |
19:22.43 | file[laptop] | I'm angry now |
19:22.49 | file[laptop] | what a mean mean ending |
19:23.08 | bkw__ | file thats why I haven't watched that movie since |
19:23.47 | wasim | or nextgen |
19:25.24 | doolph | /usr/include/linux/time.h:198: error: dereferencing pointer to incomplete type |
19:26.14 | file[laptop] | I am the very model of a modern major general |
19:26.18 | doolph | is that kernel problem or what |
19:27.36 | bkw__ | doolph, you seem to be having all kinds of problems |
19:27.42 | bkw__ | what distro are you using? |
19:28.38 | file[laptop] | dammit this movie has made me miserable now |
19:30.33 | Damin | file[laptop]: That movie really sucks... |
19:31.03 | doolph | debian |
19:31.21 | doolph | Linux asterisk 2.4.27-2-686 #1 Mon May 16 17:03:22 JST 2005 i686 GNU/Linux |
19:31.36 | bkw__ | oh lord help ya |
19:31.38 | bkw__ | go get a real distro |
19:31.40 | bkw__ | :P |
19:32.11 | zedkatuf | doolph: is that a pure debian distro? |
19:32.15 | doolph | yes |
19:32.45 | doolph | I got asterisk compile resolved already |
19:32.48 | doolph | now the problem is zaptel |
19:33.03 | doolph | i think that it's kernel heador or default kernel problem i dunno |
19:33.19 | doolph | usr/include/asm/mpspec.h:8: error: `MAX_MP_BUSSES' undeclared here (not in a function) |
19:33.25 | *** join/#asterisk tzafrir_laptop (n=tzafrir@CBL217-132-129-167.bb.netvision.net.il) |
19:35.31 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
19:35.38 | *** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com) |
19:36.37 | *** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net) |
19:36.57 | Borgon | Any of you use a vps/vds to run asterisk? |
19:37.09 | *** join/#asterisk SexyKen (n=ksandell@c-24-5-129-114.hsd1.ca.comcast.net) |
19:38.28 | Hmmhesays | yeah that brian guy on the mailing list is kind of a jerk |
19:39.01 | file[laptop] | Hmmhesays: just a little |
19:39.30 | Hmmhesays | hey file howsit |
19:39.35 | file[laptop] | peachy, you? |
19:39.48 | Hmmhesays | t'ain't bad, just finished an install, I think i'm going to take my motorcycle down to the lakes now |
19:39.55 | file[laptop] | that's crazy talk |
19:39.59 | file[laptop] | you should be a geek and stay on IRC |
19:40.02 | Hmmhesays | haha |
19:40.13 | file[laptop] | you know you wannnna |
19:40.14 | Hmmhesays | I am a geek, maybe not so much as you |
19:40.18 | Hmmhesays | fanta? |
19:40.24 | file[laptop] | yes! |
19:42.13 | *** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk) |
19:42.46 | bkw__ | Hmmhesays, you evil bastard |
19:42.58 | blitzrage | bkw__: you know js right? |
19:43.00 | file[laptop] | bkw__: he's only slightly evil |
19:43.06 | file[laptop] | bkw__: RUN! |
19:43.09 | bkw__ | blitzrage, yes kinda why? |
19:43.12 | bkw__ | its not that hard |
19:43.20 | blitzrage | bkw__: I know |
19:43.34 | blitzrage | bkw__: I'm just curious if you've ever run into problems running JS from within an included PHP file? |
19:43.46 | blitzrage | i.e. index.php includes something.php, which contains JS |
19:44.03 | bkw__ | as long as its wrapped in a <script language=javascript></script> |
19:44.14 | bkw__ | firefox won't honor it if its just <script></script> |
19:44.37 | blitzrage | yep... it is... :( same script in the index.php works, in the included php I get "document.test.event is null or not an object" |
19:44.43 | blitzrage | bkw__: its IE |
19:44.43 | Ariel_ | mog_home, can I ask you some questions about connecting a te410p to an att 5ess switch |
19:44.47 | blitzrage | http://pastebin.com/353819 |
19:44.49 | mog_home | shoot |
19:44.52 | mog_home | i am a little busy |
19:44.56 | mog_home | but ill get back to ou |
19:45.14 | bkw__ | blitzrage, do you have javascript turned off? |
19:45.30 | bkw__ | check the parsed output |
19:45.36 | bkw__ | make sure it isn't missing anything |
19:45.37 | blitzrage | bkw__: nope - like I said, it works in the index.php page, just not the included page |
19:45.40 | bkw__ | does it work if you save it static? |
19:45.43 | blitzrage | bkw__: I know the script works :) |
19:45.48 | bkw__ | thats odd |
19:45.57 | Ariel_ | mog_home, they said to use em_w for the setup. But when they send there call I dont' get the digits asterisk just says no s exten found |
19:46.06 | blitzrage | bkw__: cp/pst to index.php, works. cp/pst to the included page, error |
19:47.19 | Borgon | Any of you use a vps/vds to run asterisk? |
19:47.32 | *** join/#asterisk Dybdahl (n=Lars@oelsted.dybdahl.dk) |
19:51.16 | bkw__ | bbl |
19:51.31 | mog_home | you using featd? |
19:52.18 | Ariel_ | mog_home, no. Here are my settings: http://pastebin.ca/22060 |
19:53.02 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
19:53.41 | *** join/#asterisk r0m (n=SysOp@bl8-28-113.dsl.telepac.pt) |
19:56.02 | *** join/#asterisk rjx (n=rj@ppp-71-138-89-44.dsl.irvnca.pacbell.net) |
19:56.14 | *** join/#asterisk file (n=jcolp@mctnnbsa30w-156034035106.nb.aliant.net) |
19:56.19 | Borgon | 128RAM decent enough to run a personal asterisk server with 1 call at a time? |
19:56.37 | fugitivo | yes |
19:56.47 | rjx | hey all, asterisk@home newblet here ;) wondering if someone can help me troubleshoot an issue w/ an incoming trunk using broadvoice |
19:56.48 | Nix | without X yes... |
19:57.42 | Ariel_ | rjx, you might get more help from #amportal |
19:57.48 | rjx | ok |
19:57.49 | rjx | thanx |
19:58.01 | rjx | hrm, opless and emptier :P |
19:58.32 | *** join/#asterisk tatum- (n=nospam@s142-59-92-194.ab.hsia.telus.net) |
19:58.36 | rjx | anyhow if anyone wants to take a look, summary is basically I dont seem to be registering, all the while i've tried a few register lines as well as settings changes |
19:58.37 | rjx | http://pastebin.com/353832 |
19:58.44 | *** part/#asterisk tatum- (n=nospam@s142-59-92-194.ab.hsia.telus.net) |
19:58.53 | rjx | get that over and over, as it keeps trying to re-register, but nothing is ever received |
19:59.53 | chet | are you supposed to be registering with your internal ip addressing? |
20:00.04 | chet | or useing your external ip? |
20:00.35 | rjx | i have externip set in sip.conf |
20:00.39 | chet | since you are most likely using nat |
20:00.43 | rjx | that i am |
20:00.46 | chet | its using 192.168 |
20:00.53 | chet | which is internal |
20:00.55 | rjx | right |
20:01.00 | bkw__ | h323 ROCKS!!! |
20:01.15 | rjx | maybe it didnt take my externip settings |
20:01.20 | doolph | ? |
20:01.22 | chet | i dont think it did |
20:01.31 | rjx | ok, thx, checking on that |
20:01.48 | chet | i assume you have to reg via the external ip so broadvoice can reach you |
20:02.26 | *** join/#asterisk tzafrir_ (n=tzafrir@CBL217-132-115-194.bb.netvision.net.il) |
20:02.36 | rjx | right |
20:02.49 | rjx | wouldnt that also imply a port forward would be needed on router? |
20:03.01 | rjx | in all howto's ive read, which imply being behind a nat, i havent seen it though |
20:05.01 | rjx | do i also need a nat=yes ? |
20:05.18 | chet | im not really sure, maybe someone else can help, sorry |
20:05.43 | rjx | it's ok, thx anyhow |
20:05.49 | rjx | same error though |
20:05.50 | chet | im new to * but that looked balatantly wrong |
20:05.50 | rjx | http://pastebin.com/353841 |
20:05.54 | rjx | yeah |
20:06.13 | rjx | well those arent necessarily *errors* but im not getting any response |
20:06.24 | rjx | so i dont know what a proper response looks like |
20:08.18 | *** join/#asterisk Junbug (i=Junbug@AC8A59D4.ipt.aol.com) |
20:09.24 | Junbug | is it poss for an external script php etc... to reload asterisk? |
20:10.37 | Ariel_ | Abort (6) on Primary D-channel of span 1 |
20:10.37 | Ariel_ | Sep 3 13:10:17 NOTICE[2398]: chan_zap.c:7977 pri_dchannel: PRI got event: HDLC |
20:10.49 | bkw__ | its a frame slip |
20:10.52 | Ariel_ | argh my settings are off for the pri |
20:10.57 | Ariel_ | frame slip |
20:10.59 | chet | rjx- using asterisk@home? |
20:11.11 | Ariel_ | bkw_, so how do I fix that |
20:12.51 | *** join/#asterisk r0d3nt (i=anonymou@tinfoilhat.net) |
20:13.03 | Junbug | can i use an external script php etc... to reload asterisk? |
20:13.58 | bkw__ | Ariel_, you can't really |
20:14.08 | Qwell | Junbug: asterisk -rx "reload" |
20:14.42 | Junbug | ok thx |
20:15.03 | rjx | yes chet |
20:16.05 | chet | does your config look like this one at the top? |
20:16.07 | chet | http://lists.digium.com/pipermail/asterisk-users/2004-November/074809.html |
20:16.15 | chet | nat=yes |
20:16.19 | Junbug | Qwell: quick q all aol members are banned in EFNET #linuxhelp? |
20:16.23 | Ariel_ | I need to get these pri's working. |
20:16.31 | chet | cause i see this"# |
20:16.32 | chet | Retransmitting #3 (no NAT):" |
20:17.23 | Qwell | Junbug: yep |
20:18.17 | rjx | yeah chet, i saw that and thought it was weird too |
20:18.23 | rjx | even when i put nat=yes it still says no nat |
20:18.26 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
20:18.34 | chet | did you reload after making changes? |
20:18.41 | chet | or restart service |
20:19.14 | rjx | yes |
20:19.22 | rjx | im noticing that when i do things via AMP |
20:19.27 | rjx | they dont save even if i hit apply etc |
20:19.36 | rjx | so im just now edit'ing configs by hand |
20:19.39 | rjx | saving, and reloading |
20:19.44 | rjx | now it says NAT |
20:19.45 | chet | you can always edit it manually and restart then |
20:20.02 | rjx | that's what im doing, still not registering tho hrm |
20:20.10 | rjx | but does say NAT now though |
20:20.17 | chet | paste your config at pastebin |
20:20.35 | rjx | yeah, 1 sec |
20:20.37 | rjx | got a new error |
20:21.22 | netsurfer | hey Ariel_ - that page answered all the questions.. got it workin now :) thanx |
20:21.46 | rjx | http://pastebin.com/353859 |
20:22.32 | chet | paste your config |
20:23.40 | rjx | http://pastebin.com/353862 |
20:23.54 | rjx | i removed the passes myself, in the config, they are in there where you see x's |
20:24.27 | *** join/#asterisk SexyKen (n=ksandell@c-24-5-129-114.hsd1.ca.comcast.net) |
20:26.17 | Ariel_ | bkw_, have configured any te410p? I need to fix this slip and they gave me what there switch is set too |
20:26.58 | chet | rjx, how did you change your ip before? |
20:27.07 | netsurfer | if I use #include filename.conf in extensions.conf can I duplicate contexts ? |
20:27.15 | rjx | in sip.conf, under general, i put externip = |
20:27.41 | Corydon76-home | netsurfer: the include simply inserts the text of that file |
20:27.54 | Corydon76-home | Yes, you can include the same file multiple times |
20:28.14 | Corydon76-home | but no, you can't duplicate contexts |
20:28.38 | *** join/#asterisk pbxbart__ (n=pbxbart@p54B00652.dip0.t-ipconnect.de) |
20:28.39 | netsurfer | Corydon76-home - so if I have [default] in main file and in an #include 'd file it is actually just like one big [default] context ? |
20:28.57 | Corydon76-home | Depends on which file |
20:29.30 | Corydon76-home | The best way to do that, though, is simply not to have ANY context definition in an included file |
20:29.56 | netsurfer | Corydon76-home - then where do the extensions end up? |
20:29.58 | Corydon76-home | Just extensions, then include those extensions into whatever contexts you want them |
20:30.15 | netsurfer | ah hmmm |
20:30.39 | netsurfer | sounds like it could get messier than what the file already is lol |
20:30.43 | Corydon76-home | However, it's probably better to create special contexts and "include" them instead of "#include"ing them |
20:31.16 | netsurfer | the idea was to keep internal extensions in a seperate file, to ease the admin a bit |
20:31.23 | Corydon76-home | i.e. [internal] include=office [incoming] include=office |
20:31.54 | Corydon76-home | Sure, you can do that |
20:32.12 | Corydon76-home | Just don't duplicate contexts |
20:32.15 | chet | rjx, can you paste your sip.conf? im not really sure why its not working |
20:32.23 | sivana | using perl, is it difficult to parse a .conf file? |
20:32.28 | Corydon76-home | [internal] #include "extensions/internal.conf" |
20:32.49 | Corydon76-home | sivana: you don't know perl, do you? |
20:32.55 | sivana | nope |
20:33.07 | Corydon76-home | sivana: then yes, it's going to be difficult for you |
20:33.08 | sivana | I have a perl script.. that I paid someone to write, but I need some mods done |
20:33.17 | rjx | chet, sure, 1 sec |
20:33.46 | chet | rjx, here is a link with working config, but its not broadvoice, but assuming you have that right it should work- http://www.voip-info.org/wiki-Asterisk+FWD+NAT+Config+Example |
20:33.55 | Corydon76-home | The file isn't that difficult to parse, no matter what language you use (unless you use a language that doesn't know how to open files) |
20:35.38 | *** join/#asterisk pauldy (n=pauldy@c-67-162-255-229.hsd1.tx.comcast.net) |
20:35.40 | sivana | ok |
20:35.55 | rjx | ok chet |
20:35.57 | rjx | http://pastebin.com/353871 |
20:36.15 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
20:36.43 | *** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au) |
20:37.31 | rjx | chet, im wondering about an auth_password using broadvoice |
20:37.42 | rjx | im using the password that I am logging in to their web system with |
20:37.51 | rjx | ive put in a helpdesk ticket |
20:38.02 | rjx | because under my device settings on the web, it doesnt have an auth pass listed |
20:38.42 | *** join/#asterisk used (n=used@c-24-22-125-179.hsd1.or.comcast.net) |
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20:45.57 | *** part/#asterisk n0ise (i=n0ise@d38-16-2.commercial1.cgocable.net) |
20:49.15 | *** join/#asterisk Phoebers (n=phoenix_@82-168-146-4-bbxl.xdsl.tiscali.nl) |
20:49.27 | Phoebers | good evening :) |
20:49.54 | L|NUX | evening |
20:50.19 | *** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net) |
20:52.41 | Phoebers | I have a question :P, how do I load config files? |
20:52.52 | L|NUX | reload sip |
20:52.57 | L|NUX | :) |
20:53.09 | *** join/#asterisk jeffik (n=Jeff@toronto-HSE-ppp3985137.sympatico.ca) |
20:53.37 | L|NUX | sip reload |
20:53.41 | L|NUX | extension reload |
20:53.45 | L|NUX | blah blah |
20:53.55 | Phoebers | conf files are auto used? |
20:54.35 | Phoebers | erm read |
20:54.35 | Phoebers | wotever u call it |
20:54.35 | *** join/#asterisk tzafrir_ (n=tzafrir@CBL217-132-115-193.bb.netvision.net.il) |
20:56.40 | rjx | im on hold w/ broadvoice |
20:56.44 | rjx | which seems like forever |
20:56.52 | rjx | and their onhold music makes me wanna punch a baby |
20:58.35 | chet | yeah, but once its working you will be happy |
20:58.55 | rjx | yeah some1 just came on and hung up on me |
20:58.59 | rjx | gaaaahdamn |
20:59.05 | *** join/#asterisk tzafrir__ (n=tzafrir@CBL217-132-72-158.bb.netvision.net.il) |
20:59.19 | chet | haha |
20:59.30 | *** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net) |
20:59.35 | chet | you would think they would post a working * config since it must be so common |
20:59.35 | rjx | i have an email in, maybe i'll just put this project to rest till i get a reply |
20:59.41 | rjx | oh they have one |
20:59.51 | rjx | i think i need a diff't password |
21:00.02 | rjx | ive tried like 10 diff configs, all similar |
21:00.46 | ericm_06 | rjx: remember, the web portal password is diff. then the auth. password |
21:00.57 | *** join/#asterisk ManxPower (i=eric@1Cust2442.an1.dfw28.da.uu.net) |
21:01.11 | rjx | ah ha! thanks |
21:01.14 | rjx | you just confirmed my thoughts |
21:01.28 | *** join/#asterisk huslage (n=huslage@c-24-22-125-179.hsd1.or.comcast.net) |
21:01.28 | rjx | i need to call support and have them change my device and get that pass |
21:01.33 | rjx | that'll probably fix my prob |
21:01.43 | rjx | well at least it's a step in the right direction |
21:01.59 | ericm_06 | well, good luck |
21:02.09 | rjx | their onhold music is the worst i think ive ever heard |
21:02.22 | rjx | *shuts down *@home box till he gets an email reply |
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21:21.22 | ManxPower | YAY! My main server is back up! |
21:22.01 | Qwell | ManxPower: congrats |
21:23.00 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
21:24.06 | ManxPower | All our AT&T T-1s are down, but the frame relay network is up |
21:24.50 | *** join/#asterisk BVEBVE (n=bvebve@virgo.minbas.cu) |
21:26.08 | ManxPower | And the spam is flowing |
21:26.33 | Qwell | ManxPower: where are your servers? |
21:26.48 | ManxPower | Qwell, Covington LA. |
21:26.55 | ManxPower | 30 miles north of downtown New Orleans |
21:26.57 | Qwell | pretty bad there? |
21:27.27 | ManxPower | Qwell, no. power is on, many trees down. One of the offices (Mandeville LA) has a tree thru part of it's roof |
21:27.44 | Qwell | ahh |
21:27.49 | ManxPower | I'm still in TX. But the MIS manager is at corporate office in Covington |
21:27.51 | Ariel_ | ManxPower, have you seen this:Sep 3 14:04:05 NOTICE[13585]: chan_zap.c:7977 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
21:28.00 | ManxPower | Ariel_, Yup. |
21:28.02 | Ariel_ | do you know of a quick fix? |
21:28.17 | Qwell | ManxPower: where you staying at? |
21:28.22 | ManxPower | Ariel_, Um, HDLC abort errors can be as hard to fix as echo. |
21:28.31 | Ariel_ | ahh |
21:28.33 | Ariel_ | it works |
21:28.35 | ManxPower | Qwell, 7 miles outside of Atlanta TX (30 miles from Texarkana) |
21:28.43 | Qwell | ManxPower: no, I mean, hotel or something? |
21:29.19 | *** part/#asterisk BVEBVE (n=bvebve@virgo.minbas.cu) |
21:29.39 | ManxPower | Staying with the BF's family |
21:30.00 | ManxPower | three of us, three cats, and a dog all evacuated. |
21:30.03 | Qwell | ahh |
21:30.28 | *** join/#asterisk tzafrir_ (n=tzafrir@DSL217-132-221-74.bb.netvision.net.il) |
21:31.58 | Ariel_ | ManxPower, glad your safe which is important |
21:32.16 | Ariel_ | ManxPower, so where should I start to trouble shoot and change settings |
21:32.22 | ManxPower | 90% of the city where I live(d) is reported to be totally destroyed. |
21:32.33 | ManxPower | Ariel_, What distro? |
21:32.48 | Ariel_ | fedora 4 using head |
21:33.15 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
21:33.24 | ManxPower | Ariel_, look in /etc/sysconfig/harddisks and enable all the options, then reboot. |
21:33.44 | Ariel_ | ok will do |
21:34.07 | *** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net) |
21:34.38 | ManxPower | Ariel_, HDLC abort means "got corrupted data from the card, usually from missed or delayed interrupts" |
21:35.10 | Ariel_ | great will start doing checks tonight when they are less active |
21:35.17 | Phoebers | w00t, I made my first connectivity with asterisk!!! hooray!!! |
21:36.03 | *** join/#asterisk tzafrir__ (n=tzafrir@CBL217-132-231-4.bb.netvision.net.il) |
21:36.35 | ManxPower | Ariel_, the occasional HDLC abort error is not usually a problem |
21:36.57 | Ariel_ | yes your right but they don't want to see them. |
21:37.12 | ManxPower | Ariel_, other common causes are IRQ shareing, evil built in ethernet controllers, and evil ide controllers |
21:37.48 | *** join/#asterisk kg (n=kg@chello062179062077.chello.pl) |
21:37.51 | Ariel_ | wow that is great info. I am writing it down. Tonight I am going to start to play with those settings. |
21:38.06 | fugitivo | you forgot evil sata controllers |
21:38.35 | ManxPower | fugitivo, Yeah, evil SATA controllers and evil RAID controllers |
21:38.54 | ManxPower | fugitivo, were you the one that got rid of HDLC errors by switching away from SATA? |
21:39.27 | fugitivo | no, but i experienced problems with sata, usb, and evil hardware on an evil motherboard |
21:40.12 | ManxPower | Ah |
21:40.20 | MikeJ__ | EVIL? |
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21:41.23 | MikeJ__ | hello ericm_06_ |
21:41.30 | MikeJ__ | welcome to the party |
21:41.42 | ericm_06_ | hello |
21:47.38 | *** join/#asterisk xai (n=pasta@cpe-70-112-17-10.austin.res.rr.com) |
21:48.16 | xai | manx: where are you? |
21:48.49 | xai | ManxPower: where you at now? |
21:49.07 | ManxPower | xai, texas |
21:49.23 | xai | ManxPower: i'm gonna dialog you.. |
21:49.52 | ManxPower | ok |
21:54.14 | DarthClue | ManxPower: have you seen the pictures of the waveland area? |
21:54.41 | ManxPower | DarthClue, I'm trying to avoid those pictures |
21:54.52 | MikeJ__ | noaa has some arial pictures |
21:55.01 | DarthClue | Manx: just to see if your place is still standing ... http://ngs.woc.noaa.gov/katrina/KATRINA0000.HTM |
21:55.13 | stepcut | argh, sipphone.com's new user sign seems to be busted -- I never get the email I am supposed to get :-( |
21:55.22 | ManxPower | DarthClue, I'm trying to avoid those pictures |
21:55.37 | ManxPower | But thank you for trying to insist I look at depressing pictures. |
21:55.38 | DarthClue | ManxPower: understood. |
21:56.12 | DarthClue | ManxPower: last i knew you weren't sure if it was still standing or not. |
21:57.35 | ManxPower | I don't want pics, I want to see in person |
21:59.16 | *** join/#asterisk iCEBrkr_ (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
21:59.38 | MikeJ__ | ManxPower, do you have any word yet? |
22:00.21 | ManxPower | MikeJ__, no |
22:00.38 | MikeJ__ | The last person I knew in the area got out of the french quarter yesterday and is on her way back to MI |
22:00.39 | ManxPower | My domain and servers should be back to full operation this evening, however. |
22:01.40 | MikeJ__ | ManxPower, where in waveland is your place? |
22:01.41 | ManxPower | my landlines should be forwarded soon as well |
22:01.52 | ManxPower | MikeJ__, Old Spanish trail and McLauren |
22:07.22 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
22:08.50 | *** join/#asterisk f_meehan (n=fmeehan@whoami8.cedval.org) |
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22:11.18 | MikeJ__ | ManxPower, I found an arial.. looks like the storm surge did not make it that far up |
22:11.30 | Borgon | whats the command to patch asterisk with a diff? "patch -p0 < blah.diff ? |
22:14.39 | ManxPower | MikeJ__, *nod* |
22:16.23 | blitzrage | Borgon: aye |
22:16.40 | DarthClue | ManxPower: Intersection of OST and McL, which direction from there? |
22:16.49 | blitzrage | McL? |
22:17.18 | DarthClue | Old Spanish Trail and McLauren |
22:17.42 | Borgon | blitzrage: thanks |
22:19.16 | ManxPower | DarthClue, at the corner |
22:19.24 | ManxPower | SE corner |
22:19.30 | Byte | is there a way to put arguments onto multiple lines? |
22:20.13 | DarthClue | right on the corner, with driveways on both OST and McL? looks like it is still there. |
22:20.42 | ManxPower | DarthClue, yup |
22:21.15 | DarthClue | looks like a few houses in that area got hit, but yours looks like it is still there. |
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22:25.14 | Byte | http://redrum.arlott.org/tmp/macro-call.txt <-- does that macro look ok? it seems to work but I don't know how many arguments I can add before it'd stop working right |
22:27.01 | *** join/#asterisk clive- (n=pirch@rrba-146-117-44.telkomadsl.co.za) |
22:29.48 | Phoebers | good night |
22:30.21 | *** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au) |
22:32.22 | clive- | does anyone know what ast_verbose would do in channel.c ? |
22:34.57 | *** join/#asterisk ManxPwr (n=eric@1Cust4089.an1.dfw28.da.uu.net) |
22:35.36 | clive- | Manx, hi do you know what ast_verbose would do in channel.c ? |
22:37.35 | *** join/#asterisk Borgon (i=Borgon@70-100-53-251.dsl1.tbr.ga.frontiernet.net) |
22:37.46 | bkw__ | Byte, wtf kind of macro is that |
22:37.47 | Borgon | After i apply an asterisk patch do i have to recompile and build from srcs again? |
22:38.16 | bkw__ | you can make that smaller |
22:38.25 | Byte | how? |
22:38.38 | bkw__ | all those args |
22:38.40 | bkw__ | how nasty |
22:38.44 | bkw__ | you should just pass it a number |
22:38.49 | bkw__ | and let the macro do all the dirty work |
22:38.51 | bkw__ | too many args |
22:40.21 | Borgon | bkw__: after applying RPID patches, is a recompile from /usr/src/asterisk required? |
22:40.28 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivflke.dialup.mindspring.com) |
22:40.28 | Qwell | Borgon: yes |
22:40.30 | bkw__ | Borgon, yes |
22:40.32 | Byte | so there's a simplier way to express: exten => _01.,1,Macro(call,e164,${UKCID},44${EXTEN:1},sip,${UKCID},${EXTEN}@sipgate2) ? |
22:40.35 | Borgon | ok |
22:40.47 | bkw__ | Byte, yes you seem to have some duplication of code there |
22:41.01 | bkw__ | the same thing is passed twice |
22:41.05 | bkw__ | in two differnt args |
22:41.06 | Byte | where? |
22:41.15 | bkw__ | ${UKCID} |
22:41.17 | Byte | the callerid could change depending on what is used to make the call |
22:41.26 | bkw__ | and ${EXTEN} should be passed once |
22:41.26 | Byte | so that's a bad example of why I'm passing the caller id |
22:41.48 | bkw__ | you can manage the differences in the macro |
22:41.58 | Byte | if I pass ${EXTEN} once, how am I supposed to tell it to remove 1 digit from the start and add 44? |
22:42.07 | Qwell | same way |
22:42.14 | bkw__ | you do that in your macro ${ARG1:1} |
22:42.20 | bkw__ | i'm just saying thats an ugly macro :P |
22:42.28 | Byte | but the macro doesn't know what to do :P |
22:42.31 | tzanger | bkw__: your asterlinke mail goes into no detail. what specific issues are you seeing with chan_iax2? |
22:42.50 | bkw__ | tzanger, outbound audio to the pstn will be clippy and choped to hell |
22:42.59 | bkw__ | i'm kinda shocked you haven't heard that |
22:43.18 | tzanger | bkw__: I haven't |
22:43.27 | tzanger | and every single call we take and make is to PRI |
22:43.41 | bkw__ | hrm |
22:43.41 | tzanger | bkw__: is this only under significant load? |
22:43.46 | bkw__ | not really |
22:43.48 | bkw__ | 20 calls |
22:44.01 | bkw__ | you'll have 10-15 seconds of audio just disappear at times |
22:44.18 | bkw__ | and as josh and i found out |
22:44.19 | Qwell | bkw__: Why haven't you fixed it yet? :p |
22:44.23 | bkw__ | it either happens really badly for you |
22:44.25 | bkw__ | or not at all |
22:44.28 | tzanger | bkw__: hmm |
22:44.29 | bkw__ | and when it doesn't happen WTF |
22:44.33 | tzanger | I have been seeing iax2 jb sync issues |
22:44.44 | bkw__ | like if a call is up |
22:44.47 | tzanger | but only on some iax2 connectiosn |
22:44.54 | bkw__ | and say a few registers hit the box |
22:45.04 | bkw__ | no audio for that instant when its doing the register |
22:45.06 | *** join/#asterisk anonymouz666 (n=lynx@allende.redetaho.com.br) |
22:45.09 | bkw__ | when it has 20 or so calls on the box |
22:45.29 | tzanger | bkw__: that's really weird |
22:45.35 | bkw__ | it is |
22:45.39 | bkw__ | I had to switch to sip |
22:45.44 | bkw__ | so people I called could actually hear me |
22:45.44 | tzanger | is this something that just started happenning? I.e. if you roll back 3 months does it still occur? |
22:45.51 | tzanger | bkw__: yeah I saw that in the email |
22:45.52 | bkw__ | I can't roll back |
22:46.07 | bkw__ | we already use things that are in the current CVS that didn't exist 3 months ago |
22:46.18 | *** join/#asterisk drbrown (n=chatzill@63.238.118.90) |
22:46.19 | bkw__ | its an iax2 issue |
22:46.19 | tzanger | bkw__: just for testing, I mean |
22:46.23 | bkw__ | because it doesn't happen on SIP |
22:46.30 | tzanger | bkw__: I understand that |
22:46.51 | tzanger | what I'm looking for is WHEN it started happennign on IAX2, to try and pinpoint what the change to chan_iax2.c was that does it |
22:46.56 | bkw__ | tzanger, well like I said some people it happens to.. some it doesn't and we can't nail down exactly what causes it.. I have checked timestamps |
22:46.59 | tzanger | there have been lots of changes to all of the channel drivers over the past little whiel |
22:47.00 | bkw__ | I have checked jitter buffer on and off |
22:47.18 | tzanger | bkw__: yeah bnut it does happen to you so you're in the best position to try and nail down when it started to happen |
22:47.18 | bkw__ | its really strange really |
22:47.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@CBL217-132-115-152.bb.netvision.net.il) |
22:47.28 | bkw__ | I have done all I can to try |
22:47.30 | Qwell | bkw__: really? |
22:47.33 | bkw__ | its the strangest thing |
22:47.39 | bkw__ | nothing makes sense |
22:47.43 | *** join/#asterisk CaNaBiS (i=canabis@ip68-111-51-215.br.no.cox.net) |
22:47.46 | bkw__ | i'll try again next week to get more time to check it out |
22:48.03 | tzanger | bkw__: yeah that's the shitty part... it usually happens to people with no additional time :-S |
22:48.13 | CaNaBiS | hey, I am in Baton Rouge, making calls has been difficult. Are there any providers like voicepulse or broadvoice that have a free trial period? |
22:48.17 | Qwell | tzanger: Murpheys Law |
22:48.18 | bkw__ | I think tuesday I'll have time |
22:48.36 | tzanger | has anyone heard from ManxPower since the hurricane? |
22:48.37 | bkw__ | CaNaBiS, who do you need to callo? |
22:48.40 | bkw__ | tzanger, yes |
22:48.42 | tzanger | he was in NO IIRC |
22:48.43 | Qwell | tzanger: yeah, he's in TX with family |
22:48.46 | bkw__ | ManxPower, is in here |
22:48.48 | tzanger | Qwell: oh good |
22:48.49 | CaNaBiS | bkw_, family in TN |
22:49.05 | bkw__ | CaNaBiS, ok you just need to make a few calls? |
22:49.10 | CaNaBiS | nawp |
22:49.18 | bkw__ | nawp? |
22:49.20 | tzanger | I mean I wasn't sure what I was gonna do if I couldn't tease him about STABLE. :-) |
22:49.21 | CaNaBiS | nope |
22:49.25 | bkw__ | how many calls? |
22:49.51 | CaNaBiS | I can make calls, theyre just scattered |
22:50.21 | CaNaBiS | and when I go to TN I need to be able to call back here which is the hardest |
22:50.59 | *** join/#asterisk santiago (n=santiago@63.245.86.163) |
22:51.06 | CaNaBiS | I thought that I had remembered seeing a provider that had a trial period |
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23:01.13 | *** part/#asterisk santiago (n=santiago@63.245.86.163) |
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23:07.10 | doughecka | anyone have one of these zyxel phones? |
23:08.11 | *** join/#asterisk comadreja (n=comadrej@pdpc/supporter/active/comadreja) |
23:08.31 | comadreja | hello, I'm getting a connection refused on an inbound call, why could that be ? |
23:08.52 | *** join/#asterisk borgasman (n=aa@bl5-39-66.dsl.telepac.pt) |
23:08.54 | borgasman | Every geeks dream clock muahahaha »» http://systemsecure.org/ssforum/viewtopic.php?t=162 |
23:11.03 | comadreja | that is a "Rejected connect attempt from..." |
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23:19.58 | PBXtech | does anyone make a pre-installed vicidialer config? |
23:20.20 | tzanger | not I no |
23:20.53 | tzanger | ha |
23:21.00 | tzanger | wha the hell is hour 13-15? |
23:21.27 | doughecka | '\? |
23:21.28 | tzanger | I have a binary clock at home |
23:24.08 | *** join/#asterisk liberie (n=liberie@unaffiliated/liberie) |
23:24.18 | smcmahon | Hour 13 - 15? That would be 1PM - 3PM ... |
23:24.22 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
23:25.07 | smcmahon | I need a binary clock that would be cool, I learned binary at the age of 3 yrs old lol |
23:27.26 | tzanger | smcmahon: so how doe sit work...3pm then go to 12 midnight, or does it skip those three hours or? |
23:28.01 | smcmahon | How does what work? tzanger... |
23:28.09 | *** part/#asterisk clive- (n=pirch@rrba-146-117-44.telkomadsl.co.za) |
23:29.00 | *** join/#asterisk smcmahon (n=admin@digitaldatabits.net) |
23:29.25 | ManxPower | ~docs |
23:29.27 | jbot | from memory, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
23:29.28 | ManxPower | ~mailinglist |
23:29.29 | jbot | well, 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/ |
23:29.42 | Qwell | smcmahon: thinkgeek.com, they have two clocks, and a watch or two |
23:30.58 | *** join/#asterisk tzafrir_laptop (n=tzafrir@DSL217-132-221-100.bb.netvision.net.il) |
23:31.17 | MikeJ__ | waving? |
23:31.52 | tzanger | ManxPower: hey! Glad ot see you're at least out of harm's way... how are things? |
23:32.05 | doughecka | ManxPower: hows it hangin? |
23:32.24 | ManxPower | tzanger, slightly better. main customer's corporate HQ is back up. |
23:32.31 | ManxPower | most of the frame relay network is working. |
23:32.42 | ManxPower | Power is on. The AT&T T-1s are down, however. |
23:33.00 | doughecka | where is it located? |
23:33.02 | tzanger | ManxPower: at least you'll have some reprieve from the real estate people :-) |
23:33.03 | ManxPower | doughecka, I'm alive |
23:33.08 | ManxPower | doughecka, Covington LA |
23:33.12 | doughecka | ah |
23:33.26 | ManxPower | the Mandeville LA office has a tree thru one part of the roof, but no damage to the NOC there. |
23:33.30 | doughecka | btw anyone use zyxel phones? |
23:33.40 | doughecka | ManxPower: and thats whats important |
23:34.08 | ManxPower | I need to write up a donations page for us. |
23:34.20 | smcmahon | The LED CLock is pretty neat.. Totally understand that one |
23:34.24 | ManxPower | I'm in decent shape, but the two others that evacuated with me are not in good shape moneywise. |
23:34.55 | ManxPower | they both work for the nola tourist office |
23:34.58 | Damin | "She touched my pepe, Steve." |
23:35.00 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
23:35.11 | smcmahon | LOL |
23:36.35 | Damin | Oh.. sorry about that.. |
23:36.40 | *** join/#asterisk clona (n=clona@c213-158-235-5.sdsl.no) |
23:38.08 | tzanger | which led clock |
23:38.23 | tzanger | Damin: hahahaha I forgot about that movie |
23:38.27 | tzanger | "hey teeve" |
23:38.33 | comadreja | I'm getting a rejected connect attempt from.... why could that be ? |
23:39.02 | comadreja | I'm quite noob, sorry :/ |
23:46.30 | *** join/#asterisk fugitivo (n=ajf@201.255.104.41) |
23:53.00 | *** join/#asterisk nemisus (n=nemisus@203-217-87-135.dyn.iinet.net.au) |
23:56.48 | Damin | tzangre: Yeah.. it showed up on my PVR the other day.. |
23:57.01 | Damin | tzanger: So I watched it... |
23:57.15 | tzanger | Damin: :-) |
23:58.32 | Damin | tzanger: I asle watched the Pilot episode of the 1987 Buck Rogers series w/ Twiki and Dr. Theopolis |
23:58.48 | JDLSpeedy | can someone help me add a privacy screener to asterisk? |
23:59.24 | tzanger | JDLSpeedy: what is a privacy screener |
23:59.31 | JDLSpeedy | I just found out how to record a name and play it back to the person |
23:59.56 | Damin | JDLSpeedy: Get a GrandStream BT100 and use Ringotne #2 on it. ;) "You have a call from 216-619-2000" |