00:00.11 | `paul | yup |
00:00.18 | `paul | modprobe ztdummy |
00:00.19 | jaytee | and you don't type /etc/init.d/zaptel start you type service zaptel start |
00:00.21 | `paul | returns |
00:00.36 | `paul | FATAL: Module ztdummy not found. |
00:00.56 | jaytee | the other command is for debian based systems. Red Hat based systems do it different |
00:02.22 | `paul | i remember installing it once.... when it failed it automatically loads ztdummy |
00:03.08 | jaytee | did you do a "chkconfig zaptel on" after you compiled and ran make install? |
00:04.25 | `paul | chkconfig zaptel on <--- doesnt return anything |
00:04.40 | jaytee | it shouldn't, it just runs |
00:05.08 | `paul | ok still same error when i start the service |
00:05.09 | `paul | Loading zaptel framework: FATAL: Module zaptel not found. |
00:05.12 | jaytee | type locate ztdummy |
00:07.16 | `paul | all in /usr/src/zaptel2/kernel/ztdummy.o |
00:07.23 | `paul | i mean /usr/src/zaptel2/kernel folder |
00:07.30 | `paul | and /usr/src/zaptel2 |
00:08.06 | jaytee | something isn't right with your compile/install |
00:08.32 | `paul | ill try to compile it again |
00:08.36 | jaytee | wait |
00:08.42 | `paul | yes? |
00:08.51 | beek | `paul: You did a "make install" as root? |
00:09.09 | `paul | beek: yup |
00:09.11 | jaytee | before you do, go into each of your source folders for zaptel and asterisk and do a make clean |
00:10.43 | `paul | ok im done ill compile zaptel first ok? |
00:11.07 | `paul | ./configure returns configure: *** Zaptel build successfully configured *** |
00:11.26 | `paul | i dont have zaptel hardware i just need it for the meetme application |
00:11.49 | jaytee | then follow this guide, it's worked for me every time except when I use Asterisk 1.4.22 with the latest zaptel. I'd recommend using Asterisk 1.4.21 and zaptel-1.4.10 or 1.4.11 |
00:11.52 | jaytee | http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
00:12.12 | `paul | im using 1.4.19 |
00:12.53 | jaytee | the older version of the tarball for zaptel can be found here: http://downloads.digium.com/pub/telephony/zaptel/releases/ |
00:12.54 | Talkradio | is there a setting in asterisk for how long the message has to be played before extension dialing works? |
00:13.20 | Talkradio | you gotta wait like 5 seconds or so right now before they work |
00:13.38 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
00:14.07 | jaytee | `paul, if you use that howto from the wiki after cleaning out your existing stuff in /usr/src/ for asterisk you should be fine |
00:14.47 | `paul | is libpri required for zaptel (ztdummy) |
00:14.49 | jaytee | Talkradio, what message? |
00:14.53 | Madkiss | hi all |
00:14.55 | jaytee | `paul, no |
00:15.02 | Talkradio | the ivr message |
00:15.17 | jaytee | `paul, libpri is only for pri circuits |
00:15.32 | Madkiss | i am having some iax-problems. I had a setup with one iax-connection that worked nicely. now I added iaxmodems on both sides, and I can't send faxes between the two sides of the iax-link because obviously, the remote-asterisk expects my side to authenticate as iaxmodem |
00:15.33 | jaytee | if you have a T1 card you'd need it |
00:15.45 | Talkradio | when you call in to main number and it starts it's greeting it's like 5 seconds before you can dial an extension |
00:17.03 | jaytee | Talkradio, there may be a Wait statement in your IVR code. Could be any number of things. set verbose 10 and pastebin a call attempt |
00:17.18 | Talkradio | ok |
00:17.19 | jaytee | and pastebin the extensions.conf |
00:17.23 | Madkiss | apparently, the iax-driver tries to connect the incoming connection to one of the hylafax-contexts. which is bullshit, of course. |
00:17.35 | Talkradio | let me remote in brb |
00:18.31 | `paul | arrrggghhhh |
00:18.33 | `paul | Loading zaptel framework: FATAL: Module zaptel not found. |
00:18.33 | `paul | <PROTECTED> |
00:18.33 | `paul | Waiting for zap to come online... |
00:19.17 | `paul | Error: missing /dev/zap! |
00:20.44 | jaytee | `paul, cd into your /usr/src/zaptel folder |
00:20.49 | jaytee | type make clean |
00:21.07 | `paul | ok |
00:21.20 | `paul | done |
00:21.37 | jaytee | `paul, just thought of something. Is SELINUX disabled? |
00:21.52 | `paul | wahts that? how can i know? |
00:22.10 | beek | `paul: getenforce |
00:22.52 | `paul | getenforce .... Disabled |
00:23.08 | beek | `paul: That's not the problem then. |
00:23.16 | jaytee | ok, that takes care of that question |
00:23.34 | beek | `paul: make clean && make install and then pastebin all of the output |
00:23.42 | jaytee | wait!!!! |
00:23.44 | joat | paul, did you recently upgrade the kernel? |
00:23.47 | `paul | is there any other solution for conference... |
00:24.04 | joat | what beek said |
00:24.15 | `paul | i got kernel-devel a while ago |
00:24.28 | jaytee | he should run a make menuselect in zaptel, disable the unnecessary drivers you don't need but you'll need ztdummy and ztdynamic |
00:24.31 | `paul | yum install kernel-devel |
00:24.34 | joat | wouldn't have mattered unless you use a new kernel |
00:24.48 | joat | then you need to remake zaptel (or dahdi) |
00:24.56 | `paul | ok ill go |
00:24.58 | `paul | make clean |
00:25.01 | `paul | ./configure |
00:25.05 | `paul | make menuselect |
00:25.18 | `paul | then select ztdummy and ztdynamic right? |
00:25.48 | joat | don't remember if it's that granular... make zaptel? |
00:25.58 | jaytee | yes, and if you want you can comment out any of the other modules like torisa, wctdm, wcfxo etc. |
00:26.00 | joat | how old is that version? |
00:26.30 | `paul | which one joat? |
00:26.38 | jaytee | joat, he's got asterisk 1.4.19 but I don't know which zaptel version |
00:26.52 | joat | for conference? ztdummy if you don't have hardware |
00:27.19 | joat | s/conference/meetme/ |
00:27.24 | jaytee | which fails when he tries to start the zaptel service. it can't find the modules because they haven't been compiled |
00:27.43 | joat | odd... |
00:27.46 | `paul | ok ill paste make output |
00:27.55 | jaytee | when I had him do a locate it could only find the object files in the source folders |
00:28.09 | Madkiss | as soon as i remove the iaxfax-context-stuff, it's working nicely again |
00:28.26 | `paul | http://pastebin.com/m2be759ae |
00:28.31 | Madkiss | can anyone point me to the right direction? i am out of ideas |
00:28.32 | `paul | tehre |
00:29.18 | joat | you did "make install"? |
00:29.30 | `paul | yup done with make install |
00:29.37 | joat | to ask the obvious (sorry) |
00:29.49 | joat | lsmod|grep zt |
00:29.55 | jaytee | make config |
00:30.21 | `paul | lsmod grep doesnt return anything |
00:30.31 | `paul | i did make config |
00:30.31 | jaytee | because it hasn't been loaded yet |
00:30.52 | jaytee | ok, now do chkconfig zaptel on and then service zaptel start |
00:30.57 | `paul | did you see the paste bin? |
00:31.01 | jaytee | yes |
00:31.03 | `paul | anything wrong ther? |
00:31.09 | jaytee | not that i can see |
00:31.15 | joat | didn't error out |
00:31.37 | `paul | Loading zaptel framework: FATAL: Module zaptel not found. |
00:31.39 | beek | `paul: I don't see any sign of the "make install" in that pastebin |
00:32.00 | `paul | i did a make> output.txt |
00:32.13 | `paul | so thats only make |
00:32.16 | jaytee | yeah, I didn't see that either but he just said he was pasting the make |
00:32.18 | `paul | i did make install after |
00:32.27 | beek | `paul: *THAT* was the output I wanted to see. |
00:32.27 | jaytee | any errors during make install? |
00:32.36 | `paul | ok wait |
00:32.42 | `paul | ill paste make install |
00:32.45 | joat | does "modprobe ztdummy" error? |
00:33.15 | jaytee | if he does a make install he shouldn't need to modprobe the module, it will do it for him |
00:33.36 | `paul | http://pastebin.com/m389e20a |
00:33.47 | joat | true he's received a module not found error then reinstalled |
00:33.48 | `paul | make install > output.file |
00:34.27 | `paul | i mean ive already done this on another server why wouldnt it work on this one |
00:34.43 | beek | `paul: do a uname -a |
00:34.56 | beek | What's the output |
00:35.54 | beek | `paul ? |
00:36.01 | `paul | 2.6.18-8.el5 #1 SMP Thu Mar 15 19:57:35 EDT 2007 i686 i686 i386 GNU/Linux |
00:36.07 | `paul | XXXXXXXX.com 2.6.18-8.el5 #1 SMP Thu Mar 15 19:57:35 EDT 2007 i686 i686 i386 GNU/Linux |
00:36.40 | jaytee | is this CentOS 5.1? |
00:36.50 | joat | since you've run "make install", try "lsmod|grep zt" |
00:36.51 | `paul | how do i check? |
00:37.08 | beek | 2.6.18-92.1.22.el5.centos.plus-i686 |
00:37.26 | beek | `paul: reboot your computer to get the correct kernel loaded. |
00:37.38 | `paul | after reboot |
00:37.48 | `paul | ill just do nothing? |
00:38.18 | `paul | well i cant reboot the server right now >.< |
00:38.26 | beek | You're mismatched with kernel-devel and what you're running. Reboot your computer and go through the steps again. |
00:38.41 | `paul | ok |
00:38.53 | jaytee | beek, good catch!!! |
00:39.04 | `paul | maybe ill try that later and be back here tom same time..... |
00:39.24 | beek | `paul: It'll take just a couple of minutes... aren't you curious? |
00:39.42 | `paul | argh |
00:39.49 | `paul | its being used right now |
00:40.04 | beek | `paul: I suspect that will take care of your problem. |
00:40.06 | `paul | ill email you of the result if you like |
00:40.08 | joat | passes a plaque to beek |
00:40.09 | `paul | :p |
00:40.17 | jaytee | it's odd that he'd have the kernel-devel for a newer kernel but still loading an older one, don't ya think? |
00:40.30 | `paul | cause a while ago |
00:40.38 | Kobaz | bluh |
00:40.39 | `paul | its looking for the source right |
00:40.44 | Kobaz | these avaya boxes are kinda flakey |
00:40.55 | Kobaz | second tone clock fried in like 6 months |
00:40.56 | `paul | kernel source |
00:41.00 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
00:41.09 | `paul | so i did an yum install kernel-devel |
00:41.28 | beek | `paul: just ensure that you do a yum update so that you have the newest kernel & kernel-devel packages. Reboot and go through the aforementioned steps. It will work as advertised then. |
00:41.53 | beek | thanks joat for the plaque |
00:42.19 | *** join/#asterisk Zippoman (n=bobperry@cpe-76-95-113-203.socal.res.rr.com) |
00:42.24 | `paul | im doing yum update right now |
00:42.27 | `paul | thanks guys |
00:42.31 | `paul | ill reboot it later |
00:42.55 | joat | Kobaz: don't like the sound of that... we just installed one at work |
00:43.47 | beek | `paul: while you're downloading I think that you should take jaytee's advice regarding the versions. |
00:43.59 | Kobaz | joat: why not an asterisk box? |
00:44.14 | `paul | what bout versions? |
00:45.10 | joat | Kobaz: wasn't our choice, corporate headquarters overrode our asterisk proposal |
00:45.14 | jaytee | if you're using asterisk 1.4.19 I'd recommend using zaptel 1.4.10 or 1.4.11 |
00:45.18 | Kobaz | those punks |
00:45.35 | joat | heh |
00:45.36 | Kobaz | joat: this is like a 15 year old box, perhaps the new ones are better |
00:45.47 | jaytee | I'm using zaptel 1.4.10 with asterisk 1.4.21 and it works fine |
00:45.50 | joat | okay, i'm less nervouse |
00:45.58 | joat | s/nervouse/nervous/ |
00:46.02 | Kobaz | but the tone clock was new |
00:46.08 | joat | can't type worth a dang tonight |
00:46.20 | Kobaz | they seem to be about 5 bucks on ebay |
00:46.21 | joat | gremlins then |
00:46.28 | joat | eatin' the cards |
00:46.56 | Kobaz | http://cgi.ebay.com/Avaya-Lucent-Definity-TN2182B-Tone-Clock-V1-card_W0QQitemZ200289846818QQcmdZViewItem |
00:48.37 | joat | that go in a giant blue cabinet? |
00:49.05 | joat | or was that the infinity (it's been awhile) |
00:49.42 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-22db24a2d8f7d4b2) |
00:50.19 | Kobaz | grey |
00:52.16 | Madkiss | jesus. with a dialstring like IAX2/FOOBAR, why does iax assume that the user who wants to connect is JOHNDOE just because there is a JOHNDOE context in iax.conf as well? |
00:52.58 | jaytee | naval grey like Nortel? |
00:55.04 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
00:55.37 | *** join/#asterisk telnettech (n=telnette@199.2.112.71) |
00:56.07 | telnettech | ok who in here is in a European country and can help me get my zaptel config right? |
00:56.18 | jaytee | I remember when I was 5 thinking "when I grow up I wanna be a firetruck" but then later on I let family and friends influence my career choices and look at me now. stuck in IT working on VOIP crap for peanuts. |
00:56.43 | jaytee | telnettech, are ya using CCS or CAS? |
00:56.49 | telnettech | CCS |
00:57.03 | jaytee | and HDB3? |
00:57.08 | telnettech | yes |
00:57.23 | telnettech | cause of the zaptel problem, my sound files dont play |
00:57.58 | jaytee | and this is the ISDN30 link? |
00:58.06 | telnettech | i can do a noload =>chan_zap.so in the modules.conf file and then start asterisk without starting zaptel and they play fine |
00:58.10 | telnettech | yes |
00:58.33 | jaytee | wow, that's a weird one |
00:58.47 | jaytee | when the link is up can you make calls? |
00:59.24 | telnettech | well i dont know yet cause the Aruban Telco hasnt installed my ISDN30 yet |
01:00.42 | telnettech | ;: |
01:00.56 | jaytee | so the server is in Aruba? |
01:02.48 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-221-124.phlapa.east.verizon.net) |
01:02.57 | telnettech | yes with me |
01:04.41 | Madkiss | where is [TK]D-Fender if you need him? *sigh* |
01:06.05 | jaytee | you're back in Aruba? geez, nice gig! |
01:06.47 | Katty | seanbright: :< |
01:06.49 | Katty | hugs seanbright |
01:10.10 | jaytee | telnettech, when you've got zaptel loaded and you try to play sound files do you get any console errors? |
01:10.30 | telnettech | no errors.....it shows that the sound file is playing |
01:11.18 | jaytee | same test phone, same file each time? |
01:11.20 | telnettech | it wont play mulitple sond files, ie when you are checking voicemail messages, it doesnt play all the of the menu option files |
01:11.25 | telnettech | yes |
01:11.43 | telnettech | i even tried to use my xlite to play and i cant hear anything on laptop |
01:12.20 | jaytee | but if you unload zaptel it all works fine? |
01:12.30 | telnettech | yes |
01:12.47 | jaytee | asterisk and zaptel versions? |
01:12.50 | telnettech | if i stop zaptel and tell asterisk to not load the chan_zap.so it works fine |
01:13.21 | telnettech | asterisk is 1.2.28 and zaptel is 1.2.21 |
01:13.46 | jaytee | have you upgraded zaptel after installing asterisk? |
01:13.54 | beek | GN guys |
01:13.58 | jaytee | nite beek |
01:14.05 | Zippoman | can someone help me out |
01:14.23 | telnettech | i tried to use the 1.2.27 zaptel but it doesnt have the zd_ethmf module that I need for the TDMoE |
01:15.11 | jaytee | makes sign of the cross and mutters, "fuckin TDMoE" |
01:16.44 | jaytee | telnettech, Redfone? |
01:16.50 | telnettech | we didnt have luck with internal cards when they started this and so this is what we have come with |
01:16.54 | telnettech | yes |
01:17.30 | jaytee | sorry man, I wouldn't touch one of those with ....... hmmmm......anything |
01:17.44 | telnettech | jaytee: that is where i got the zaptel and libpri tars from |
01:17.45 | jaytee | bet their tech support is awesome though! :-) |
01:18.12 | jaytee | so when you got those did you recompile asterisk? |
01:18.25 | jaytee | after compiling those tars? |
01:18.29 | telnettech | i loaded them from the start |
01:18.59 | jaytee | so you went libpri, zaptel, asterisk in that order? with no upgrades to any after the fact? |
01:19.06 | telnettech | before asterisk...i have reloaded this server 4 times since i startd this install in mid december |
01:19.08 | Katty | jayyyyyyyyyyyteeeeeeeeeeeeeeeeeeeeeeeeeaaaaaaaaa |
01:19.28 | jaytee | Katttttyyyyyyyyyyyyyyyyyy |
01:19.35 | telnettech | thats the order....i had to look at the workbook from class in the beginning to make sure i did it right |
01:19.38 | jaytee | thinks of the movie WALL-E :-) |
01:19.47 | Katty | haha |
01:19.54 | Katty | eeeeeeeeeeeeevvvvvvvvvvvaaaaa? |
01:20.01 | jaytee | heheehe |
01:20.15 | jaytee | I love the scene where he finds a spork and tries to add it to his collection |
01:20.19 | Katty | *blinkblink* |
01:20.22 | Katty | *plant on head* |
01:20.25 | Katty | eeevaaa? |
01:20.52 | jaytee | "Try blue! It's the new red!" |
01:20.56 | Katty | Ooooo |
01:21.52 | jaytee | ok, now i have to watch it again!!! |
01:22.18 | jaytee | just moved my Totem player over to my right hand 19" monitor. Nice to have dualies |
01:22.20 | Katty | i like the pizza plant comment at the end. |
01:22.47 | jaytee | My New Year's Resolution: 2880x900 FTW!!! |
01:22.53 | Katty | hehe |
01:23.08 | Katty | i want a chair like that |
01:23.40 | telnettech | cool....the singapore development crew just came online and they are lookng at it |
01:23.42 | *** join/#asterisk sack (n=sack@222.Red-83-49-103.dynamicIP.rima-tde.net) |
01:24.17 | drmessano | I have my team in Tunis looking at an issue for me right now |
01:24.24 | jaytee | lol |
01:24.32 | telnettech | where are you dr: |
01:24.50 | drmessano | Kuala Lumpur |
01:24.58 | telnettech | im in Aruba |
01:24.59 | jaytee | he's in Georgia being his usual wiseass self |
01:25.07 | telnettech | it is tuesday night |
01:25.11 | jaytee | which is why we all love him so much |
01:26.27 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
01:26.28 | carrar | Entrapment was a great movie |
01:26.35 | *** join/#asterisk Maliuta_CA (n=biteme@59.167.238.153) |
01:26.57 | jaytee | yeah |
01:27.06 | carrar | (speaking of Kuala Lumpur) |
01:27.07 | carrar | heh |
01:27.10 | jaytee | anything with Catherine Zeta Jones is worth watching |
01:27.15 | carrar | yeah |
01:27.25 | telnettech | yeah it was but my favorite is the Bourne trilogy for action |
01:27.38 | jaytee | yeah, I got those on DVD |
01:27.46 | drmessano | jaytee: Guess what.. my wish is coming true |
01:27.53 | drmessano | Australia's use of coal and carbon emissions policies are guaranteeing the âdestruction of much of the life on the planetâ, a leading NASA scientist has written in a letter to Barack Obama. In the letter he says: "Australia exports coal and sets atmospheric carbon dioxide goals so large as to guarantee destruction of much of the life on the planet" |
01:27.57 | telnettech | and all the Bond movies except the ones with Timothy Dalton |
01:28.13 | drmessano | War with Australia! |
01:28.24 | jaytee | telnettech, ugh! timothy dalton |
01:28.33 | jaytee | I was never a big fan of Roger Moore either |
01:28.33 | telnettech | he was terrible!!!!! |
01:29.00 | jaytee | I think Timothy Dalton and Steven Seagal went to the same acting school |
01:29.22 | telnettech | roger moore was ok not as good as sean himself but he was better than Timothy Dalton |
01:29.35 | jaytee | yes, I'll give him that! |
01:29.48 | telnettech | no steven seagal was in the CIA....didnt you know that ? |
01:30.03 | drmessano | Cant I Act? |
01:30.11 | jaytee | for real? I didn't know that. that explains the lousy acting |
01:30.41 | telnettech | watch TNT sometimes.....they like to have him do action movies for just their channel |
01:30.51 | telnettech | he is a fat guy like us |
01:31.26 | drmessano | Only Steven Segal would do a direct to video action movie |
01:31.31 | drmessano | Like Under Siege 5 |
01:31.48 | telnettech | there is one where he is in vietnamese pajamas |
01:31.53 | telnettech | what a laugher he is |
01:31.56 | jaytee | who you callin fat? :-) |
01:32.29 | drmessano | I always wanted what kind of person it takes to do a direct to video sequel of a hit movie.. Like Home Alone 4 |
01:32.36 | drmessano | Or Beethoven's 6th |
01:32.41 | drmessano | Then I look at Steven Segal |
01:32.46 | telnettech | well ok like me....im no spring chicken.....thats what my 13 yr old told me last week |
01:32.50 | drmessano | and I am like "Ah, I see what I did there" |
01:35.08 | drmessano | Someone needs to make a T.38 fax modem driver for Windows |
01:35.54 | telnettech | dr: are you running asterisk on a windows server? |
01:36.07 | drmessano | No |
01:36.40 | NovceGuru | http://upload.wikimedia.org/wikipedia/commons/9/9b/T.30_Protocol_Figure_01.jpg |
01:36.46 | NovceGuru | that looks like it was made in mspaint |
01:36.59 | drmessano | lol |
01:37.20 | drmessano | Um |
01:37.35 | drmessano | WOW |
01:37.41 | drmessano | Someone does make one.. $138 per channel |
01:37.59 | drmessano | Good god thats nuts |
01:38.33 | NovceGuru | who would bother buying it? |
01:38.47 | drmessano | For that price, no clue |
01:39.00 | jaytee | http://www.entertainmentearth.com/prodinfo.asp?number=AU11727 |
01:39.03 | NovceGuru | ya |
01:39.09 | drmessano | For under $40 I would |
01:41.28 | drmessano | I have an application that is going to be sending weather warnings to hearing impared folks with fax machines |
01:41.52 | drmessano | and right now my answer is Modem >> ATA >> Asterisk >> Provider |
01:42.20 | drmessano | Id like to cut the modem and ATA out of the mix and use software to emulate it |
01:42.36 | jaytee | hylafax? |
01:42.48 | drmessano | Has to be on Windows |
01:42.52 | jaytee | learning curve makes climbing K2 look easier |
01:45.11 | drmessano | Im at the point of rebuilding our project and removing points of failure |
01:45.28 | drmessano | and I see the modems and ATA as being a problem |
01:46.21 | drmessano | $189 for 2 channels <---- Bargain |
01:46.25 | jaytee | so you go from provider to asterisk to ATA to modem to the hearing impaired person's modem and computer? |
01:46.42 | jaytee | or their fax machine |
01:47.35 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
01:47.44 | drmessano | No, i'm sending a fax over the PSTN to their Fax machine on a PSTN line.. But in order to do it over VoIP, I need modems (which are external serial modems), and a T38 aware ATA |
01:47.52 | drmessano | Which is a cluster |
01:48.23 | jaytee | have you talked to coppice about this? |
01:48.35 | drmessano | Im sure he doesnt like Windows :) |
01:49.12 | jaytee | so you're using a Windows host to get the fax info and distribute it? |
01:49.34 | drmessano | The aggregator server application is windows based |
01:50.42 | drmessano | Without software, it needs to leave the box as an analog fax call |
01:50.43 | NovceGuru | drmessano: could you use a hylafax virtual printer on the windows based aggregator and hylafax ? |
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01:50.55 | drmessano | Hmmm |
01:52.21 | NovceGuru | there's http://winprinthylafax.sourceforge.net/ but I prefer http://whfc.uli-eckhardt.de/ as it lets you dynamically fill out the cover letter info |
01:52.25 | drmessano | I dont believe so.. It would need Windows faxing to believe it's a modem, as the server app ghooks into it |
01:52.28 | drmessano | -g |
01:52.35 | drmessano | Lemme look to be sure |
01:52.38 | NovceGuru | doh :\ |
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01:54.56 | drmessano | Looks like Windows Faxing or Winfax.. Which means it would need Hylafax support |
01:55.52 | NovceGuru | Winfax....what a piece of complete utter shit |
01:56.19 | drmessano | Indeed |
01:56.26 | drmessano | I knew someone who bought it once |
01:56.30 | drmessano | It was hooooorrid |
01:57.17 | *** join/#asterisk andresmujica (n=andresmu@190.27.1.233) |
01:57.21 | NovceGuru | me too! Fought with it for years and then just installed windows fax |
01:57.49 | drmessano | We have several businesses that live off windows fax.. One of which has a cluster of 6 modems |
01:57.52 | drmessano | and it works great |
01:58.44 | NovceGuru | Haha thats awesome. I know a guy that manages clusters of hylafax servers, I think they have > 1000 PSTN lines coming into their setup |
01:59.01 | NovceGuru | hmm thats scarry, last release of hylafax was dec 2007 |
02:00.10 | drmessano | Seems writing a T.38 modem emulator is more work that writing a printer emulator to hook into App X |
02:00.33 | drmessano | Theres a couple softphones that have T.38, create fax printers |
02:00.44 | NovceGuru | hm |
02:02.14 | telnettech | we use procomm plus |
02:02.42 | telnettech | the lady in the admin office uses it to fax all day any work orders and other stuff out of the company |
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02:04.19 | NovceGuru | I have a company I've been trying to get to use paperless faxing for years |
02:04.36 | NovceGuru | instead of printing and sticking in the fax machine...just print to the other damn printer in your list |
02:04.50 | telnettech | we also have a fax server that has adobe on it so incoming faxes are pdf for filing away |
02:04.54 | NovceGuru | and :O type in your cover sheet details instead of scratching in with a pen |
02:05.12 | NovceGuru | you don't need adobe to pdf faxes |
02:05.56 | telnettech | that is just what they have setup |
02:06.12 | NovceGuru | yeah, it's very nice either way :) |
02:06.39 | NovceGuru | especially with foxit readers annotation/typewriter feature, you can just type write on the fax and file print --> fax printer back to them |
02:06.50 | NovceGuru | right |
02:09.53 | drmessano | I found ONE commercial product |
02:10.03 | drmessano | Not even something to compare it to |
02:11.52 | f0urtyfive | anyone have a sip connectivity test? |
02:11.59 | f0urtyfive | or something that I can give a sip URI to ring? |
02:12.01 | drmessano | I swear this whole fax thing is so confining |
02:12.07 | f0urtyfive | like a little web app |
02:16.22 | drmessano | Ah well |
02:17.08 | NovceGuru | drmessano: yeah seems a little convoluted |
02:20.35 | telnettech | jaytee: you still in here |
02:21.09 | jaytee | telnettech, nope! I'm over there >>>>>>> |
02:21.26 | drmessano | OMFGGGGGGGGGG |
02:21.45 | drmessano | Brooktrout SR140, 2 channel Fax over IP solution for Windows |
02:21.45 | telnettech | just checking....must be a good movie or you are constipated |
02:21.49 | drmessano | $1800 |
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02:21.56 | telnettech | either way you have been quiet |
02:22.09 | jaytee | yeah, watchin WALL-E again |
02:22.13 | drmessano | $1800 to emulate a fax |
02:22.15 | jaytee | and eating some ramen |
02:22.28 | drmessano | jaytee: do you have a ferris wheel on your lawn? |
02:22.38 | jaytee | no |
02:22.39 | drmessano | I'm just sayin.. |
02:22.58 | jaytee | I'm missing a connection here? |
02:23.25 | drmessano | All the disney movies, the 1981 Chevy Caprice in mint condition, thick rimmed glasses.. Polyester pants... |
02:23.30 | drmessano | People talk |
02:23.35 | drmessano | Just sayin |
02:24.17 | telnettech | dr: you see that in a movie that americans have ferris wheels in their yards? |
02:24.48 | drmessano | What the heck are you talking about? |
02:25.02 | drmessano | I was implying something ELSE.. not that he's an AMERICAN |
02:25.36 | jaytee | drmessano, put the bong down and step away from the stash |
02:26.24 | jaytee | I wear rimless glasses, don't drive a 1981 Chevy Caprice and hate polyester and would never buy any clothing that has it, even a blend |
02:26.25 | drmessano | jaytee: Take the melted milk duds out of your pockets and hand over the Raggety Ann dolls |
02:27.16 | drmessano | Actually, telnettech was right.. I was just trying to prove you're an american |
02:27.31 | drmessano | By asking you if you had a ferris wheel on your lawn |
02:27.37 | drmessano | Ha, caught you.. |
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02:28.05 | telnettech | what movie did you see that in....i have got to find it and watch it |
02:28.18 | telnettech | that is so stereotypical |
02:28.51 | drmessano | Oh, I so know |
02:30.04 | jaytee | what movie was that in? |
02:30.20 | drmessano | 1941? |
02:30.25 | telnettech | oh |
02:30.40 | telnettech | i remeber the scene now with john belushi flying the plane |
02:30.42 | telnettech | lol |
02:31.13 | drmessano | I just remember the uber dorky nerd guy from Wargames on the ferris wheel spinning down the dock |
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02:32.13 | jaytee | Matthew Broderick? or the guy that played Melvin? |
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02:32.31 | drmessano | Melvin |
02:33.01 | drmessano | The one fro the "Mr Potato Head" scene |
02:33.16 | drmessano | MR POTATO HEAD, MR POTATO HEAD, BACKDOORS ARE NOT SECRETS |
02:33.32 | drmessano | I think.... |
02:34.05 | jaytee | the one who's screaming Mr Potato head or the dweeb he's screaming at? |
02:34.19 | drmessano | No, referring to the dweeb |
02:34.30 | telnettech | was matthew broderick in the movie? |
02:34.36 | drmessano | Yes |
02:34.39 | drmessano | He was the star |
02:34.43 | drmessano | But you know |
02:34.44 | jaytee | Herbie Kazlminsky |
02:34.52 | drmessano | Funny shit here.. we were talking about that today |
02:35.02 | jaytee | in real life Eddie Deezen |
02:35.27 | drmessano | Did you notice that Norad, with all its government money, had the same voice synthesizer as David? |
02:36.16 | telnettech | no i dont normally talk with NORAD |
02:36.45 | drmessano | IM TALKING ABOUT IN THE MOVIE |
02:36.51 | drmessano | God, keep up |
02:37.04 | jaytee | Maury Chaykin played Jim, the screaming guy and also played Sgt Downs in Iron Eagle II and the Major in charge of Kevin Costner who would piss himself and ended up shooting himself in the head in Dances with Wolves. |
02:37.35 | drmessano | Good god |
02:37.35 | telnettech | ok so you are talking about wargames right DR |
02:37.54 | drmessano | telnettech: No, I am talking about my many visits to Cheyenne Mountain |
02:37.55 | telnettech | how is that related to kevin bacon jaytee....lol |
02:38.29 | telnettech | i never have been there but you are saying that the voices in real life Norad are the same as the computer, David, from wargames? |
02:38.43 | jaytee | and years later that same voice synthesizer, in spite of all the advancements like Cepstral and other speech engines, is still being used by Stephen Hawking. |
02:38.56 | drmessano | ROFL |
02:39.01 | `paul | ei jaytee |
02:39.04 | `paul | beek |
02:39.16 | drmessano | I would LOVE to meet stephen hawking... Just to get him to voice that |
02:39.17 | jaytee | I was at NORAD in the late seventies and we had no voice synth |
02:39.22 | `paul | it worked!!!!!!!! |
02:39.28 | `paul | remember me? |
02:39.40 | drmessano | "Hey man, can you ask me "Shall we play a game". Please?" |
02:40.23 | jaytee | `paul, yeah we remember! so you switched to an older zaptel version and it worked? |
02:40.30 | `paul | nope |
02:40.35 | `paul | after the reboot |
02:40.45 | `paul | i compiled again |
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02:40.46 | jaytee | ah, so the kernel mismatch WAS the problem. |
02:40.50 | `paul | then it worked |
02:40.51 | `paul | yeah |
02:40.58 | `paul | thanks guys |
02:41.17 | drmessano | The kernel may be dead, but we're still enjoying his chicken |
02:41.40 | [TK]D-Fender | Road-kill Helper.... makes a great meal! |
02:41.41 | jaytee | weird that you'd be booting with an older kernel. You must have run yum install kernel-devel withouth doing a yum -y update first |
02:42.01 | `paul | yeah exaactly |
02:42.13 | drmessano | Stop with all the Linux talk.. I cant keep up |
02:42.22 | drmessano | Lets talk more about DLLs |
02:42.26 | `paul | lol |
02:42.28 | jaytee | drmessano, I recommend adderal or provigil |
02:42.45 | `paul | has the exploit for IE been fixed |
02:43.09 | `paul | ? |
02:43.11 | jaytee | yep, all the holes in Windows have been plugged (smirk) |
02:43.23 | drmessano | Sorry, jaytee, I wasn't paying attention |
02:43.25 | drmessano | What? |
02:43.29 | drmessano | Oh look, purple |
02:44.01 | drmessano | I like how every drug that makes you more sane has to make you fatter too |
02:44.05 | drmessano | My goal for 2009 ---> Crazy and thin |
02:44.19 | `paul | i downloaded the sample in wer a javascript and xml can make the calculator run |
02:44.19 | jaytee | if cheese was an operating system's security, Windows security would be Allsace Lorraine swiss |
02:44.41 | drmessano | jaytee: A straight person would have stuck with "Swiss" |
02:44.44 | drmessano | But ok......... |
02:45.19 | jaytee | don't forget big fella, I rollerblade! ;-) |
02:45.46 | drmessano | See that moves the needle in the other direction |
02:46.00 | drmessano | When I think "Not straight", I think of standard rollerskates |
02:47.00 | drmessano | I blame Hollywood, and years of my dad calling me ugly names because I wanted to be in the Ice Capades |
02:47.01 | drmessano | :( |
02:47.31 | *** part/#asterisk `paul (n=temp_acc@122.55.36.3) |
02:48.35 | drmessano | Ok, I got it |
02:48.53 | drmessano | I need a TDMoE modem emulator for windows |
02:50.35 | telnettech | dont use TDMoE.....that is my problem tonight |
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02:50.51 | telnettech | and my development team and figure it out either |
02:50.57 | telnettech | i am in big trouble |
02:51.38 | drmessano | I doubt there's a TDMoE anything for windows |
02:51.43 | drmessano | But I will take your advice |
02:52.52 | telnettech | jaytee: development guy is cussing refone...lol |
02:52.59 | drmessano | Thing is.. |
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02:53.32 | drmessano | TDM card with 2 FXS to cut the ATA out of the loop, still puts me in the $150 category |
02:53.44 | drmessano | For a cheap knockoff card, that is |
02:53.56 | drmessano | Still need the modems and cabling |
02:54.09 | drmessano | That $188 sounds kinda like a bargain lol |
02:55.18 | jaytee | sorry, I was roasting coffee |
02:55.30 | jaytee | telnettech, sounds like par for the course with Redfone |
02:55.36 | drmessano | French Coffee? |
02:55.43 | jaytee | TDMoE is sucky anyways |
02:55.50 | jaytee | French Coffee? no |
02:56.01 | jaytee | Kaui from Hawaii |
02:56.20 | telnettech | it looks like they are gonna stay with TDMoE cause they want to use thin rackmount servers |
02:56.33 | drmessano | I only drink Kopi Luwak |
02:57.27 | [TK]D-Fender | telnettech: ludicrous |
02:57.28 | jaytee | the pain comes when you have to debug issues on a TDMoE span. Not as straightforward as hardware. I like my T1 PRI on a digium card just fine |
02:58.05 | [TK]D-Fender | TDMoE = Retard's Choice... fresh brewed, just add water |
02:58.29 | telnettech | we havent had any problems until this install with them |
02:58.34 | telnettech | but if the zaptel is running, then my sound files play but i cant hear them |
02:58.48 | [TK]D-Fender | telnettech: Stupid dead-end tech that doesn't play with anything else. |
02:58.56 | telnettech | that is the only thing holding me back |
02:59.13 | jaytee | telnettech, [TK]D-Fender's right. and the rationale for using TDMoE is flawed. Even a Dell 1750 1u rackmount server has a pci slot in it. |
02:59.28 | telnettech | he is part of a bigger development team.......unfortunately he is caught in a numbes game for dev mgr |
02:59.32 | [TK]D-Fender | jaytee: more than 1.... |
03:00.11 | telnettech | i know....i do read a little even though i dont play alot with stuff....and like i said I have been dong tis for 4 months |
03:00.11 | [TK]D-Fender | jaytee: And thats before going with an external solution that ISN"T a flaming piece of shit... |
03:00.27 | telnettech | im learning networking, Linux, scripts, and asterisk on the fly |
03:03.04 | jaytee | [TK]D-Fender, there's a small riser card with two pci slots but I think the one on the right side of the chassis is only for a RAC card. |
03:03.24 | telnettech | ok guys....calling it a night....gotta get up early to be on a conference call with ops mgr and dev mgr about this site |
03:03.40 | jaytee | telnettech, nite guy. good luck!!! |
03:03.50 | NovceGuru | sounds fun, later |
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03:04.46 | jaytee | [TK]D-Fender, telnettech's a decent guy. Kind of got thrust into a role he wasn't prepared for though. He and I were in class together back in November. |
03:07.52 | jaytee | I'm just glad I got good advice here about staying away from Redfone when I was first investigating using Asterisk with PRI back in late 2006. The Asterisk consultants my boss brought in for a bull session tried to sell us one. |
03:08.34 | jaytee | everything I've read in forums or seen from people in here paint them out to be exactly the nightmare piece of junk that you and ManxPower said they were. |
03:09.07 | NovceGuru | I just picked up a TDM404EF, hopefully it was a good choice |
03:09.42 | drmessano | I think [TK]D-Fender is just jaded because he couldn't make TDMoE work |
03:09.49 | drmessano | and its been his arch nemesis |
03:10.06 | drmessano | Hence the TDMoExcitement |
03:13.58 | drmessano | Um |
03:14.07 | drmessano | lol j/k? |
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03:19.43 | NovceGuru | I think you pissed him off |
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03:23.02 | drmessano | I doubt it |
03:23.37 | drmessano | I think [TK]D-Fender is part italian.. He doesn't get quieter when you piss him off |
03:25.04 | russellb | drmessano: i thought it was funny, heh |
03:26.38 | drmessano | I'm sure TK is busy.. I've never not seen him go all drmessano about TDMoE.. |
03:27.56 | drmessano | [TK]D-Fender: Can you recommend a good Grandstream based TDMoE solution |
03:28.02 | drmessano | [TK]D-Fender: I am a bit stuck |
03:28.29 | NovceGuru | throw on the gasoline! |
03:30.24 | jaytee | a French Canadian from Quebec who's part Italian? Wow! talk about putting nitroglycerin in the blender and punching Puree! |
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03:34.37 | jaytee | oh, god! the environmental whackos are arguing about nuclear power and bashing hippies in #ubuntuforums. looks like a fun time's about to start |
03:34.42 | cabbey | are there any good docs for the end users of an asterisk system? All the docs I'm finding are for the administraitors of the pbx |
03:35.06 | jaytee | you need a manual to figure out how to dial a phone? |
03:35.32 | cabbey | trying to find one that handles simple stuff like "how do I call another extension?" |
03:35.44 | drmessano | Ok, I GOT IT |
03:36.11 | jaytee | ya pick up the handset and you punch the keys and if your admin did their job you shouldn't even need to press a Dial or Send key |
03:36.21 | NovceGuru | cabbey: wouldn't that partly depend on the phone? |
03:36.27 | drmessano | Windows Serial Modem Emulator >> Serial cable >> Linux Box running PPP >>> Out fax modem >> ATA >>> Asterisk >> ITSP |
03:36.36 | drmessano | 7 hops, nothing but net |
03:36.40 | NovceGuru | HOLY HELL |
03:36.45 | cabbey | well then, aparently our admin didn't :) |
03:36.47 | jaytee | drmessano, sounds like a win! :-) |
03:36.54 | drmessano | Im all over that shit |
03:36.58 | jaytee | cabbey, what kind of phone is it? |
03:37.09 | cabbey | polycom 330 |
03:37.13 | jaytee | Polycom? Grandstream? |
03:37.23 | jaytee | ah, got one right here. do you know someone's extension? |
03:37.28 | cabbey | yep |
03:37.40 | cabbey | 706 is the pbx admin :) |
03:37.52 | jaytee | pick up the handset, press the keys for the extension and press Dial in the upper left above the keypad |
03:38.02 | NovceGuru | should be able to dial 7xx and it go go go |
03:38.12 | NovceGuru | just sayin! |
03:38.23 | cabbey | this is our first attempt at a remote phone that's not on the same lan... |
03:38.42 | cabbey | heh, that's what I thought, but no go |
03:39.15 | jaytee | http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html |
03:39.16 | NovceGuru | haven't googled at all...does SIP traverse an openvpn...vpn without any trouble? |
03:39.53 | jaytee | there's a link for documentation on the 330. You'd need to know the SIP version to download the correct User guide but the 2.2 version will probably suffice. it's a pdf. |
03:40.16 | cabbey | oh hey, if I let it sit long enough it eventually gives me a fast busy signal three times and then goes back on hook |
03:40.17 | jaytee | ah, not on the same lan. NAT issues |
03:40.23 | cabbey | yeah |
03:40.33 | jaytee | or bad dialplan digitmap on the phone |
03:40.50 | NovceGuru | cabbey: I'd suggest a vpn |
03:41.05 | jaytee | i'd suggest a new admin :-) |
03:41.07 | NovceGuru | although I know nothing about your setup |
03:41.08 | cabbey | yeah, that's on the todo list :) |
03:41.10 | NovceGuru | that too, heh |
03:41.36 | cabbey | hmm... tcpdump on my router says a pile of traffic out on udp 5060... but nothing coming back |
03:42.05 | NovceGuru | might be an issue |
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03:45.57 | drmessano | Maybe theres a diode on the line |
03:46.02 | drmessano | Easy to remedy |
03:46.17 | drmessano | Unplug all equipment, and grab a frayed extension cord |
03:46.34 | NovceGuru | the one plugged into the wall? |
03:46.37 | drmessano | The kind that not only holds up the corner of the end table, but has been powering the lamp on it for 20 years |
03:47.00 | drmessano | Cut off the business end, shove it in the smartjack |
03:47.15 | drmessano | Then, plug the other end in a convenient wall outlet |
03:47.20 | drmessano | Should clear that diode right out |
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03:47.34 | cabbey | with a nice pop sound and some blue smoke to boot |
03:47.57 | drmessano | "blue smoke" <--- only if you've been a good boy this year |
03:48.13 | cabbey | heh |
03:48.41 | drmessano | May also be a firewall problem |
03:49.06 | drmessano | In which case, the criminal destruction of telco property *PROBABLY* didn't solve your issue |
03:49.26 | cabbey | well, I'm starting to think my problem here ain't at my end... thanks for confirming my sanity that dialing an extension really should have been that simple... I"m going to send these traces along and call it a night |
03:49.45 | drmessano | "aren't", not "ain't' |
03:50.02 | NovceGuru | atleast he used proper punctuation |
03:50.11 | drmessano | Aint that the dang truth |
03:50.12 | cabbey | heh |
03:50.31 | *** part/#asterisk cabbey (n=cabbey@24-159-193-106.static.roch.mn.charter.com) |
03:51.26 | drmessano | Oh god.. |
03:58.18 | jaytee | He's out at the moment, leave a message at the beep and He'll get back to you. BEEP |
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04:14.21 | x86 | wtf is astcanary? |
04:15.17 | x86 | jaytee: when my wife says "oh god" or "jesus christ", I simply reply "Yes?" |
04:15.43 | jaytee | astcanary is for asterisk realtime |
04:15.55 | x86 | what does it do specifically? |
04:16.03 | jaytee | as long as the canary is "singing" everything is good |
04:16.53 | x86 | realtime as in realtime configuration (SQL/LDAP-backed SIP peers, etc)? or realtime like realtime priority? |
04:17.10 | jaytee | I don't run realtime but as I understand it it's a process monitor. |
04:17.17 | jaytee | the first one |
04:18.02 | jaytee | it monitors the database connectivity |
04:18.19 | jaytee | might want to Google it for more info |
04:18.59 | jaytee | <PROTECTED> |
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04:31.34 | Nugget | I'll stop procrastinating tomorrow. |
04:32.15 | drmessano | I never cease to amaze myself at the utterly useless info I retain |
04:34.04 | drmessano | Sitting here plugging in NWS products.. and for some reason I picked them up intentionally.. |
04:54.30 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
05:07.21 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
05:10.08 | jaytee | [TK]D-Fender, what's up? connection issues? |
05:10.31 | [TK]D-Fender | jaytee: Firefox bouncing like a jack-hammer |
05:10.58 | jaytee | mine craps out after viewing too many YouTube vids |
05:11.27 | [TK]D-Fender | jaytee: Same here, and BSODs on flash quite a bit lately |
05:11.52 | [TK]D-Fender | jaytee: Which FUBAR's my watching The Daily Show and pisses me off to no end |
05:11.53 | jaytee | BSOD? Windows? your running Windows? ugh! |
05:12.16 | [TK]D-Fender | jaytee: yeah... I do have Ubuntu on this PC as well.. just can't quite kick the habit... |
05:12.59 | jaytee | yeah, tough habit to break |
05:13.22 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
05:17.59 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-6d67412bed415535) |
05:18.48 | drmessano | Hmmm |
05:19.30 | *** join/#asterisk sucituanbo (n=full@c-24-21-121-148.hsd1.wa.comcast.net) |
05:19.48 | *** join/#asterisk dandate (n=dandate2@c-71-202-125-220.hsd1.ca.comcast.net) |
05:20.29 | dandate | hey guys i am setting up an asterisk machine and was wondering if I could disable DHCP on my router for my cox cable modem and still have functionality |
05:22.06 | *** join/#asterisk hi365_m (n=hi365@bzq-79-182-160-246.red.bezeqint.net) |
05:22.38 | [TK]D-Fender | dandate: Networking is networking. If things point the right way they work |
05:23.02 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) |
05:23.42 | [TK]D-Fender | dandate: Which has nothing to do with * |
05:26.44 | drmessano | O.o |
05:27.06 | drmessano | If you disable DHCP, it will limit your ability to use DHCP |
05:28.41 | [TK]D-Fender | SMRT |
05:42.44 | drmessano | Hmmm |
05:47.08 | *** join/#asterisk reneger (n=reneger@dslb-088-078-123-090.pools.arcor-ip.net) |
05:47.42 | LemensTS | dandate: if you disable dhcp on the router you will just have to have all the devices under it as static ip's. |
05:53.44 | *** join/#asterisk jtodd (n=jtodd@188.sub-70-214-33.myvzw.com) |
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05:59.24 | [TK]D-Fender | W-T-F |
05:59.40 | [TK]D-Fender | turn my back and a random reboot... |
06:02.36 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
06:04.39 | jaytee | it's a feature!!! |
06:04.53 | jaytee | Vista's full of them! might wanna upgrade :-) |
06:05.49 | LemensTS | ive had vista for 3 weeks, id have to say xp is more stable. |
06:06.07 | jaytee | most definitely |
06:06.09 | LemensTS | vista isnt as bad as i thought tho. |
06:06.12 | jaytee | and less of a RAM hog |
06:06.25 | LemensTS | i cant tell this laptop is a haus it runs great |
06:06.35 | jaytee | not as bad as it's made out to be if you've got a really good graphics card and enough ram |
06:06.57 | LemensTS | yea i havent noticed slow down at all. i got 3gig tho. |
06:07.33 | jaytee | but Windows 7 will be WAY better, even on older hardware. MS learned it's lesson the hard way (yet again) and listened for a change. |
06:08.03 | jaytee | but in the next 4 years things will evolve much more quickly |
06:08.10 | LemensTS | reviews said it was slightly better than vista |
06:08.11 | [TK]D-Fender | Bringing in my X-10 gear tomorrow to hook my entryway buzzer to * |
06:08.38 | *** join/#asterisk Cherebrum (i=jgarland@209.9.237.93) |
06:08.50 | Cherebrum | Someone try this out and let me know if it works ok.... http://www.tollfreetollfree.com/ |
06:09.07 | jaytee | with the new i7 architecture all the old bottlenecks are gone and if MS codes to take advantage of the architecture's abilities then Win 8 or 9 will be incredible |
06:09.15 | LemensTS | does x10.com sell cameras or porn videos? |
06:09.45 | Cherebrum | x10 home automation stuff |
06:10.03 | LemensTS | lol look at their website |
06:10.09 | LemensTS | you'll see what i mean |
06:10.35 | Cherebrum | that site is UGLY with hot girls on it |
06:10.43 | jaytee | Cherebrum, all I get is a small login box in the upper left corner. What am I supposed to do with that? |
06:10.53 | drmessano | I need a windows app to monitor a parallel port for closures.. I bet its like $1200 |
06:11.07 | LemensTS | jaytee: login and make a call |
06:11.22 | LemensTS | it started asking me to use my camera, i decided to quit |
06:11.41 | jaytee | it doesn't even let me login with the creds supplied |
06:11.50 | LemensTS | hmm..it let me |
06:11.53 | LemensTS | ie7 |
06:12.07 | [TK]D-Fender | LemensTS: I always thought that was a lame joke... |
06:12.07 | Cherebrum | jaytee: click login |
06:12.31 | Cherebrum | jaytee: It doesn't seem to work in linux |
06:12.38 | Cherebrum | it works on windows and on my mac |
06:12.49 | Cherebrum | it's a flash based SIP softphone |
06:12.53 | drmessano | YEAH JAYTEE.. NOT THAT YOURE STOOPED OR ANYTING.. PUT IN INFO AND HIT LOGIN, NOT ALT-F4 |
06:13.01 | [TK]D-Fender | LemensTS: Sex sells, but every F-ING image... geez |
06:13.13 | LemensTS | i cant get half the sites to work in firefox/safari/iceweasal, etc |
06:13.14 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
06:14.16 | LemensTS | Cherebrum: is this your site? i can test it if its safe |
06:14.31 | jaytee | drmessano, I just signed up on the Atlanta Craigslist using your email and forged creds and placed an ad in Men looking for Men "Bottom seeks dom top to make me his bitch" |
06:14.32 | *** join/#asterisk asciii (n=WinNT@211.25.195.202) |
06:14.38 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-97b5e7cd676f852d) |
06:15.35 | jaytee | well, I got the phone pad to show up and a call went through but it didn't seem to hear me when I spoke my city and state. |
06:15.42 | jaytee | I don't have a webcam on this pc |
06:16.18 | drmessano | Jaytee: I dont cam |
06:16.28 | drmessano | jaytee: Dream on, rollergirl |
06:17.06 | jaytee | ok, enough of this bs for one nite. I'm going to bed |
06:17.16 | drmessano | LOL |
06:17.17 | drmessano | Nite |
06:17.21 | jaytee | see ya round, nite |
06:17.23 | drmessano | Now I want to listen to Skateaway |
06:17.50 | drmessano | Possibly the best Dire Straits song ever |
06:18.50 | *** join/#asterisk murdock_ut (n=chatzill@181.sub-97-153-209.myvzw.com) |
06:21.03 | Cherebrum | LemensTS: yes... it's my site |
06:21.26 | Cherebrum | LemensTS: I'm using this softphone: http://code.google.com/p/red5phone/ |
06:22.15 | Cherebrum | jaytee: you have to allow it to receive audio from your microphone |
06:22.28 | Cherebrum | jaytee: it pops up and says, "allow webcam/microphone?" |
06:22.42 | Cherebrum | oh.. he left |
06:25.13 | murdock_ut | I'm having a problem reparking a call that was parked using one step parking and picked up before it times out. Any ideas? |
06:25.27 | murdock_ut | Using 1.6 |
06:26.02 | drmessano | HA 4chan hijacked the macworld feed on macrumors.com |
06:26.15 | drmessano | "STEVE JOBS HAS JUST DIED" |
06:26.23 | drmessano | Thats hilarious |
06:26.40 | drmessano | I wonder how far Apples stock dropped over the 3 mins it took them to catch it |
06:26.51 | drmessano | Probably like 75% |
06:27.23 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
06:27.41 | drmessano | I feel for apple.. All those years of being an underdog, they start kicking ass in their consumer electronics offerings, and even now Steve Jobs coughs and the stock price plummets.. like 1987 |
06:28.09 | Cherebrum | BTW: You can also send tollfree calls via SIP to tollfreetollfree.com G729 or G711 are accepted. ;) |
06:29.17 | phpboy | I wish to goodness I can get this right |
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06:29.44 | phpboy | I need to speak to tzafrir, I think his script 'dahdi_genconf' doesn't work 100% correctly |
06:30.36 | joobie | phpboy, what's wrong with it? |
06:32.04 | Cherebrum | I'm going to bed |
06:32.05 | Cherebrum | nite |
06:38.24 | phpboy | joobie: It doesn't dedect my card |
06:38.35 | phpboy | however, if I configure it manually it works just fine |
06:38.43 | phpboy | in fact, it's still running just fine |
06:43.34 | dandate | is there a way to dig up irc chat history? someone here had told me of a few good providers that would charge 1 cent per minute outbound for use with my asterisk call center |
06:47.50 | dandate | and shit, how do i use voip with a hard phone? |
06:50.29 | murdock_ut | dandate: Use an adapter. |
06:55.51 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
06:55.54 | dandate | will my linksys pap2 work? |
06:56.23 | murdock_ut | murdock_ut: Yes, but I've seen them do weird stuff before... |
06:56.48 | murdock_ut | Mainly hearing dtmf tones when someone is talking. Little weird. |
06:56.58 | *** join/#asterisk SlicerDicer (n=SlicerDi@24-119-155-26.cpe.cableone.net) |
06:56.59 | drmessano | PAP2s? |
06:57.49 | murdock_ut | I think it was the pap2-na not pap2-t or vice versa |
06:57.57 | dandate | yes it is the pap2-na |
06:58.04 | dandate | will that work with a hard fone? |
06:58.09 | drmessano | Never heard of that problem.. and I have used dozens of them |
06:58.41 | drmessano | If youre hearing DTMF, then something is mixing with it.. and not the adapter |
06:59.04 | drmessano | or your 3rd neck is sitting on the keypad of your $6 walmart phone |
07:00.11 | phpboy | this is absolutely lovely :/ |
07:00.24 | phpboy | Hylafax doesn't wanna work anymore :T |
07:14.51 | dandate | how do i tell if a SIP provider is used for inbound or outbound? |
07:15.55 | drmessano | What do you mean "Tell"? |
07:23.59 | dandate | well i am trying to configure an asterisk machine for the first time =) |
07:24.14 | dandate | i am running an inbound call center www.newlineequity.com |
07:25.08 | drmessano | ok.. |
07:25.08 | dandate | so i need to beable to port my sales phone numbers to the machine, then have the machine either dial out the #s to my work at home employees hard phones, or connect directly to their softphone |
07:25.31 | drmessano | um ok |
07:26.03 | dandate | so i would need a SIP provider to ... |
07:26.06 | dandate | omg i'm confused |
07:26.40 | dandate | i know i have to set up a trunk outbound, can i use my comcast digital voice as my inbound? |
07:27.15 | drmessano | Sure |
07:27.28 | drmessano | You can pretty much do whatever you want |
07:27.35 | drmessano | You need to figure out what you want to do first |
07:27.52 | drmessano | How many Comcast lines do you have? |
07:28.08 | dandate | well i have 2 comcast lines that i advertise for |
07:28.31 | dandate | i need them both to come into the phone system, listen to the music, connect to the next available rep |
07:29.09 | drmessano | Ok, so you get a TDM card with 2 FXO ports, connect to the Comcast gateway |
07:29.29 | drmessano | Where does the SIP provider come in? |
07:29.36 | dandate | woah i have to google tdm and fxo |
07:30.01 | drmessano | ~book |
07:30.02 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
07:30.05 | drmessano | Start reading |
07:30.14 | drmessano | If youre this lost, you need to do some reading first |
07:30.28 | dandate | well i talked to an expert briefly on the subject, this is what he told me |
07:30.32 | dandate | [00:06] <C4thY> porting your numbers to an iax or sip provider ont he internet |
07:30.32 | dandate | [00:06] <C4thY> and then making it all true voip |
07:30.32 | dandate | [00:07] <C4thY> you dont need ata devices at all |
07:30.32 | dandate | [00:07] <C4thY> like bandwidth.com or teliax.com |
07:31.43 | drmessano | um ok |
07:32.38 | drmessano | I would suggest reading the book |
07:32.42 | drmessano | Make your own decisions |
07:33.39 | dandate | ok i have an FXO port, thats the linksys phone adapter right? |
07:34.14 | dandate | it says on it, "linksys phone adapter with 2 ports for VOIP" |
07:34.35 | dandate | the vender says that it was unlocked also |
07:34.47 | drmessano | No, the Linksys is an FXS |
07:34.49 | drmessano | Not an FXO |
07:35.11 | drmessano | It can connect to a provider and give you phone lines, like your Comcast box does |
07:35.21 | drmessano | But does not connect to existing service |
07:35.52 | dandate | do you mean the SIP providers like broadband.com? |
07:35.58 | drmessano | yeah |
07:36.50 | dandate | well i wanted to keep it as basic as possible, so because i have the comcast box i can skip inbound trunks right? |
07:37.06 | dandate | and just sign up for an outbound trunk like voipjet? |
07:37.25 | drmessano | Are you gonna use the Comcast for outbound and inbound or not? |
07:37.49 | dandate | hmm i'd like it to go both ways |
07:38.05 | drmessano | IF you keep the Comcast box you still need to connect those ports to the PBX.. which means a card with FXO ports |
07:38.26 | dandate | ok I see |
07:38.32 | dandate | can u recommend a product? |
07:38.47 | drmessano | Digium, Rhino, or Sangoma |
07:39.10 | drmessano | I dont have a specific recommendation |
07:40.24 | dandate | ok i see that would cost $83, I have unlimmited call forwarding on my comcast lines, so what if i forwarded them to an inbound sip provider instead? |
07:40.52 | drmessano | That would be stupid.. if youre gonna do that, then port the numbers over and ditch the COmcast |
07:41.13 | dandate | wellll it came in a package with the internet lol |
07:41.31 | drmessano | Its still costing you more |
07:41.47 | dandate | alright so when u say port the numbers over that means re-lease those numbers from a diff provider? |
07:41.54 | drmessano | yes |
07:42.10 | dandate | alright and that provider would be something like bandwidth.com or teliax? |
07:42.14 | drmessano | yes |
07:42.22 | dandate | any recommendations on those 2? |
07:42.23 | dandate | heh |
07:42.32 | drmessano | I use neither, so no |
07:42.49 | dandate | well i guess i should just jump in and try something |
07:43.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
07:44.30 | dandate | does anyone know anything about broadvoice? |
07:44.39 | The_Lightside | hi all, has anyone had issues with asteriskrealtime on queues with 300+agents |
07:45.48 | The_Lightside | in particular, the MOH stopping playing |
07:46.13 | dandate | omg i hit the jackbot! i only advertise in the state of california and broadvoice offers unlimmited inbound and outbound in state for 9.95 per month |
07:46.38 | mchou | dandate: how much total minutes of (comcast) phone you use a month? |
07:47.09 | dandate | ehh its hard to tell but i have 3-5 full time employees handling about 300 calls per day |
07:47.21 | mchou | dandate: whoa |
07:47.26 | mchou | lol |
07:47.41 | dandate | but the comcast is unlimited |
07:47.42 | drmessano | On TWO lines? |
07:47.55 | mchou | dandate: you a telemarketer? |
07:48.01 | dandate | yeah 2 lines, i have to rotate employees and its a bitch, thats why i got this vpbx so i can have as many reps on as needed |
07:48.08 | dandate | yes we direct market www.newlineequity.com |
07:48.15 | mchou | oh lord |
07:48.38 | mchou | astdb blacklist |
07:48.41 | dandate | people sign up because we show them how to take over a pre-foreclosure loan with no down payment or credit check |
07:48.48 | dandate | whats that mena? |
07:48.52 | mchou | lol |
07:49.16 | dandate | its possible, but u gotta be a go get em |
07:49.29 | dandate | our clients literally have to go knock on the door and buy the house for zero down payment |
07:49.44 | mchou | dandate: sont take it personally, I hate telemarketers |
07:49.53 | dandate | no we are inbound |
07:49.56 | dandate | people call us |
07:50.14 | mchou | dandate: anything is possible, but the real proof is how LIKELY |
07:50.32 | mchou | I may win the jackpot tomorrow.... |
07:50.33 | dandate | it all depends on the person i guess, we've seen a person get into a home within 3 days |
07:50.44 | mchou | but I'm not counting on it. |
07:50.54 | dandate | one out of 5 homes are facing foreclosure so if u went door to door in ur neighbor hood u might find a motivated seller |
07:51.08 | dandate | someone facing divorce could have too much house, can't sell cuz noone has a loan |
07:51.10 | mchou | indeedy |
07:51.36 | dandate | but most of our clients just get signed on and forget about it then get hit up for monthly billing |
07:51.44 | *** part/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
07:51.45 | dandate | now thats telemarketing! |
07:52.03 | drmessano | waits for someone to google the channel logs |
07:52.18 | mchou | drmessano: heh |
07:52.35 | mchou | real estate scams revealed (tm) |
07:52.59 | mchou | all those late nite infomercials |
07:53.12 | drmessano | http://www.myrealwebsite.com <--- BIG SCAM, SHHH DONT TELL THE INTARWEBS |
07:53.14 | dandate | actually the process is not a scam, my company provides everything they need, but is often too bogged down to help anyone further after registration |
07:53.25 | drmessano | keep digging |
07:53.49 | dandate | because when you take over payments on a loan you prevent the original person from having a foreclosure on their record |
07:54.22 | mchou | yes. hit that angle in my investments b4 |
07:54.40 | mchou | but there's also opportunity cost |
07:54.43 | dandate | and u can contract to provide future payment if necessary, which is some incentive to the owner than losing all equity |
07:54.58 | dandate | right u would have to clear up the default to stop the foreclosure |
07:55.02 | dandate | but we tell people its the same as renting |
07:55.04 | mchou | all the time knocking on doors cost real $$ |
07:55.23 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
07:55.32 | dandate | it could, but so could getting outbid at auctions |
07:55.44 | dandate | i mean noone said it was easy to buy a house zero down no credit check! |
07:55.51 | dandate | we just said the process was simple =P |
07:55.57 | mchou | lol |
07:56.00 | dandate | ok we did say it was easy |
07:56.02 | dandate | thats telemarketing |
07:56.22 | drmessano | Youre a regular Billy Mays |
07:56.29 | mchou | haha |
07:56.31 | dandate | shit i told someone its easier than renting and they bought it |
07:56.38 | drmessano | ITS EASY!!!!! |
07:56.44 | mchou | count on drmessano with the wonderful retort |
07:57.30 | drmessano | Just break off a small piece of foreclosure putty and squeeze to activate |
07:57.39 | drmessano | Then the magic begins! |
07:58.56 | dandate | thats how it has to be, because people facing foreclosure have disconnected phone lines |
07:59.08 | dandate | or if they are connected, they don't answer it cuz its prolly the bill collectors |
07:59.41 | dandate | then even worse, if the house is already vacant...u gotta send a letter to the forwarding address, also try to locate their family memmbers |
08:00.13 | dandate | but if u pull it off u could save up to 50% off a home |
08:00.23 | Coolthreads | anyone got any good tutorial on python for asterisk |
08:00.56 | dandate | i have a question, i need to have 5 people using my phone system, does that mean i need to purchase 5 seperate sip trunks / phone numbers from broadvoice.com? |
08:03.40 | rob0 | ask them? |
08:04.14 | rob0 | they don't care how many users you have, but they probably have a limit on concurrent calls |
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08:06.35 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
08:15.15 | dandate | i am trying to sign up for broadvoice but it wants the mac address of my pap2-na , how do i get this/ |
08:18.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
08:19.46 | dandate | do i just provide the mac address of my router? |
08:20.58 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
08:21.54 | mchou | dandate: best to ask them which one they want |
08:21.58 | dandate | haha my dumbass it was on the back of the model |
08:22.07 | mchou | most likely they want mac of your pap2 |
08:22.36 | mchou | broadvoice is lame to even ask for that |
08:22.41 | mchou | avoid |
08:23.12 | mchou | they are going to push some stupid ass configs your way |
08:23.33 | mchou | or do some other crap that wont be in your best interests |
08:27.27 | dandate | ok |
08:27.53 | dandate | well the tutorial i downloaded said to select "generic SIP" in which case they don't ask u for anything at all |
08:28.30 | dandate | but the pricing is really turning me on, i can get unlimmited inbound and outbound in state for just $9,95 a month |
08:28.57 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
08:30.30 | dandate | is there a better deal than that? |
08:30.43 | rob0 | Yeah, seems odd that they would need the MAC, maybe they just use it as a customer identifier. |
08:30.59 | dandate | well it said on the website they would create settings for me |
08:31.11 | dandate | and in the option for generic it said i would have to customize all the settings |
08:31.27 | drmessano | LOL |
08:31.39 | drmessano | They're gonna lock your device |
08:31.47 | drmessano | Good luck with that |
08:31.55 | dandate | oh shit |
08:32.08 | dandate | i gave them my mac address already, but i didn't fill out the credit card ap |
08:32.10 | drmessano | I thought you wanted to use ASTERISK |
08:32.21 | dandate | i do i was just confused cuz i have this linksys pap2 |
08:32.34 | drmessano | .... |
08:32.49 | drmessano | If they asked for your CAR KEYS would you get confused and MAIL THEM? |
08:33.05 | dandate | haha no but u think they might have locked my stuff ?? |
08:33.19 | drmessano | I dont see how, unless they have pixie dust |
08:33.55 | dandate | k |
08:34.00 | dandate | hela scarin me man |
08:34.02 | rob0 | How can they possibly lock an unlocked device? |
08:34.10 | drmessano | Easily |
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08:34.26 | drmessano | Pop in the provisioning info and turn provisioning on |
08:34.28 | drmessano | Done |
08:36.36 | drmessano | <Protect_IVR_FactoryReset ua="na">Yes</Protect_IVR_FactoryReset> <-- Thats how I roll |
08:37.48 | drmessano | Unlocked is the dumbest fucking term ever, IMO |
08:38.10 | drmessano | Theres several different things to consider with it being "locked" or "unlocked" |
08:38.49 | drmessano | First is whether or not there custom code in the firmware that ties it to a provider |
08:39.17 | drmessano | In the case of the Vonage PAP2s, you dont ever get rid of being a factory reset away from being a vonage box again |
08:39.46 | drmessano | Then theres the PAP2s that were never "locked" and not sold as Vonage devices |
08:40.08 | drmessano | Not "Unlocked".. but NOT TIED TO ANYONE TO BEGIN WITH |
08:40.47 | drmessano | Either one of those scenarios, an "unlocked" or unbranded PAP2 can be locked down with an XML config |
08:41.25 | drmessano | and once I block IVR factory reset and send it an encrypted XML, good luck using it again without me pushing a new config |
08:43.24 | drmessano | Actually, the encrypted XML is half true.. Once I put it on a firmware revision (3.1.7 or higher) that can be told to only look for an encrypted XML file, you cant spoof a config to it |
08:43.43 | drmessano | So yeah |
08:44.04 | drmessano | About as useless as that friggin "Brick" term that I can't stand |
08:44.34 | mchou | what do you mean? |
08:44.34 | drmessano | Makes me want to throw bricks at the next person that "bricks" something just so they can feel what it's really like to be "bricked" |
08:44.52 | mchou | 'my pap2 is now bricked' |
08:45.10 | drmessano | Well, in my little world, "Brick" means TOAST, DEAD, NON USABLE, DOORSTOP, THROW AWAY |
08:45.16 | mchou | i.e. wont boot |
08:46.22 | mchou | maybe I can 'unbrick' it using jtag or whatever |
08:46.35 | drmessano | But in circles where a product, any product, can have the firmware hacked or changed, and theres various states of "not being 100% usable right now", I hear constantly how someone "bricked" this or "bricked" that, but got it back working |
08:46.42 | drmessano | Really? How the fuck do you do that? |
08:47.06 | phpboy | Can I setup two IAX2 links to the same host, one with gsm only support and the other with slinear only support? |
08:47.14 | mchou | basically reinstall the boot loader, iirc |
08:47.22 | drmessano | "I loaded the new firmware and got stuck in diagnostic mode" or "firmware reload mode" is not as cool as "ZOMG BRICKED MAH ROUTER LULZ" |
08:47.24 | drmessano | But not true |
08:48.13 | drmessano | If you can make the device usable with a software reload of ANY KIND, you have NOT BRICKED IT |
08:48.31 | drmessano | If you throw it in the trash or use it as a centerpiece at the dinner table, its bricked |
08:48.32 | mchou | drmessano: yes, that is indeed true |
08:48.53 | drmessano | That crap pisses me off |
08:49.06 | drmessano | Lame ass Digg terminology |
08:49.29 | drmessano | I left my lights on last night.. went out to start my car, and found out I bricked it, LULZOMG |
08:49.37 | mchou | haha |
08:50.48 | drmessano | "I tried to get my next door neighbor drunk so I could poke her while her old man was out of town, but I gave her too much vodka and she passed out on my back porch." "Ha, you bricked her" |
08:50.52 | drmessano | Yep |
08:50.54 | drmessano | Guess so |
08:51.38 | mchou | that's what I call opportunity cost :) |
08:51.49 | drmessano | Damn right |
08:52.07 | drmessano | Piss-poor ROI |
08:52.15 | mchou | hehe |
08:52.48 | drmessano | Buy a date an $80 meal and she gives you a peck on the cheek.. ROI FAIL |
08:53.15 | mchou | that's prisoner's dilemma |
08:53.34 | mchou | do you byuy her another $80 meal? |
08:53.38 | mchou | buy* |
08:53.56 | mchou | and get 3x the return? :) |
08:54.35 | mchou | or get zip? |
08:54.42 | drmessano | Well, you can recalculate the ROI... If you buy her another $80 meal, and she was really saving up to give you $300 worth of Nookie if you made the investment in her, then you've got provable returns |
08:55.26 | mchou | yeah, but you dont know that in advance |
08:55.52 | drmessano | True.. Which I guess is technically required for a calculable ROI |
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08:56.06 | drmessano | Theres no expectation of a return |
08:56.13 | drmessano | So really, its more like a lottery |
08:56.17 | drmessano | lol |
08:56.21 | phpboy | I'm having a serious problem, I'm trying to send phone calls over one IAX2 link to 192.168.0.140 (gsm) and faxes over a second IAX2 link ysing (slinear) and it doesn't seem to wanna work :( |
08:56.38 | drmessano | sln? |
08:56.44 | phpboy | yes |
08:56.48 | drmessano | Fascinating |
08:57.00 | drmessano | Why are you doing that? |
08:57.21 | phpboy | I need to relay my E1 via another box for incomming calls (LONG story) |
08:57.26 | phpboy | it's a temp solutions |
08:57.30 | phpboy | *solution |
08:59.09 | drmessano | I didnt realize you could even use SLN over a peer |
09:02.08 | phpboy | Well, I can do it if I only run sln to the remote host |
09:02.16 | phpboy | but I can't seem to get sln and gsm working ;/ |
09:03.14 | dandate | does 5 people holding for a rep count as 5 concurrent calls? |
09:11.21 | phpboy | it seems I've learnt something new |
09:11.45 | phpboy | setup two diff IAX2 links from the src houst to ONE IAX2 link on the remote host |
09:11.54 | phpboy | to force 2 diff codecs |
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09:19.46 | phpboy | But the question is, does it actually work like this? ... |
09:20.36 | phpboy | FUCK! |
09:20.40 | phpboy | I was wrong :T |
09:22.04 | phpboy | oh wait |
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09:28.39 | dandate | the SIP trunk providers i'm looking at do not mention concurrent call limits. Does this mean that there are any? |
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09:33.08 | Assid | takes a number and waits for help on trying to get his cisco 7960 working |
09:33.28 | Assid | usiong sccp, can only get 1 way audio, other sip devices same location work fine |
09:34.19 | Assid | also why it doesnt retain the information uploaded to it via tftp is something i dont get |
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10:25.34 | wonderworld | we have a PRI and whenever something goes wrong at the other side (like "number has changed", "number out of service", etc.) asterisk returns the DIALSTATUS and hangs up. is there a way to make asterisk actually play those original messages from the telco side? thanks |
10:26.04 | BBHoss | wonderworld: to the user? |
10:26.08 | wonderworld | yes |
10:26.49 | BBHoss | not sure, but it gets that from the signalling of the pri, no audio is passed i don't think, when its just signalling |
10:28.13 | wonderworld | i already tried to set prisignaling to inband in zapata.conf, but i think that would affect incoming calls only, right? |
10:28.52 | BBHoss | i don't have much experience with pri in asterisk, just know a little about the underlying tech |
10:29.31 | BBHoss | your best best is to detect the fail and play your own message probably |
10:30.50 | BBHoss | wonderworld: what kind of card is it? |
10:30.52 | wonderworld | i am already doing this, but there aren't enough different DIALSTATUSes to really tell the user what was going on |
10:30.58 | wonderworld | digium te201p |
10:31.27 | wonderworld | for example the CHANUNAVAIL status can mean 3 different things |
10:31.33 | BBHoss | well digium should be able to help you, i would give them a call in a few hours, 4:30 here now |
10:31.49 | BBHoss | are there no channel vars for the pri info? |
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10:33.15 | wonderworld | i am not sure, i used DIALSTATUS till now. i'll have to read up. but there *should* be a way to pass the audio thru? when i call a "failed" number from my cell phone, i can hear the message.. |
10:33.44 | *** join/#asterisk orTix (i=blaat@j167028.upc-j.chello.nl) |
10:33.49 | orTix | hello, |
10:33.52 | BBHoss | thats probably the switch sending it to you via audio, there should be a way to do it, not sure |
10:35.00 | orTix | i got an astrisk server here with SIP on Cisco ip phone 7940 serie but i almost solve all the problems expect this one "W362 No valid names Provisioned" does anyone know howto solve this? I already googled it.. |
10:35.03 | BBHoss | wonderworld: priindication = passthrough maybe? |
10:35.11 | BBHoss | in zapata.conf |
10:35.17 | BBHoss | says you need bristuff though |
10:35.50 | BBHoss | orTix: cisco SIP firmware is trash, a possible solution is to buy a better phone for SIP |
10:36.03 | orTix | rofl |
10:36.09 | wonderworld | BBHoss: thanks. gonna read that up |
10:36.11 | BBHoss | seriosuly |
10:36.25 | BBHoss | wonderworld: says it doesnt work though so who knows |
10:37.25 | BBHoss | priindication = outofband or priindication = inband |
10:37.31 | BBHoss | inband is apparently default |
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10:39.23 | BBHoss | thats in chan_dahdi.conf wonderworld |
10:40.03 | ScriptFanix | Hi |
10:40.39 | BBHoss | hello ipv6'er |
10:42.06 | ScriptFanix | My SIP Phone (Siemens C470 IP - behind NAT) sometimes have troubles passing a call through my asterisk server (not on the same network, not natted) |
10:42.49 | orTix | BBHoss maby its shit, but i need it to get it working.. |
10:43.06 | BBHoss | orTix: yeah i understand, no idea here though |
10:43.11 | orTix | hmm ok |
10:43.21 | orTix | tnx iig :) |
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10:44.06 | ScriptFanix | I get this multiple times, then the call finally gets through: http://paste.quarantedeux.net/190 |
10:48.02 | ScriptFanix | not a blocking problem, but it's annoying ;) |
10:48.34 | wonderworld | hehe BAD! BAD! BAD! |
10:48.37 | wonderworld | funny message |
10:48.56 | ScriptFanix | :) |
10:49.38 | ScriptFanix | pictures asterisk banging his head on against the wall |
10:49.53 | orTix | :o |
10:50.03 | orTix | maby ScriptFanix can react on my question ;) |
10:50.22 | verywiseman | what is prefer ,fxo fxs ata or fxo fxs pci card? |
10:50.25 | BBHoss | ScriptFanix: probably the phone has a sip quirk |
10:50.51 | BBHoss | verywiseman: for small installs. atas are fine |
10:51.42 | verywiseman | can you suggest for my some brand which have good PCI card? |
10:51.58 | BBHoss | digium, openzap, sangoma, etc |
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10:55.55 | ScriptFanix | orTix: what question ? |
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11:01.31 | orTix | =:11:34:53:=- <orTix> i got an astrisk server here with SIP on Cisco ip phone 7940 serie but i almost solve all the problems expect this one "W362 No valid names Provisioned" does anyone know howto solve this? I already googled it.. |
11:01.32 | orTix | that |
11:02.19 | tzafrir_laptop | orkid, and you see that error where exactly? |
11:05.17 | orTix | w362 No valid line names provisioned |
11:05.34 | tzafrir_laptop | Where is it emmited? |
11:05.44 | tzafrir_laptop | coming from where? |
11:05.50 | orTix | On this cisco phone @ status |
11:05.57 | orTix | "status messages" |
11:06.14 | orTix | And big in the screen it says "phone unprovisioned (sip)" |
11:07.59 | tzafrir_laptop | orkid, oops, I meant orTix above. Sorry |
11:08.47 | orTix | I know :) np |
11:09.07 | tzafrir_laptop | orTix, while I have 0 experince with provisioning those phones, I suppose a pastebin of the provisioned data would help |
11:09.34 | tzafrir_laptop | (with passwords and other sensetive information masked out or whatever) |
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11:11.03 | orTix | Short way, you need the data of my cfg, asterisk -vvvvvr and telnet debug ? |
11:11.11 | orTix | and the tftp data? |
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12:03.08 | ITGuru | I've got one wierd problem - I have three devices, and they connect to asterisk, but they do not actually fully connect - The ring, but when you dial out, you get instant engaged tone. This is not the same as other handsets configured on the same machine, to do the same thing |
12:13.15 | Assid | ScriptFanix: you got it working ? |
12:13.41 | Assid | i got an issue where my cisco can call a sip phone but only able to send, cant receive audio |
12:15.10 | BBHoss | ITGuru: turn on debug and verbosity and see if you see anything interesting, else pastebin it |
12:15.35 | ITGuru | BBHoss, no probs - I'll jump on that right now |
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12:17.47 | espent | hi |
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12:18.26 | espent | any agi experts here |
12:18.29 | espent | ? |
12:19.04 | Assid | BBHoss: any idea on this? sccp audio reaches sip.. but not the other way around.. cant even get the cisco phone to g |
12:19.11 | Assid | to hear voicemailmain |
12:19.53 | BBHoss | Assid: the sccp driver is pretty shitty, there are patched ones floating around i think |
12:20.16 | BBHoss | try asking RypPn, i think he was talking about them the other day |
12:20.22 | Assid | i actually used some kind of patch |
12:20.39 | Assid | yeah he was around last night. but had toleave :( |
12:20.58 | Assid | i think im using : http://sourceforge.net/projects/chan-sccp-b/ |
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12:22.03 | ScriptFanix | Assid: it works, it just sometimes takes ~10seconds to get the call through... |
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12:22.17 | ScriptFanix | Assid: btw, i have no experience with Cisco phones |
12:23.40 | DarKnesS_WolF | if i have bindaddr=127.0.0.1 in sip.conf should i be able to register to any external sip proxy/server or not ? |
12:25.03 | espent | when i call Dial from my agi, how can I continue executing commands, i wont go further before the called up party hangs up |
12:25.53 | ScriptFanix | (the more i play with asterisk @home, the more i'm thinking about putting an axe through the 3Com NBX 100 here at work and setting up an asterisk box with appropriate cards) |
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12:26.29 | Assid | ScriptFanix: i spoke for more than 10 secs.. nothing |
12:28.15 | carranca | Hi, im trying to install a 3rd party app for video gateway purporses, (app_h324m) and it isnt working, ive debuged a little and now i want to see the information that is passing through the B-channel, using dahdi_monitor i can get a raw dump but isnt there a way to get a pcap dump to load it in wireshark like the old ZapDump? |
12:28.25 | ScriptFanix | Assid: I have no idea what your problem is, I'm quite new to telephony and thus, asterisk |
12:29.13 | Assid | hrmm k.. nvm then |
12:29.16 | Assid | will ask arnd |
12:41.56 | aiksa[LV] | is there a way to change zaptel timing source after it has been initialized? |
12:53.09 | *** join/#asterisk qdk (n=qdk@79.138.251.161.bredband.3.dk) |
12:57.03 | freddyk | i have a question about voice and fax detection on a line |
12:57.19 | freddyk | i have a fax connected to an ata on a sip peer |
12:57.43 | freddyk | i would like to hangup a call if no fax is detected in 15 seconds |
12:57.46 | freddyk | is this possible ? |
12:59.50 | tzafrir_laptop | aiksa[LV], you don't control it directly |
13:00.28 | tzafrir_laptop | but generally whenever the zaptel timing master span gets in alarm, there is "election" of a new master span |
13:00.52 | ITGuru | BBHoss, verbosity is 3, and core debug is 4 - i'm using asterisk -r at the CLI - but I'm a bit unsure what to look for there doesn't seem to be anything strange |
13:01.37 | BBHoss | turn sip debug on and try to make a call where it has problems, then pastebin it |
13:03.02 | ITGuru | BBHoss, darn, I'm remote - I'll ask someone there to do so |
13:03.04 | *** join/#asterisk mgdm_ (n=michael@serenity.mgdm.net) |
13:03.16 | BBHoss | ITGuru: you cant ssh in? |
13:03.22 | *** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
13:03.30 | andresmujica | Hi, good morning! |
13:03.38 | ITGuru | BBHoss - I am SSH'd in - you asked to make a call on the affected extension |
13:03.43 | BBHoss | ahh ok |
13:03.44 | BBHoss | gotcha |
13:03.54 | ITGuru | BBHoss, all the other extensions are working fine, just these three. |
13:04.03 | BBHoss | its when the endpoint attempts to make a call right |
13:04.07 | BBHoss | but they ring fine? |
13:04.10 | andresmujica | i wonder if anyone knows about a solution for PRI failover ? |
13:04.24 | andresmujica | i'm looking this http://www.beronet.com/content/view/83/91/lang,en/ |
13:04.41 | andresmujica | but it seems to need 220VAC and is not clear if it supports 60hz.... |
13:06.15 | ITGuru | BBHoss, That is correct, they ring, but no call goes out |
13:06.28 | BBHoss | what kind of phones are they |
13:06.29 | ITGuru | calls* |
13:06.55 | ITGuru | Mitel 5340 x 2 - and Polycom Sound Station |
13:07.01 | ITGuru | @ BBHoss |
13:07.23 | BBHoss | hmm, mitels aren't really know for standards compliance :) |
13:07.41 | carranca | has anyone configured asterisk with the app h324m? |
13:08.45 | *** join/#asterisk yam_ (i=yam@pdpc/supporter/active/yam) |
13:08.50 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
13:08.54 | *** join/#asterisk ipguy2 (n=ipguy2@124-170-238-129.dyn.iinet.net.au) |
13:09.02 | ipguy2 | hi all |
13:09.26 | ipguy2 | i just setup my first * server and i'm really excited |
13:09.32 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
13:10.59 | ipguy2 | currently only using it for one phone line at home with a sip phone |
13:11.10 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.69.17) |
13:11.15 | ipguy2 | + vioicemail |
13:11.24 | ipguy2 | thinking of setting up gtalk |
13:11.45 | orTix | what for phone are you using? |
13:12.10 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
13:12.18 | ipguy2 | it's a "GigasetC470 IP" |
13:12.33 | tzafrir_laptop | aiksa[LV], hmm... but why do you need this? |
13:12.36 | ipguy2 | nice phone, cheap, but a little slow |
13:12.49 | orTix | hmm oke |
13:13.26 | ipguy2 | i'm wondering what other people are using * for, like gtalk etc... any other cool things ? |
13:14.00 | ipguy2 | i'd like to read up about dialplans and extensions so i can create an answering service etc.... |
13:17.07 | freddyk | is there any way to hangup a call if it's not a fax even if it's outgoing or incoming |
13:17.07 | *** join/#asterisk styelz (n=yoohoo@2001:5c0:1100:a00:0:0:0:1) |
13:22.28 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
13:25.40 | *** join/#asterisk telnettech (n=telnette@199-2-116-183.setardsl.aw) |
13:26.10 | PanGoat | freddyk: you mean... just hang up a call? |
13:26.21 | PanGoat | hangup() |
13:26.36 | freddyk | pangoat |
13:27.14 | freddyk | pangoat: i have a line that i need to dedicate to fax but often, people in the office, uses it for personal calls. I just need to hangup if it's a voice call and not a fax |
13:27.45 | jks | anybody got an idea how to enable a snom 370 for configuration? - I can access the webinterface, but I'm not allowed to enter anything in the text fields to change SIP info for example |
13:31.08 | PanGoat | freddyk: Ahhh... I see |
13:32.25 | PanGoat | freddyk: can't you just query the extension that the channel is currently on? if it's the extension used for fax, don't hangup(). |
13:32.50 | PanGoat | http://www.voip-info.org/wiki-Asterisk+fax <-Asterisk as a fax/voice switch |
13:32.59 | PanGoat | towards the middle |
13:33.32 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
13:33.42 | PanGoat | I'm guessing though, being a * newbie |
13:35.21 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
13:36.25 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
13:43.29 | eppigy | hello |
13:43.32 | eppigy | i am dave |
13:44.40 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
13:44.44 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
13:45.37 | ipguy2 | open the pod bay doors dave ! |
13:45.54 | eppigy | :[ |
13:46.07 | ipguy2 | :-) |
13:48.09 | freckle | ipguy2: I use * from lots of stuff, what do you want to know? |
13:51.12 | *** join/#asterisk ta^3 (n=tacvbo@189.146.181.5) |
13:53.33 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.69.17) |
13:57.21 | *** join/#asterisk andresmujica (n=andresmu@correo.seaq.com.co) |
14:05.31 | x86 | I'm getting one-way audio on inbound calls only (from the PSTN)... outbound calls following the same path work fine |
14:06.03 | x86 | I've got a 4-port FXO digium card connecting asterisk to 4 POTS lines |
14:06.34 | x86 | any ideas what might cause that? |
14:07.33 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
14:07.36 | x86 | Linksys SPA941 SIP phones --> LAN --> Asterisk --> TDM400P --> PSTN |
14:07.51 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
14:08.23 | Mimmus | hi, I'm having a problem with mail notification of voicemail messages |
14:08.32 | Mimmus | It's not the first server I build :-( |
14:09.19 | Mimmus | I don't see any trace in /var/log/messages |
14:11.20 | Katty | hummm. |
14:11.40 | eppigy | TRABAJO |
14:11.50 | Katty | morning |
14:12.12 | eppigy | good morning |
14:12.24 | eppigy | it is a beautiful day |
14:12.46 | Katty | is it? i've not pried my eyes open yet. |
14:13.07 | Katty | i remember something about it being frosty |
14:13.47 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
14:13.53 | eppigy | i am in atl |
14:13.55 | coppice | what is the frost you speak of? do you mean there are places where the temperature drops to freezing? |
14:13.56 | eppigy | it is perfect |
14:13.58 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
14:14.06 | eppigy | I will be in portland next week |
14:14.12 | eppigy | so it will be a little chilly then |
14:14.14 | eppigy | :D |
14:15.31 | jaytee | portland maine? |
14:15.51 | *** join/#asterisk TSCDan (n=demord@cpe-69-207-149-216.rochester.res.rr.com) |
14:16.40 | jaytee | it's snowing here. |
14:16.43 | Carlos_PHX | Frost. I see some in my freezer. |
14:18.01 | [TK]D-Fender | 10" of snow coming down.... |
14:18.09 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
14:18.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
14:18.26 | beek | [TK]D-Fender: My world is ice-covered. |
14:18.30 | coppice | snow is for suckers who haven't figured out where the tropics are |
14:19.42 | TSCDan | We are going to be switching over to a VoIP system shortly (internally) that connects to our T1's for POTS lines (True VoIP in the future). We have a tech support dept that currently has several physical modems sitting in their cubes to dial out to customers. We want to have a modem pool for these users to connect to from their PC instead of physical devices, I was looking for some sort of virtual modem device for Windows machine |
14:19.47 | *** join/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu) |
14:21.33 | [TK]D-Fender | TSCDan: Who needs a modem to place a call? |
14:21.49 | eppigy | jantman: portland oregon |
14:22.05 | TSCDan | [TK]D-Fender: We have support personnel that need to dial into customer's machines. |
14:22.05 | eppigy | jaytee: portland oregon |
14:22.07 | jantman | eppigy: huh? |
14:22.12 | eppigy | jantman: leave me alone |
14:22.31 | jantman | "eppigy: jantman: portland oregon" ??? |
14:22.35 | eppigy | D: |
14:22.42 | eppigy | why u fakin logs? |
14:22.55 | jantman | *sigh* |
14:23.05 | [TK]D-Fender | TSCDan: Ok, this is rare stuff, best of luck with that and when you consider going "full VoIP" you can kiss that idea pretty much goodbye |
14:23.19 | jaytee | Portland, Oregon is nowhere near as "chilly" as Portland, Maine. I've been to both places many times. |
14:23.26 | jantman | anyone in here know of a good VoIP forum - something that's active and friendly to non-Asterisk-specific questions? |
14:24.15 | jaytee | I much prefer Portland, Oregon but then if I had my preference I'd prefer Ashland on the Cali border in the Rogue Valley. Lived there 5 years and loved it. |
14:24.26 | TSCDan | [TK]D-Fender: We will most likely be keeping several POTS lines just for them in the future, but still using the VoIP gateway from their PC's. |
14:24.43 | tzafrir_laptop | coppice, tropics is where the tropical storms come from ;-) |
14:25.57 | coppice | tzafrir_laptop: we strategically placed taiwan, and the philipinnes to take the worst of those |
14:26.05 | andresmujica | i wonder if anyone knows about a solution for PRI failover ? |
14:26.14 | andresmujica | i'm looking this http://www.beronet.com/content/view/83/91/lang,en/ |
14:26.25 | andresmujica | anyone has ever used it? |
14:30.46 | Linuturk | DigitalIrony: ping |
14:30.50 | x86 | on inbound calls from TDM400P to Linksys SPA942 phones, getting one-way audio (I can hear caller, caller can not hear me), asterisk CLI shows "chan_sip.c: Can not find address for host '7001'" |
14:31.16 | x86 | where 7001 would be my SIP peer |
14:32.01 | [TK]D-Fender | x86: Where are you configs? Where is the SIP debug? Where's Waldo? Where in the world is Carmen Sandiego? Who killed J.R.? |
14:32.03 | *** join/#asterisk silentaudience (i=psybnc@194.105.146.253) |
14:32.06 | andresmujica | nat enabled on the linksys? maybe the external address field is filled? |
14:35.57 | joesuffceren2 | >anyone know of a 64bit tsp (tapi) driver for asterisk? |
14:39.17 | jantman | does anyone in here know of a good VoIP forum - something that's active and friendly to non-Asterisk-specific questions? |
14:39.52 | ITGuru | I'm having a seriously crazy time with some MITEL handsets, they refuse to dial out, giving me an instant engaged tone, but they ring when in a ring group - any ideas? |
14:40.13 | [TK]D-Fender | jantman: Here might be good, depending on the question |
14:40.36 | [TK]D-Fender | ITGuru: Idea : Go look at your SIP DEBUG |
14:40.39 | *** join/#asterisk Shizuo (i=pato@200-171-49-211.dsl.telesp.net.br) |
14:40.45 | Shizuo | What is the cheapest card for Asterisk? |
14:41.01 | jantman | ok... i've got some experience with Asterisk, but only with internal (to the organization) VoIP or analog PSTN, i'm having some issues finding a service provider |
14:41.29 | angryuser | Shizuo: selfmade clone, a week of soldering 5$ |
14:41.37 | *** join/#asterisk awannabe (n=brad@ip98-167-184-247.ph.ph.cox.net) |
14:41.49 | Shizuo | angryuser: Clone of what? |
14:41.51 | BBHoss | Shizuo: an x100p clone |
14:42.12 | jantman | developing a prototype app for asterisk... just allows employees to call in, go through an IVR and make some selections, and dumps their responses in a database along with CID information... needs to handle about 4 simultaneous calls, and want VoIP service since I don't have multiple analog lines. |
14:42.14 | awannabe | hello all, how do you set outgoing callerID to be private/anonymous? I think you have to add a SIP header but I cannot find what to put. |
14:42.44 | angryuser | does x100p able to provide timing btw ? |
14:42.52 | BBHoss | yeah |
14:42.58 | Shizuo | Isn't that extremely hard? |
14:43.24 | angryuser | Shizuo: i was ironic, you can but a x100p clone |
14:43.30 | angryuser | *buy |
14:43.36 | jantman | should I be looking for IAX or SIP (will probably end up being IAX since I'll probably be on a dynamic IP)? Also, I can't find a definitive answer to the difference between trunk and termination service... what do I need if I want, say, 4 simultaneous calls to the same number (hunt group) and incoming only? |
14:44.07 | [TK]D-Fender | jantman: ... |
14:44.10 | [TK]D-Fender | ~itsplist-us |
14:44.11 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
14:44.13 | [TK]D-Fender | ^^^^^^^^^ |
14:44.23 | Shizuo | I can't seem to find prices, though |
14:44.25 | [TK]D-Fender | jantman: SIP recommended |
14:44.28 | Shizuo | I'll keep searching... |
14:44.30 | jantman | ok |
14:44.32 | jantman | thanks |
14:44.52 | jantman | what about trunks/termination and hunt groups (or whatever they're called in VoIP lingo)? |
14:45.05 | Shizuo | Is it hard to code your own drivers for Asterisk? Does this software contains a good driver API? Or is it criptic like Linux? |
14:45.08 | jantman | essentially I need one number that people call, which can handle four simultaneous calls |
14:45.22 | jantman | in the old world, it would be four lines or a PRI with a hunt group |
14:45.24 | [TK]D-Fender | ITGuru: And no SIP debug does not mean they haven't registered. You can prove that separately.... and SHOULD |
14:45.40 | ITGuru | Thanks for the response - I already did a SIP debug, and no joy, didn't spit out a thing, as if the handsets had not registered at all |
14:46.01 | NovceGuru | oh FFS time warner business phone that i'm being forced to use does not offer sip trunking, only POTS out of their shitty ass modem |
14:46.14 | [TK]D-Fender | jantman: the idea of "hunt group" is an analog concept for otherwise unassociated lines. |
14:46.33 | BBHoss | NovceGuru: just dump the firmware and get the sip credentials |
14:46.36 | [TK]D-Fender | jantman: Anywhere else its just "I want 1 DID supporting X channels" |
14:46.45 | angryuser | Shizuo: ask tzafrir |
14:46.47 | jantman | ahhh, thanks! |
14:46.49 | NovceGuru | BBHoss: haha that'd be nice |
14:46.56 | BBHoss | sure its possible |
14:46.56 | ITGuru | I guess my next question, would be how to I prove that separately? |
14:46.58 | [TK]D-Fender | NovceGuru: You sound surprised |
14:47.14 | [TK]D-Fender | ITGuru: "sip show peer 12345" |
14:47.17 | jantman | so I guess that would be SIP termination, as opposed to trunk? |
14:47.22 | NovceGuru | [TK]D-Fender: well it's "new" service they're offering that comes in over their fiber connection |
14:47.29 | NovceGuru | I thought....maybe.....blah |
14:47.32 | BBHoss | people have dumped the magic jack software and gotten the user/pass out of it, unlimited channels and minutes for 20 bucks/year |
14:47.36 | [TK]D-Fender | jantman: Origination actually, but close enough |
14:47.43 | jantman | ok, thanks! |
14:47.58 | [TK]D-Fender | ~itsp |
14:47.58 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
14:48.16 | jantman | ahhhh |
14:48.27 | NovceGuru | BBHoss: nice! |
14:48.32 | [TK]D-Fender | jantman: What you are looking for is within the scope of "cheap" |
14:48.36 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
14:48.38 | ITGuru | [TK]D-Fender, I have 7 registered subscriptions, and that is the 6 handsets, plus softphone for testing |
14:48.49 | [TK]D-Fender | jantman: 4 chan for a single DID... easy to find. |
14:49.00 | jantman | D-Fender: even better, since for now this is just a prototype job |
14:49.06 | [TK]D-Fender | ITGuru: None of which answers the "what about this ONE peer?" |
14:50.21 | ITGuru | [TK]D-Fender, I would like to dump just the debug info for that particular peer, how would I go about that - i seem only to be able to dump for all peers, and this is a live system, with like 3 calls a minute |
14:50.49 | [TK]D-Fender | ITGuru: "sip show peer 12345" <- PAY ATTENTION |
14:51.47 | *** join/#asterisk chendy (n=chatzill@58.61.197.37) |
14:51.55 | *** part/#asterisk TSCDan (n=demord@cpe-69-207-149-216.rochester.res.rr.com) |
14:51.59 | ITGuru | [TK]D-Fender, I feel dumb.... now I got the info I need |
14:52.05 | ITGuru | goes to shit in the shame corner |
14:52.12 | ITGuru | and by shit - I mean sit! lol |
14:52.25 | *** join/#asterisk tobias (n=tobias@cpe-076-182-095-118.nc.res.rr.com) |
14:53.23 | NovceGuru | blah this setup is going to fell ghetto |
14:53.36 | *** join/#asterisk moy (n=moy@74.12.127.97) |
14:54.47 | ITGuru | sip show peer 0301 - This is the peer having the problem - it's output is here - http://pastebin.com/da0cffd9 @ [TK]D-Fender BBHoss |
14:55.38 | [TK]D-Fender | ITGuru: Reg. Contact : sip:0301@192.168.10.172:5060;transport=UDP |
14:55.48 | *** part/#asterisk jantman (n=jantman@nat02-hill-ext.rutgers.edu) |
14:56.09 | [TK]D-Fender | ITGuru: Its registered. If you see no SIP debug on a call attempt then the phone doesn't LIKE what you're telling it to do. Most likely a local dialplan error |
14:58.58 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-3e5eaffbbeafa5b5) |
15:07.00 | aiksa[LV] | could it be that a massive change in clock +/- 2h might crash asterisk? |
15:07.14 | aiksa[LV] | system clock on linux of course |
15:07.19 | ITGuru | [TK]D-Fender, I'm guessing that a lot of work to do as well? |
15:07.22 | aiksa[LV] | ? |
15:07.54 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-5766518fd7928974) |
15:07.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
15:08.09 | tzafrir_laptop | aiksa[LV], I'm not aware of such a bug |
15:08.11 | [TK]D-Fender | ITGuru: ...huh? |
15:08.44 | aiksa[LV] | tzafrir_laptop: i just launched ntpdate from one of my timeservers which switched local clock back by two hours |
15:09.19 | tzafrir_laptop | aiksa[LV], you do have proper timezone settings? |
15:09.39 | aiksa[LV] | and in the very instant asterisk process disapeared |
15:09.50 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
15:09.59 | ITGuru | [TK]D-Fender, Each extension should be identical in the dial plan - It's wierd that there is an issue. What if I put in different extension credentials? An extension that I know is working? Should this help me to narrow down where the problem is? |
15:10.03 | aiksa[LV] | tzafrir_laptop: - ill check that |
15:10.08 | aiksa[LV] | it still seams strange |
15:10.24 | aiksa[LV] | and might explain a few issues i have had in the past |
15:10.38 | aiksa[LV] | /var/log/asterisk/full gives no hints as to why the process died |
15:10.47 | [TK]D-Fender | ITGuru: You're saying the PHONE can't dial. I said LOCAL DIALPLAN. Your PHONE doesn't like what you are dialing. |
15:10.51 | [TK]D-Fender | ITGuru: not ASTERISK. |
15:11.16 | [TK]D-Fender | ITGuru: And stop calling devices "extensions" |
15:12.01 | aiksa[LV] | [TK]D-Fender: "Each extension should be identical in the dial plan " - make your head spin doesnt it? |
15:12.19 | aiksa[LV] | just a bunch of identical lines in extensions.conf |
15:13.35 | ITGuru | Okay, Thanks for the lesson, I just wanted to narrow down where the problem is - The phone rings, but that is it - when you pick it up, you hear nothing - if it was just to do with what your dialling on the handset, then it should be able to take incoming calls - it rings, but when you pick it up, nothing |
15:14.05 | aiksa[LV] | tzafrir_laptop: just checked - the timezone is set correctly, so it could be wrong settings of system clock. Ill change them in BIOS tonight |
15:14.07 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
15:14.13 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
15:14.25 | [TK]D-Fender | ITGuru: And now you're talking about audio issues, not a "can't place calls to *" issue. Let me know when you've got your head screwed on straight. |
15:14.51 | [TK]D-Fender | ITGuru: EVERYTHING appears to be bad and we're hearing it in bits and pieces and I really can't waste time on a runaround now. |
15:15.27 | tzafrir_laptop | aiksa[LV], hwclock --systohc |
15:15.46 | aiksa[LV] | sets the hwclock to current "date"? |
15:15.59 | jtodd | Don't know of anyone else has asked this recently, but does anyone know of a pre-built program (based on Asterisk) that will call a list of customers and do automated surveys? |
15:16.08 | ITGuru | [TK]D-Fender, Easy with the language, I understand that I'm not getting the terminology 100% correct, and I understand that I'm probably not conveying my issues as well as I could, but seriously, no need for insults, I'm obviously not as competent as you are. |
15:16.12 | verywiseman | can i use Grandstream FXS/FXO ATA with asterisk |
15:16.14 | jtodd | I know those are annoying, but assume that it will be used for good instead of evil. :-) |
15:16.38 | jdnWEST | jtodd: YOUR THE GUY! (everyone get him!) |
15:17.06 | jtodd | "Your automobile insurance has expired! Press 1 to rene<gaaaaack>" |
15:17.13 | [TK]D-Fender | ITGuru: There was no insult there. I'm just not wasting time trying to sort out your now apparently numerous issues with it. |
15:17.42 | [TK]D-Fender | ITGuru: When you've got things straightened out I might offer to help with some specific point once progress is made. |
15:18.20 | NovceGuru | so thoughts on vitelity for business class (5 channels) sip trunking? |
15:18.23 | [TK]D-Fender | verywiseman: ... |
15:18.24 | [TK]D-Fender | ~gs |
15:18.25 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
15:19.15 | ITGuru | [TK]D-Fender, I understand, I'm sorry - I'm just trying to get through the issues - I may be wrong, and it could be all related, and be just one issue. What is confusing me, the fact that all the endpoints can be seen by the asterisk server - so the problem must be the phones - but the phones are all identical, and just 2 of them do not work, the rest do |
15:19.18 | aiksa[LV] | [TK]D-Fender: btw I have had experience (cough, cough) with BT102 in the past. Is every product from gs THAT bad? |
15:19.43 | [TK]D-Fender | aiksa[LV]: YMMV, and I would never risk it. |
15:19.57 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
15:20.15 | aiksa[LV] | ITGuru: those phones can make calls (ring appears on the other end) but no audio goes through? |
15:20.47 | aiksa[LV] | is the network topology the same for all of the phones vs. asterisk? |
15:21.17 | ITGuru | aiksa[LV], the phones in question ring when they are dialled, but cannot call out - and the topology of the network is the same for all phones |
15:21.56 | aiksa[LV] | ok. when those phones ring, can you lift up and hear audio? |
15:22.10 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
15:22.21 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
15:23.09 | SQLDarkly | Good Morning. |
15:23.50 | kannan | will "reload " command in * CLI reload manager.conf? |
15:24.15 | SQLDarkly | just do asterisk -rx "restart gracefully" |
15:24.24 | SQLDarkly | with the quotes |
15:24.26 | [TK]D-Fender | kannan: Should |
15:25.50 | *** join/#asterisk mog (n=mog@nat/digium/x-42ccadcb380d1df3) |
15:25.50 | *** mode/#asterisk [+o mog] by ChanServ |
15:27.29 | Shizuo | tzafrir_laptop: Hi |
15:27.54 | tzafrir_laptop | Shizuo, hi |
15:28.00 | Shizuo | Why do you guys hate Grandstream? Did they denied you guys funding? |
15:28.11 | Shizuo | tzafrir_laptop: Is is easy to develop drivers for Asterisk? |
15:29.22 | tzafrir_laptop | Shizuo, I mostly have exprience with chan_zap/chan_dahdi and the Zaptel/Dahdi drivers |
15:29.48 | Shizuo | tzafrir_laptop: I'm thinking of building a device |
15:29.50 | orTix | Do or did you guys also use or used softphone?> |
15:29.54 | tzafrir_laptop | do you have any prior experince with C? Linux? Drivers for what? |
15:30.02 | [TK]D-Fender | Shizuo: Perhaps making shoddy products has something to do with it. |
15:30.02 | Shizuo | tzafrir_laptop: Lots of C experience |
15:30.10 | Shizuo | [TK]D-Fender: I don't agree |
15:30.33 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
15:30.33 | *** mode/#asterisk [+o russellb] by ChanServ |
15:30.39 | Shizuo | [TK]D-Fender: This looks like groupthink to me |
15:30.52 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
15:30.56 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-531e82755baeb200) |
15:31.14 | [TK]D-Fender | Shizuo: I guess you are grand judge of that then, aren't you.... |
15:31.17 | Shizuo | tzafrir_laptop: The thing is: Some apps have really shoddy device driver APIs |
15:31.19 | coppice | if making shody products was a reason to not use stuff, nobody would use any VoIP kit at all |
15:31.25 | Shizuo | coppice: Exactly |
15:31.44 | eppigy | BOOYA |
15:31.49 | Shizuo | coppice: I bet Grandstream rejected to fund the project, making the group leaders tell everyone to hate the company |
15:32.07 | [TK]D-Fender | Shizuo: Wow, and that isn't paranoia. |
15:32.18 | Shizuo | [TK]D-Fender: That's just normal OSS behaviour |
15:32.23 | coppice | oh, grandstream stuff is nasty, but is it really worse than most other stuff? |
15:32.28 | *** join/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net) |
15:32.46 | [TK]D-Fender | Shizuo: No, the constant complaints from users complaining about audio issues flakey firmware, etc don't count |
15:32.48 | Shizuo | [TK]D-Fender: I coexist with lots of people from lots of project and that behaviour is not only normal, but it has become the rule, recently |
15:33.17 | Shizuo | Audio issies where? Only at Asterisk? Using any app? |
15:33.31 | [TK]D-Fender | Shizuo: Fine and if you want to walk in with that hypothesis here then you are applying your theory blindly. |
15:33.47 | Shizuo | [TK]D-Fender: It's safe to do that nowadays |
15:33.56 | jtodd | Shizuo: No, that is not the case. I believe most of the people on this channel have made their own opinions through direct experience, as most of the people that you're speaking with have been doing VoIP for >5 years. |
15:34.14 | SQLDarkly | Fender are you knocking Theory again ;) |
15:34.24 | SQLDarkly | j/k |
15:34.26 | Shizuo | jtodd: I own 2 racks full of GXW4024 gateways |
15:34.30 | coppice | people are working around problem in Sonus, Audiocodes, Cisco and all the biggest vendors. the industry standard for VoIP firmware is utter crap |
15:34.30 | Shizuo | jtodd: They're fine |
15:34.41 | [TK]D-Fender | Shizuo: So you're ok to make sweeping statements walking in the door blind yet those who've dealt with people with actual complaints here for YEARS don't get to judge based on experience? That is retarded |
15:34.45 | jtodd | Shizuo: I have no beef with Grandstream currently, though my past experiences with their initial consumer products were less than favorable. Recently, though, I have no idea how well they work. |
15:35.07 | Shizuo | [TK]D-Fender: Unfortunately, groups are fast to pick enemies |
15:35.30 | Shizuo | [TK]D-Fender: That behaviour is called groupthink for a reason |
15:35.46 | eppigy | oh man |
15:35.49 | eppigy | here we go |
15:35.54 | jtodd | Shizuo: Sometimes a cigar is just a cigar. |
15:35.56 | Shizuo | Having the official bot badmouth a vendor because is stupid |
15:35.58 | rob0 | I was dissuaded from Grandstream by this channel, but I think the opinions sounded fair. Asterisk isn't out begging for funding, Digium is a profitable company. |
15:36.09 | eppigy | Shizuo: do you have a degree in psychiatry or some shit |
15:36.09 | [TK]D-Fender | Shizuo: yes, so are "holocost-deniers" |
15:36.15 | eppigy | sociology |
15:36.20 | eppigy | english major? |
15:36.26 | Shizuo | ? |
15:36.34 | eppigy | [TK]D-Fender: the holocaust is 100% exagerated |
15:36.44 | eppigy | well I guesss 7-% |
15:36.48 | eppigy | 70% |
15:36.57 | *** join/#asterisk maddog01 (n=minotaur@d221-91-175.commercial.cgocable.net) |
15:36.57 | Shizuo | Holocost just looks like Holographic Cost to me |
15:37.02 | SQLDarkly | 60% of the time is works Everytime! |
15:37.03 | [TK]D-Fender | Shizuo: And all the psychology you can must doesn't undo a known checkered past. |
15:37.17 | [TK]D-Fender | muster* |
15:37.21 | Shizuo | [TK]D-Fender: Ok, if you're happy with the boundaries given to you by your masters, ok |
15:37.26 | Shizuo | I guess most people tend to live that way |
15:37.41 | SQLDarkly | lol masters. |
15:37.41 | jtodd | notes the sign beside the bridge that says "Do not feed the trolls." |
15:37.41 | [TK]D-Fender | Shizuo: Masters huh? And who would those be? |
15:37.44 | rob0 | Shizuo is coming across like a fool, with a faulty assumption leading to faulty conclusions. |
15:38.06 | rob0 | Pipe up with some evidence about the "denied funding". |
15:38.19 | Shizuo | rob0: Sorry, but the official (# and not ##) channel for Asterisk should never badmouth hardware vendors like that |
15:38.31 | Shizuo | rob0: It's childish and makes the whole thing look amateur |
15:38.41 | rob0 | ~sangoma |
15:38.41 | jbot | somebody said sangoma was a Canadian based company that makes PRI and Analog cards. See their site at http://www.sangoma.com/ |
15:38.46 | SQLDarkly | The channel isnt badmouthing |
15:38.51 | [TK]D-Fender | rob0: No, he only layed funding as "suspect". He has absolutely no factual basis, only vauge psychology theory. |
15:38.55 | rob0 | That's a Digium competitor. |
15:39.01 | SQLDarkly | its an opinion |
15:39.21 | Shizuo | Its the official Asterisk channel saying that company X is crap |
15:39.24 | [TK]D-Fender | Shizuo: O RLY? |
15:39.50 | SQLDarkly | If I crap then guess whats its crap. Sometimes companies really make crap |
15:39.54 | [TK]D-Fender | Shizuo: Since when does this channel have to stifle people who happen to frequent it? |
15:40.00 | SQLDarkly | Cisblows Call Manager for one |
15:40.00 | Shizuo | No problem... |
15:40.17 | Shizuo | In fact, it's not my channel and not my product, so |
15:40.19 | jjshoe | welcome to #bicker full of whiney bitches since 1996 |
15:40.29 | Shizuo | :D |
15:40.42 | SQLDarkly | Great here comes the Chuck Norris puns |
15:40.45 | rob0 | My "groupthink" was reinforced by this analysis. |
15:40.50 | eppigy | bottom line 6 million jews did not die in the holocaust |
15:41.02 | SQLDarkly | Starting to sound like WoW Barrens Chat in here |
15:41.05 | [TK]D-Fender | Shizuo: the channel isn't saying anything. Just a shared optinion from many prominent members who have no particular affiliations with any hardware vendor. That's a diverse bunch. |
15:41.17 | SQLDarkly | Bingo Fender |
15:41.24 | Shizuo | [TK]D-Fender: Sure, yeah, sure |
15:41.31 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
15:41.57 | [TK]D-Fender | Shizuo: YOU'RE the newb here, and want to tell us "how it is"? Sorry, nobody is buying that line. |
15:42.23 | Shizuo | Whatever, men |
15:42.28 | rob0 | jtodd was right, I'm done. |
15:42.56 | SQLDarkly | Ok boot the fool and lets get back to doin what we do. "Trashing Vendors" |
15:42.59 | timeshell | Is there an oslec patch that will work with dahdi? |
15:43.07 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
15:43.09 | Shizuo | "Oh, we are not a hate group, our leaders share a opinion that these people should be shot and killed, but that's not the group's opinion" |
15:43.14 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
15:43.38 | Shizuo | It's a good line to avoid being jailed, but fails at common sense |
15:43.57 | *** join/#asterisk voipnet-tech (n=voipnett@216.195.128.62) |
15:43.57 | clyrrad | Do any of you guys know of a way to generate a ring tone for X number of seconds? |
15:44.12 | eppigy | clyrrad: Shizuo is our resident expert |
15:44.18 | Shizuo | Thanks |
15:44.18 | clyrrad | I know about "ringing" but cant specify an interval |
15:44.20 | voipnet-tech | is this the right channel for asterisk support? |
15:44.30 | eppigy | voipnet-tech: yes please direct questions to Shizuo |
15:44.58 | voipnet-tech | thanks, and hello. Shizuo is this correct? |
15:45.04 | SQLDarkly | Yes it is. Just an off-topic conversation going on. |
15:45.09 | *** join/#asterisk arpu (n=arpu@chello080109017021.12.14.vie.surfer.at) |
15:45.12 | [TK]D-Fender | Shizuo: You're right and noone should advise others against products that are the product of mass recalls like Bridgestone tires whose blowouts have killed people. |
15:45.41 | [TK]D-Fender | Shizuo: And of course nyou're also comparing our opinions to those advocating MURDER. NIIIICCCE |
15:45.44 | SQLDarkly | People kill people not tires |
15:45.48 | SQLDarkly | or is it guns |
15:45.51 | SQLDarkly | i forget |
15:46.01 | SQLDarkly | no doubt lol |
15:46.03 | [TK]D-Fender | SQLDarkly: No. Guns don't kill people, *I* do :p |
15:46.16 | voipnet-tech | I'm getting a Segmentation fault when I try to dial console/dsp. chan_alsa is dying. this is a new linux install. have not had this problem on other servers. what should I try |
15:46.23 | *** join/#asterisk DarkRift (n=dark@65.92.169.136) |
15:46.24 | Shizuo | [TK]D-Fender: Hey, I don't want to stand between companies and your inconditional hatred for them |
15:46.33 | Shizuo | [TK]D-Fender: Feel free to keep acting like that |
15:46.38 | clyrrad | Hrm, anyone have any suggestions for me? Im all eyes :) I just need to play a ringing sound to the caller for X number of seconds.... |
15:47.41 | [TK]D-Fender | Shizuo: unconditiona? No, being plaged by handset echo is a GREAT basis for dislike of a product <- |
15:47.46 | jjshoe | clyrrad did you search for "play ringing" on voip-info.net ? |
15:47.53 | NovceGuru | s |
15:47.58 | *** join/#asterisk mintos (n=mvaliyav@123.236.161.151) |
15:48.20 | Shizuo | [TK]D-Fender: Sure! That sure allows for eternal badmouthing of the said company, under the official bot of the official channel |
15:48.43 | Shizuo | [TK]D-Fender: And I really mean eternal! No more Firestone tires for me or Greatsuck gateways for my company! |
15:48.59 | Shizuo | [TK]D-Fender: I feel much safer now after hearing your advice |
15:49.13 | [TK]D-Fender | Shizuo: You, being a user of their products are therefor unbiased as well and couldn't possibly be suffering from "fanboy-ism" <- Want to play the psych game? goes both ways. |
15:49.14 | jjshoe | is this a good time to do the following? |
15:49.17 | jjshoe | ~grandstream |
15:49.18 | jbot | hmm... grandstream is the Yugo of VoIP hardware. Run. Run away now. |
15:49.27 | SQLDarkly | clyrrrad why do you want to play a fake ring? Just record a sound and play it back......... I do not see the point but I am not in your shoes so have at it ;) |
15:49.33 | Shizuo | [TK]D-Fender: Sure! I'm a fanboy of VoIP gateway hardware! |
15:49.50 | jjshoe | SQLDarkly I've used it in cases where a t1 picks up before the telco can even get a ring in |
15:50.00 | [TK]D-Fender | Shizuo: Not possibly even a specific brand which is under contention!. |
15:50.09 | [TK]D-Fender | load chan_hypocrisy.so |
15:50.12 | coppice | [TK]D-Fender: when a snom or many other handsets have echo problems, people say turnt he gain down. When a grandstream does it they say grandstream sucks. realistically every VoIP phone out there is pretty mediocre |
15:50.16 | Shizuo | [TK]D-Fender: They'll be releasing new products next week, I think I'll sleep on the street, in front of the store, to be the first to get them! |
15:50.24 | voipnet-tech | i don't mean to interupt your wonderful offtopic discussion here, but I have 4 servers to get working today and I can't get console audio to work, and i need it for external paging. is there anyone here that cares to help me figure out why asterisk gives me a Segmentation Fault and exits? |
15:50.24 | [TK]D-Fender | ERROR : chan_hypocrisy.so is already loaded |
15:50.27 | SQLDarkly | jjshoe good point. I stand corrected |
15:50.28 | clyrrad | SQLDarkly: its for a special application I am developing, I background (Silence/3) for example, but I dont want the caller to think the call was dropped and have dead air, so instead I would like to play a ringing sound so the caller knows they are being connected |
15:50.50 | Shizuo | coppice: Too bad that GS never opened their deep pockets to the project, like some other companies do |
15:50.57 | Shizuo | coppice: That means eternal badmouthing of the brand |
15:51.40 | [TK]D-Fender | Shizuo: and I don' work for Digium, or code for *, nor do many of the people who've used it for years. |
15:52.01 | SQLDarkly | Shizuo hush lol this is not a channelf or your bullcrap. I HATE many companies like CISBLOWS but their 79XX phones rock balls. Its just about what works nothing more. |
15:52.06 | jjshoe | clyrrad record ringing and play it back, as SQLDarkly suggested |
15:52.27 | Shizuo | [TK]D-Fender: There is no need to convince me, ok? Your masters have teached you right: GS sucks... I'm with you on that one; |
15:52.28 | SQLDarkly | A collection of opinions is all your reacting to. Not some Digium conspiracy |
15:52.34 | Shizuo | [TK]D-Fender: No more GS or Firestone for me |
15:52.55 | jjshoe | Shizuo I agree, can you take this to #teenchat with your boyfriend? |
15:52.57 | [TK]D-Fender | Shizuo: You fail to understand : I *HATE* no masters here, and no paid loyalty to ANY BRAND |
15:53.10 | [TK]D-Fender | Shizuo: You are talking out your ass. |
15:53.15 | [TK]D-Fender | have* |
15:53.19 | Shizuo | [TK]D-Fender: They're your masters, you obviously don't hate them |
15:53.21 | clyrrad | jjshoe: I just thought there would be an eaiser way using Playtones or something similar |
15:53.39 | [TK]D-Fender | Shizuo: "they"? Who the hell is "they"? |
15:53.53 | Shizuo | [TK]D-Fender: The group is one of them |
15:53.57 | Shizuo | [TK]D-Fender: Master n 1 |
15:54.11 | Shizuo | You can name the others, as I don't know you that much :D |
15:54.16 | jjshoe | Qwell ping |
15:54.27 | SQLDarkly | Shizuo go take your NerdRage elsewhere please. People have work to do and you obviously write for the a tabloid |
15:54.34 | [TK]D-Fender | Shizuo: Pshychology double-skeap. You are so completely full of shit, its laughable. Take your entire psych background and toss it right out the window because ti is worthless here. |
15:54.35 | jjshoe | clyrrad hrm, looks like you might be able to, did you research playtones on voip-info.org ? |
15:54.37 | [TK]D-Fender | speak* |
15:54.37 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
15:54.46 | jjshoe | clyrrad I hit an article which pointed to indications.conf and seemed to make it sound plausable |
15:55.00 | Shizuo | A very people group of people :D |
15:55.02 | clyrrad | jjshoe: its what I have been doin gthe last 30 minutes, just wanted to check here to see if anyone else has done this before |
15:55.04 | Shizuo | A very polite group of people :D |
15:55.17 | [TK]D-Fender | Shizuo: "Secret Masters"... go off Illuminati-hunting... I hear its all the rage now |
15:55.24 | jjshoe | clyrrad what link are you at right now on voip-info ? |
15:55.27 | Shizuo | And that's only because I complained of an official bot badmouthing a company |
15:55.39 | clyrrad | jjshoe: Asterisk+cmd+Playtones |
15:55.41 | Shizuo | The excessive reaction to that showed that I was correct on my assumptions |
15:55.47 | jjshoe | clyrrad full url please. |
15:55.54 | SQLDarkly | DO you work for GS? |
15:55.59 | [TK]D-Fender | Shizuo: Official bot? Wow you continue to assert I have any particular affiliation.... |
15:56.17 | clyrrad | jjshoe: http://www.voip-info.org/wiki/view/Asterisk+cmd+Playtones |
15:56.20 | Shizuo | [TK]D-Fender: Oh crap, you're stupid as a doorknob |
15:56.33 | SQLDarkly | GS simply puts out garbage compared to excellent phones like the 79xx |
15:56.39 | Shizuo | [13:20] <jbot> GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
15:56.48 | SQLDarkly | and THAT cannot be denied even by GS themselves |
15:56.51 | [TK]D-Fender | Shizuo: No, you are master of the blind accusation that those of opinion must be paid to think or "controlled" |
15:57.01 | [TK]D-Fender | Shizuo: So much for self-determination. |
15:57.12 | Shizuo | No, you're stupid. I mentioned BOT several times and you still think I'm talking about YOU |
15:57.18 | Shizuo | It can not get more stupid than that |
15:57.20 | SQLDarkly | Shizuo you your windows still dont you? |
15:57.35 | Shizuo | SQLDarkly: You your doors, don't you? |
15:57.37 | [TK]D-Fender | Shizuo: Do clarify exactly what bot statement and I will enlighten you. |
15:57.57 | Shizuo | [TK]D-Fender: Pasted on the channel, Einstein |
15:58.16 | [TK]D-Fender | Shizuo: Feel free you repaste it now specifically :) |
15:58.22 | [TK]D-Fender | to* |
15:58.24 | Shizuo | No I won't, it's 8 lines away |
15:58.30 | Shizuo | If you're that limited, sorry |
15:58.33 | SQLDarkly | Shiz have you even used GS with Asterisk? Yes they do work but they have to be replaced a lot |
15:58.34 | [TK]D-Fender | ~gs |
15:58.34 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
15:58.37 | [TK]D-Fender | ^^ this guy here? |
15:59.05 | SQLDarkly | Quality phones like 79xx by Cisblows vs a Garbage GS phone that is fine for testing but not enterprise level deployment |
15:59.07 | voipnet-tech | i'm checking here for someone that might be able to give me some direction to why i can't get console/dsp calls to work |
15:59.07 | voipnet-tech | [10:58] <voipnet-tech> initially chan_alsa wouldn't compile because make menuselect had it XXX out says: depends on asound(e) |
15:59.07 | voipnet-tech | [10:59] <voipnet-tech> i think that's because i ran .configure before i installed alsa |
15:59.07 | voipnet-tech | [10:59] <voipnet-tech> now alsa is installed and working (alsamixer and play/aplay work) |
15:59.08 | voipnet-tech | [10:59] <voipnet-tech> and after rerunning ./configure and make menuselect |
15:59.10 | voipnet-tech | [10:59] <voipnet-tech> i can now select chan_alsa |
15:59.12 | voipnet-tech | [10:59] <voipnet-tech> and make, make install |
15:59.14 | voipnet-tech | [10:59] <voipnet-tech> and asterisk runs |
15:59.16 | voipnet-tech | [11:00] <voipnet-tech> if I start asterisk with safe_asterisk, i always get a busy/congested calling console/dsp |
15:59.19 | voipnet-tech | [11:00] <voipnet-tech> if i start it as just asterisk, i can call it, but then asterisk gets a segmentation fault |
15:59.25 | rob0 | yikes |
15:59.26 | [TK]D-Fender | voipnet-tech: STOP SPAMMING |
15:59.35 | LemensTS | ~pastebin |
15:59.36 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
15:59.43 | rob0 | Now we have a new focal point for our wrath. :) |
15:59.45 | [TK]D-Fender | Shizuo: So, that one? |
15:59.52 | [TK]D-Fender | Shizuo: Lets be clear.... |
15:59.57 | Shizuo | voipnet-tech: Stop trying to get support here! This is the place to tap people on the back and badmouth companies! |
16:00.04 | Shizuo | voipnet-tech: Not for support! |
16:00.08 | voipnet-tech | this is why asterisk sucks, you guys |
16:00.25 | voipnet-tech | i'm asking legitmate questions for support |
16:00.27 | SQLDarkly | Wrong Shiz. You are getting this channel off topic. |
16:00.29 | LemensTS | voipnet-tech: pastebin your problem then post the link |
16:00.36 | voipnet-tech | and you guys are spamming the room with the most offtopic crap |
16:00.46 | SQLDarkly | Yes voip but your spamming it. Address someone and start from the beginning |
16:00.52 | SQLDarkly | we dont even know what * version you have |
16:00.55 | SQLDarkly | what distro your on |
16:00.56 | SQLDarkly | etc |
16:01.06 | voipnet-tech | and i ask legit question all be-it lengthy, and i just get shit for it |
16:01.08 | jjshoe | clyrrad do you see where it says to look for what tones are defined? |
16:01.16 | SQLDarkly | Key things in order to help you clear your problem |
16:01.17 | [TK]D-Fender | Shizuo: Who are you "helping" now troll? |
16:01.24 | LemensTS | voipnet-tech: ask it or leave |
16:01.33 | clyrrad | jjshoe: yea I am still checking on it and watching all the nonsense happing in this channel :( |
16:01.33 | [TK]D-Fender | Shizuo: So answer the previous question. |
16:01.35 | voipnet-tech | i have asked |
16:01.41 | voipnet-tech | that's what the paste was |
16:01.47 | voipnet-tech | i'm explaining my situation |
16:01.50 | voipnet-tech | i'm checking here for someone that might be able to give me some direction to why i can't get console/dsp calls to work |
16:01.50 | voipnet-tech | [10:58] <voipnet-tech> initially chan_alsa wouldn't compile because make menuselect had it XXX out says: depends on asound(e) |
16:01.50 | voipnet-tech | [10:59] <voipnet-tech> i think that's because i ran .configure before i installed alsa |
16:01.50 | voipnet-tech | [10:59] <voipnet-tech> now alsa is installed and working (alsamixer and play/aplay work) |
16:01.51 | voipnet-tech | [10:59] <voipnet-tech> and after rerunning ./configure and make menuselect |
16:01.53 | voipnet-tech | [10:59] <voipnet-tech> i can now select chan_alsa |
16:01.54 | SQLDarkly | voipnet-tech you have yet to tell anything about your specs dude |
16:01.55 | voipnet-tech | [10:59] <voipnet-tech> and make, make install |
16:01.55 | LemensTS | voipnet: im not reading that you need to pastebin it. sob. |
16:01.57 | voipnet-tech | [10:59] <voipnet-tech> and asterisk runs |
16:01.57 | jjshoe | clyrrad when you follow that link, do you see ring? |
16:01.59 | voipnet-tech | [11:00] <voipnet-tech> if I start asterisk with safe_asterisk, i always get a busy/congested calling console/dsp |
16:02.02 | voipnet-tech | [11:00] <voipnet-tech> if i start it as just asterisk, i can call it, but then asterisk gets a segmentation fault |
16:02.05 | voipnet-tech | :-) |
16:02.17 | LemensTS | im not reading 3 words on each line 20 lines long |
16:02.18 | voipnet-tech | god i hate this room |
16:02.24 | voipnet-tech | Broadsoft rules, asterisk blows |
16:02.24 | SQLDarkly | voipnet-tech we need infor about your rig. |
16:02.26 | clyrrad | jjshoe: I see in indications.conf there is a tone defination for ring, I am going to try that |
16:02.26 | Shizuo | [TK]D-Fender: That issue is as clear as it can get. You're trying to make the discussion bigger as it fills your trollish void, but I'm not in the mood of cooperating |
16:02.30 | jjshoe | voipnet-tech sweet |
16:02.33 | jjshoe | clyrrad perfect :) |
16:02.39 | SQLDarkly | voipnet-tech how can we diagnose a problem with no information remotely? |
16:02.42 | SQLDarkly | answer that? |
16:02.50 | voipnet-tech | i gave you the info |
16:02.53 | [TK]D-Fender | Shizuo: Was that, or was that not the "bot" comment you're basing "rooom" bias on? |
16:02.58 | voipnet-tech | you're just refusing to read it unless i put it in pastebin |
16:03.04 | LemensTS | exactly |
16:03.09 | SQLDarkly | voipnet-tech wrong you didnt you gave symptoms of your problem |
16:03.22 | lowtek | Hey TK, good call on Telephony Depot, they rock! |
16:03.26 | SQLDarkly | voipnet-tech I asked 3 times already for versions and distro |
16:03.28 | Shizuo | [TK]D-Fender: Check what I pasted back there. Until then, I won't pollute the channel by giving you attention |
16:03.29 | jjshoe | clyrrad let me know how that works out |
16:03.35 | clyrrad | jjshoe: yup |
16:03.43 | *** join/#asterisk errr (n=mike@fedora/errr) |
16:04.10 | SQLDarkly | voipnet-tech You are insisting on insulting us saying the room sucks. Do not take out your anger on us. We are trying to help but certain information is needed. |
16:04.29 | jjshoe | welcome to #teenchat look out, everyone is on the rag right now! |
16:04.42 | SQLDarkly | voipnet-tech You obviously came here for some help the least you can do for the free support and our time is help us understand a bit more about your setup would you not agree? |
16:05.05 | voipnet-tech | yes, i'm using all the latest stable builds of everything |
16:05.08 | voipnet-tech | i said that |
16:05.23 | voipnet-tech | centos, *, zap |
16:05.37 | [TK]D-Fender | Shizuo: Well you don't seem to have confirmed that as the "room bias" bot statement. |
16:05.40 | jjshoe | latest stable build of which release? 1.2 1.4? |
16:05.41 | zoid_99 | I need a CLI command to move all messsages in someone's INBOX to someone else's INBOX. |
16:05.43 | SQLDarkly | That wasnt in your spam ;) Dont get snapping we are here to help we simply like to read organized information |
16:05.46 | voipnet-tech | 14 |
16:05.55 | jjshoe | zoid_99 asterisk cli? |
16:06.43 | SQLDarkly | now let me see what I can do for you mr meanie :) |
16:06.43 | zoid_99 | jjshoe: yeah |
16:06.43 | jjshoe | zoid_99 I don't know of any, but I'm not the most familiar |
16:06.43 | zoid_99 | voicemail forward new 1111 to 2222 |
16:06.43 | [TK]D-Fender | zoid_99: |
16:06.43 | SQLDarkly | voipnet-tech can you please pastebin the console output when you console dial for me please? |
16:06.46 | *** join/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:06.59 | KOCATEPE | hi all |
16:07.02 | SQLDarkly | Ensure debug is on |
16:07.05 | KOCATEPE | any one from Turkey ??? |
16:07.07 | voipnet-tech | sure |
16:07.08 | *** join/#asterisk Sargun (n=Sargun@66.151.148.225) |
16:07.15 | SQLDarkly | you can turn on debug in the console via logger.conf |
16:07.24 | SQLDarkly | be sure to turn it off before you go back into production |
16:07.53 | zoid_99 | jjshoe: I've started writing it, but it's going to require a little more work than I thought. |
16:08.02 | SQLDarkly | after you change the logger.conf you will need to asterisk -rx "restart gracefully" |
16:08.23 | jjshoe | zoid_99 oh? how come? |
16:08.28 | *** part/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:08.30 | *** join/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:08.36 | SQLDarkly | brb while your doing that I am in need of coffee. |
16:08.48 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
16:08.51 | *** part/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:08.52 | *** join/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:08.59 | zoid_99 | jjshoe: because I suck at C :) |
16:09.12 | [TK]D-Fender | zoid_99: You're coding a CLI command to do this? |
16:09.19 | zoid_99 | yes |
16:09.24 | jjshoe | zoid_99 ahh, why not write an agi in a language you know? |
16:09.32 | jjshoe | zoid_99 or doesn't asterisk have a system() ? |
16:09.32 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
16:09.39 | [TK]D-Fender | zoid_99: And yes, it is quite challengiing give the way the files get numbered, etc |
16:09.48 | jameswf | HEY #ASTERISK your OT |
16:09.48 | clyrrad | jjshoe: no joy, didnt do what I need, after the 3 seconds are up, there is no ringing when the actuall phones are ringing, looks like its back to the drawing board |
16:09.50 | jameswf | :) |
16:09.57 | SQLDarkly | FYI for all the naysayers. This room is a great resource just understand people come in here and start yellin and need to address their issue in an organized manner. You have to think how many questions get asked here and how many are repeat. THis room had taught me a lot over the years and my hat goes off to that surly bastard fender lmao he subjects himself to helpdesk 24/7 |
16:10.20 | jjshoe | clyrrad hrm, the console showed it trying to play the tones? honestly I've always just played back a ringing gsm :) |
16:10.22 | [TK]D-Fender | SQLDarkly: Almost :) |
16:10.27 | zoid_99 | [TK]-Fender: I thought about doing it externally but I need the locking capabilaties of app_voicemail to keep from having a race condition |
16:10.43 | voipnet-tech | well... it appears the last reboot fixed it |
16:10.45 | voipnet-tech | nvm |
16:10.47 | jameswf | SQLDarkly: point of order I believe [TK]D-Fender's parents were married |
16:10.49 | voipnet-tech | you may resume your offtopic bs |
16:10.54 | clyrrad | jjshoe: so you recorded a ringing sound for how every many seconds you needed? If so, how did you acomplish the recording of the ring sound? |
16:10.55 | voipnet-tech | thanks for trying SQLDarkley |
16:10.58 | voipnet-tech | gold star for u |
16:11.08 | [TK]D-Fender | SQLDarkly: I love how people feel I'm affiliated with XYZ. With my loyalty lasts until something better comes along or something I liked more than something else goes south :) |
16:11.09 | zoid_99 | jjshoe: yes it does but I need to ensure that the voicemails are locked while I do this |
16:11.11 | jjshoe | clyrrad just a single ring, then you can decide how much to play it |
16:11.23 | clyrrad | jjshoe: how did you record the single ring? |
16:11.23 | jjshoe | zoid_99 so why not implement the lock algo. in another language? |
16:11.57 | zoid_99 | jjshoe: because asterisk will be twiiddling with the files behind my back |
16:12.07 | [TK]D-Fender | SQLDarkly: Even Asterisk itself :) Every piece of hardware I run here at the office can play nice with PLENTY of alternative PBX solutions besides *. Not one piece of my solution "owns me". Masters they say... LOL!~!!!! |
16:12.10 | *** join/#asterisk riddlebox (i=80fc9a6f@gateway/web/ajax/mibbit.com/x-f3c0fdef7df0ff2b) |
16:12.31 | [TK]D-Fender | SQLDarkly: No F-ING clue :) |
16:12.37 | jjshoe | zoid_99 that doesn't make much sense... |
16:12.43 | SQLDarkly | D-Fender I am surprised he isnt destroying bloggers for their opinions and experiences |
16:12.55 | SQLDarkly | ok back now voipnet-tech |
16:12.55 | jjshoe | zoid_99 are you suggesting app_voicemail.c ignores locks by any application other then itself? O.o |
16:13.04 | SQLDarkly | Do you have your output ready? |
16:13.15 | LemensTS | SQLDarkly: it fixed when he restarted |
16:13.22 | clyrrad | jjshoe: Did you see my last msg? |
16:13.28 | jameswf | jjshoe: I am pretty sure you touch .lock it will cause issues |
16:13.38 | SQLDarkly | LemensTS ah excellent |
16:13.54 | LemensTS | You got a gold star for your help too if you read up heh |
16:13.54 | [TK]D-Fender | lowtek: You're welcome |
16:14.11 | [TK]D-Fender | lowSo far I an all those I have referred there have been very happy with them |
16:14.14 | zoid_99 | jjshoe: I'm saying that app_voicemail numbers files and renumbers files. If I move a file while someone is leaving a voicemail then things will get out of whack |
16:14.22 | jameswf | sometimes if an asterisk system crashes we have to go in anr rm .lock files |
16:15.00 | jjshoe | clyrrad show application ringing |
16:15.11 | jjshoe | zoid_99 presumably not if you lock it no? |
16:15.52 | clyrrad | jjshoe: This is what you used to record the ring sound? Or I am not following what you mean? |
16:15.57 | zoid_99 | jjshoe: old files may be locked but it doesn't prevent app_voicemail from creating a new voicemail file |
16:16.00 | voipnet-tech | sql problem fixed |
16:16.03 | timeshell | Looking for some detailed instructions to use OSLEC with DAHDI |
16:16.03 | voipnet-tech | new problem |
16:16.08 | voipnet-tech | who supports safe_asterisk ? |
16:16.13 | zoid_99 | jjshoe: and the new file will be numbered wrong |
16:17.28 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
16:17.38 | Shizuo | Oh Asterisk... |
16:18.03 | zoid_99 | but in app_voicemail I can use AST_LIST_LOCK (I think) |
16:18.15 | [TK]D-Fender | voipnet-tech: its part of the core |
16:18.38 | jjshoe | clyrrad type that in asterisk |
16:19.04 | jjshoe | zoid_99 ah, it's been a long time since I've looked at app_voicemail.c |
16:19.08 | clyrrad | jjshoe: yup I did, I am aware of this command, I am just not following what you are trying to tell me |
16:19.11 | *** part/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:19.14 | jjshoe | clyrrad use it. |
16:19.16 | *** join/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:19.42 | clyrrad | jjshoe: When I use that, after its doen the backgroud(silence/3) you get no ring when the phones are ringing |
16:20.07 | jjshoe | clyrrad how long do you ring for? |
16:20.17 | clyrrad | jjshoe: 3 seconds |
16:20.28 | jjshoe | clyrrad while you play silence for 3 seconds at the same time.. |
16:20.46 | clyrrad | jjshoe: not sure what you mean? |
16:21.06 | clyrrad | I call Ringing() first, then background(silence/3) |
16:22.07 | jjshoe | clyrrad ah |
16:22.12 | *** part/#asterisk KOCATEPE (n=admin@88.247.133.123) |
16:23.03 | clyrrad | jjshoe: but it does not work |
16:23.07 | jjshoe | clyrrad add a wait after the ringing |
16:25.50 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-71-22.w86-215.abo.wanadoo.fr) |
16:25.50 | [TK]D-Fender | clyrrad: Ringing WILL stop for audio... |
16:26.05 | clyrrad | jjshoe: the problem is as soon as you do Background(silence/3) the ringign stops |
16:26.18 | [TK]D-Fender | clyrrad: And why wouldn't it? |
16:26.19 | clyrrad | jjshoe: I need to record the ring sound as you initialy said, how did you manage this? |
16:26.24 | [TK]D-Fender | cyYou're telling it to play audio |
16:26.46 | clyrrad | jjshoe: To record the ring sound seems to be the only solution here since Playtones didnt work for me |
16:26.52 | lowtek | I didn't catch the first part of this, but why not just Wait(3) after Ringing()? |
16:27.14 | jjshoe | clyrrad an english definition of your call flow, and a pastebin of your context would sure help |
16:27.19 | clyrrad | lowtek: I have not tried that approach, let me give it a go |
16:27.24 | *** join/#asterisk hetii (n=vircuser@193.159.172.162) |
16:27.27 | hetii | hi |
16:27.36 | *** part/#asterisk wwalker (n=wwalker@mailbox.eclipsing.com) |
16:27.47 | jjshoe | clyrrad I just told you to try that and you didn't bother?? |
16:27.48 | lowtek | clyrrad: This is what TK was trying to say, Playing() the file silence/3 still engages the playback function and will terminate the ring function. |
16:27.59 | jjshoe | blah. |
16:28.10 | [TK]D-Fender | clyrrad: So as you've been told by others, jst "Wait(3) |
16:28.39 | clyrrad | Wait(3) does "not" work, it wont allow the user to press keys, this is why I had background(silence/3) |
16:29.17 | [TK]D-Fender | clyrrad: Then go make an audio recording of 3 seconds of ringing and background THAT |
16:29.19 | lowtek | clyrrad: Then find or create an audio file that sounds like a ring at whatever length you need and Background() it. |
16:29.34 | [TK]D-Fender | lowtek: tag, you're "it" :) |
16:29.36 | clyrrad | I need to have a bit of a pause where the user can hit a hidden feature before the phones start to ring, that is what I am trying to acomplish.... now having said that..... I cant just background(silence/3) becase the user will think the call was dropped, hence the reason I am trying to play a ringing sound.... |
16:29.42 | lowtek | lol |
16:30.05 | clyrrad | [TK]D-Fender: Yes, thats my question....... HOW to record that ring sound? I need the ring sound to match Asterisk ring sound so its seemless |
16:30.24 | [TK]D-Fender | clyrrad: Go monitor a call to "ringing" |
16:30.29 | clyrrad | lowtek: Yes, thats my question, I not sure how to record that rinning sound |
16:30.35 | [TK]D-Fender | clyrrad: this ain't Raw-Cat Science |
16:30.58 | clyrrad | [TK]D-Fender: I understand, but I have not done it b4, its easy to be an expert once you've done it once, I have not done this before |
16:31.07 | [TK]D-Fender | clyrrad: MONITOR <- |
16:31.16 | lowtek | clyrrad: You could record it right from asterisk with Record() if you want. Do a Answer(), Monitor(), Ringing(), Wait(3). |
16:31.16 | clyrrad | I am checking on that now |
16:31.28 | [TK]D-Fender | clyrrad: Start recording. Cal ringing, wait a few sec, hang up. |
16:31.32 | clyrrad | lowtek: thanks thats what I was looking for |
16:31.36 | lowtek | clyrrad: then just find the file it created, and Background() it. |
16:31.44 | *** join/#asterisk Alric (n=Alric@masq.hyperusa.com) |
16:31.44 | [TK]D-Fender | clyrrad: We're taling what, 4 lines of dialplan with 3 apps we hand-fed ? |
16:31.51 | clyrrad | lowtek: yup I see what your saying now, thanks |
16:31.59 | clyrrad | much apprecated :) |
16:32.14 | [TK]D-Fender | lowtek: No, not "record" unless he's doing a 3-way to bridge from a phone, etc |
16:32.46 | lowtek | clyrrad: right, use Monitor(), not Record(), my bad. |
16:33.15 | clyrrad | lowtek: no probs, I was just looking for the approach, what you defined was great, I can research based off that, thanks again for your help |
16:33.36 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
16:35.25 | Shizuo | Lol, Asterist |
16:36.43 | voipnet-tech | i'm gonna slit my asterist |
16:37.19 | clyrrad | haha |
16:37.45 | [TK]D-Fender | ~emo |
16:37.46 | jbot | /wrists |
16:37.55 | Shizuo | ~badmouthing |
16:38.38 | jjshoe | ~grandsuck |
16:38.54 | [TK]D-Fender | FAILURE :) |
16:39.06 | voipnet-tech | i've got one of those that i can't get to place outward calls |
16:39.15 | jjshoe | one of what? |
16:39.19 | voipnet-tech | a gateway |
16:39.22 | voipnet-tech | from grandsuck |
16:39.26 | voipnet-tech | 4 port fxo |
16:39.33 | jjshoe | oh, yeah, use it to level your desk off, or throw it away. |
16:39.39 | riddlebox | my grandstream phones work fine |
16:39.40 | [TK]D-Fender | voipnet-tech: What is an "outward call" |
16:40.04 | [TK]D-Fender | Shizuo: OMG, a testimonial, quick, squash him, his "masters" are clearly controlling him! |
16:40.05 | voipnet-tech | [TK]D-Fender, referring to calls where the gateway would dial out on the pots line |
16:40.27 | jjshoe | riddlebox I've used many a grandstream that work ok, I've also used many aastra's and polycom's that work great :) |
16:40.30 | riddlebox | ouch there are 10 foot poles |
16:41.13 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
16:41.17 | carranca | has anyone configured asterisk with the app h324m? |
16:41.25 | [TK]D-Fender | voipnet-tech: well if you provide debug and shot of your configs maybe someone might have idea of some settings to change to get it working. |
16:41.36 | [TK]D-Fender | voipnet-tech: Pasterbin is your friend. |
16:41.47 | voipnet-tech | it's not running with * |
16:41.52 | voipnet-tech | different softswitch |
16:42.12 | voipnet-tech | and i've taken full debugs and sent them to GS support and even they can't figure them out |
16:42.25 | voipnet-tech | they said that the debugs i sent to them are too confusing |
16:42.30 | voipnet-tech | i laughed |
16:42.31 | voipnet-tech | and hungup |
16:42.44 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
16:43.43 | voipnet-tech | i'll be able to fix it myself i'd bet, that's what always ends up happening, i just need time to sift thru the wireshark |
16:43.50 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
16:45.01 | kb3ien | anyone care to praise a softphone for windows, ideally a Free one, but a free (as in zero-monies) is okay too. |
16:45.10 | *** join/#asterisk DarkRift (n=dark@65.92.169.136) |
16:45.13 | voipnet-tech | x-lite |
16:45.32 | kb3ien | i never touch win32 with a *long* pole, its too krufty... |
16:45.41 | kb3ien | i'll pass the suggestion alon. |
16:45.46 | UQlev | kb3ien: phonerlite |
16:45.46 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
16:46.10 | [TK]D-Fender | kb3ien: All of the "free" ones have shortcomings that frustrate one group or another. |
16:46.21 | Shizuo | Omg, kb3ien: You're soo leet |
16:46.41 | pta200 | Any idea when Asterisk 1.4.23 will be officially released? Seems like there are a lot of fixes since 1.4.22 |
16:46.55 | [TK]D-Fender | kb3ien: Zoiper is "OK" and supports SIP/IAX & native call transfer... so that might do. X-Lite is a bit more stable, but more crippled |
16:47.04 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
16:48.19 | UQlev | [TK]D-Fender: does zoiper support proxy? |
16:48.50 | [TK]D-Fender | UQlev: No idea, I've only used it sparingly and direct to * |
16:49.26 | clyrrad | lowtek: That solution worked thanks bud much appreciated |
16:49.27 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
16:49.28 | UQlev | I have not found any proxy settings in zoiper |
16:49.34 | clyrrad | jjshoe: thanks awell for your assistance with this :) |
16:50.48 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
16:51.01 | *** part/#asterisk smach (n=smach@guy78-3-82-239-225-173.fbx.proxad.net) |
16:52.46 | NovceGuru | anybody care to recommend an ITSP for business sip trunking? (5 channels) |
16:53.44 | *** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
16:55.06 | clyrrad | ~ITSP |
16:55.06 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
16:55.13 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za) |
16:55.38 | clyrrad | ~itsplist |
16:55.44 | NovceGuru | I was thinking about actual experiences/input instead of that which i've seen 100 times :) |
16:55.59 | timeshell | Asterisk hacked? http://www.ic3.gov/media/2008/081205-2.aspx |
16:56.09 | russellb | ugh |
16:56.16 | russellb | please read blogs.digium.com ... |
16:56.26 | Shizuo | Lulz, owned |
16:56.45 | riddlebox | old news |
16:56.48 | russellb | specifically: http://blogs.digium.com/2008/12/06/sip-security-and-asterisk/ |
16:56.54 | timeshell | Guess that was a while ago |
16:57.35 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
16:58.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
16:58.55 | NovceGuru | ~itsplist-us |
16:58.56 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
16:59.49 | NovceGuru | I wonder if voicepulse ever picked up any more proxies to help with ping times |
17:02.05 | NovceGuru | they do have a shiny new site, i'm sold! |
17:02.24 | lowtek | NovceGuru: Try FlexPulse -> http://flexpulse.com |
17:02.31 | [TK]D-Fender | NovceGuru: I've had clients who've been quite happy with them. |
17:02.59 | Shizuo | Seriously... Why? |
17:03.02 | DarkRift | Anyone knows a good Canada Telco for Voip services or DID termination I can access through an asterisk installation I host ? |
17:03.04 | NovceGuru | I used them years ago when tinkering...they seemd decent |
17:03.57 | [TK]D-Fender | DarkRift: les.net & unlimitel.ca |
17:04.03 | DarkRift | Thank you |
17:04.04 | NovceGuru | my only issue is im in BFE and most ITSPs can't port my number |
17:04.08 | [TK]D-Fender | DarkRift: both very solid |
17:04.52 | DarkRift | Alright, I'll have a look, thank you |
17:05.13 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
17:06.12 | *** part/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
17:06.18 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
17:06.36 | [TK]D-Fender | lowtek: boingBOING |
17:07.33 | lowtek | AT&T sucks. |
17:07.42 | thehar | indeed |
17:07.48 | Shizuo | Just like GS |
17:07.57 | Shizuo | At least that's what the local bible says |
17:09.03 | NovceGuru | blah voicepulse can't port me :\ |
17:10.19 | NovceGuru | I have one POTS line that I am keeping for fax, I wonder if I can have 2 numbers attached to it, but only foward one of them |
17:10.29 | NovceGuru | I doubt it |
17:10.41 | [TK]D-Fender | NovceGuru: Have any providers been able to offer to? |
17:11.17 | NovceGuru | I don't believe so, but possibly. I know sometimes their website says they can't but they actually can |
17:11.57 | [TK]D-Fender | NovceGuru: Dirty trick, but ask Vonage. If they say yes, then you can do a 2-stage swap to swpie it from them once they've got it. |
17:12.11 | [TK]D-Fender | NovceGuru: Sometimes you just need to take it out of the first guy's grip |
17:12.24 | NovceGuru | yeah, I think p8 was able to |
17:12.54 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
17:12.59 | pta200 | Any idea when Asterisk 1.4.23 will be officially released |
17:13.21 | NovceGuru | er, they can't, shooooooooot |
17:13.23 | [TK]D-Fender | pta200: Never |
17:13.32 | pta200 | riiiiiiiiiight |
17:13.38 | NovceGuru | although time warner is able to, but I don't want to use their shitty analog lines |
17:13.50 | [TK]D-Fender | ptathis is an OSS project. they don't have deadlines and this kind of question will never get a solid date handed out. |
17:14.20 | [TK]D-Fender | pta200: and that wasn't "never" as in "will never be released", just "Never going to get an answer" |
17:14.34 | NovceGuru | voange can't also...hell |
17:14.44 | [TK]D-Fender | pta200: Know how long people were asking "when is 1.6 noon RC going to be released"? MONTHS |
17:14.53 | pta200 | I don't doubt it |
17:15.12 | pta200 | I just thought I would ask after reading through the change log |
17:15.20 | [TK]D-Fender | pta200: So don't hold you breath on an answer for 1.4.23 |
17:15.27 | pta200 | go it |
17:15.29 | pta200 | got it |
17:15.50 | [TK]D-Fender | pta200: if there is something particularly important to you in there upgrade to a newer SVN release |
17:16.40 | *** join/#asterisk murdock_ut (n=chatzill@64-42-64-98.atgi.net) |
17:17.33 | timeshell | Any inside news on the skype channel? |
17:17.47 | [TK]D-Fender | timeshell: Nope |
17:17.55 | [TK]D-Fender | timeshell: Still closed beta |
17:18.04 | NovceGuru | ohhh broadvoice can xfer it! |
17:18.42 | *** join/#asterisk outtolunc (n=me@adsl-76-211-236-48.dsl.pltn13.sbcglobal.net) |
17:20.50 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
17:20.50 | *** join/#asterisk bbryant (n=Brett_Br@adsl-068-016-200-248.sip.chs.bellsouth.net) |
17:23.20 | carranca | Hi, i have a weird scenario for you, i have an asterisk 1.4.22 with libpri and dahdi installed, i have a Digium TE205P with 2 Pri connected with a crossed cable (loopback) and the following configuration http://pastebin.com/m32269f20. As you can see i have 2 sip phones (402 and 403) and if i make a call with the prefix 201 I use one of the spans and with 202 the other span (continues).... |
17:24.08 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
17:24.12 | *** join/#asterisk qdk (n=qdk@79.138.230.80.bredband.3.dk) |
17:24.19 | carranca | both isdn have different context, if i call 2011234 i hear 1,2,3,4 and hangup... if i call 2021234 i hear the monkeys.... so far so good |
17:25.12 | verywiseman | can i use Grandstream FXS/FXO ATA with asterisk? |
17:25.37 | carranca | but if i call 201402 (i should be able to talk to my other SIP phone aka gateway) the phone rings but when i answer i get no sound |
17:25.42 | lowtek | verywiseman: Yes, but it will probably blow up or let notify child molestors of where your children go to school. |
17:29.18 | murdock_ut | using 1.6 and one step parking, I am unable to repark a call. Any ideas? |
17:32.28 | verywiseman | lowtek, what is you mean? |
17:33.16 | lowtek | verywiseman: Nothing, lol, I just don't like Grandstream. |
17:33.34 | verywiseman | why? |
17:34.08 | [TK]D-Fender | carranca: where is SIP/402 relative to your * server? |
17:34.46 | NovceGuru | hmm, so to have multiple lines with broadvoice...you have to have an account for each line |
17:35.01 | [TK]D-Fender | NovceGuru: Remember what I suggested : 2 stage porting |
17:35.24 | verywiseman | lowtek, why? |
17:35.25 | carranca | [TK]D-Fender, what do you mean? SIP/402 register itself like a regular SIP phone |
17:35.26 | [TK]D-Fender | NovceGuru: I never said you should jsut leave it in the first place that can grab it |
17:35.40 | NovceGuru | yeah, I called voicepulse about porting from braodvoice and they said no :\ |
17:35.50 | [TK]D-Fender | carranca: It is a NETWORKING question. Describe the path between * and your phone |
17:35.54 | NovceGuru | Seems silly |
17:36.00 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
17:36.03 | [TK]D-Fender | NovceGuru: This is just so much BS... |
17:36.06 | NovceGuru | once it's off the POTS it should be able to go anywhere? </noob> |
17:36.18 | NovceGuru | I know! |
17:36.33 | [TK]D-Fender | NovceGuru: This shouldn't be any kind of issue and they are just playing games with you... |
17:37.02 | NovceGuru | they said since they dont have a rate center in my area even if broadvoice has my number there's nothing they can do |
17:37.15 | [TK]D-Fender | NovceGuru: IIRC number portability is supposed to be ensured... the enforced part is another matter however. |
17:37.40 | [TK]D-Fender | NovceGuru: telcos usually get away with being assholes in this regard |
17:37.47 | NovceGuru | yeah :\ |
17:37.56 | NovceGuru | I've had issues with every port i've done |
17:38.11 | NovceGuru | last one, the telco keep reactivating the number |
17:38.22 | [TK]D-Fender | NovceGuru: FUN... |
17:38.33 | NovceGuru | yeah |
17:38.51 | carranca | [TK]D-Fender, well, i connect myself to the company switch with a physical SIP phone (Zultis). That is the SIP/402, my machine is also connected to the phone (so the phone is also a switch) and in there i have a SIP/403 softphone. Asterisk is in the server room connected to the company switch |
17:38.52 | NovceGuru | broadvoice accounts come with 2 channels so I *guess* I'd only need 2 accounts |
17:39.10 | NovceGuru | or I could keep the main number with the local telco and drop long distance and just use it for incoming |
17:39.29 | [TK]D-Fender | carranca: And your other phone? |
17:39.30 | NovceGuru | and not go with 4 "unlimited" lines with time warner and STILL probably save money |
17:39.57 | carranca | [TK]D-Fender, thats it, i have two, a physical one (402) and a soft phone 403 |
17:40.04 | [TK]D-Fender | NovceGuru: Assuming your telco doesn't try to rape BV as well |
17:40.12 | [TK]D-Fender | carranca: Both local LAN? |
17:40.21 | carranca | [TK]D-Fender, yes |
17:40.37 | [TK]D-Fender | carraAnd the 2 can call each other just fine through *? |
17:40.46 | NovceGuru | [TK]D-Fender: how could they do that? |
17:41.09 | carranca | [TK]D-Fender, have you read my whole question? the problem is not connecting the phones themself, that was the easy part, the problem is passing throug dahdi in a loopback |
17:41.10 | [TK]D-Fender | NovceGuru: Same way you said they kept reactivating that other number |
17:41.24 | thedonvaughn | Hey, is there a way to turn off hold music for queues and just have the caller listen to ringing? |
17:41.38 | [TK]D-Fender | carranca: Yes, but I want to rule out that your endpoints don't have any possible issues themselves. |
17:41.46 | [TK]D-Fender | carranca: you need to validate the whole path |
17:41.53 | NovceGuru | well in theory we'd be "staying with" the local telco for the main number and fax, then having an account with voicepulse for rollover and long distance outbound |
17:41.59 | [TK]D-Fender | carranca: And the enpoints are the first thing you prove |
17:42.06 | carranca | [TK]D-Fender, from the phone 403 if i call 402 its all good, but if i call 201402 the phone rings but i have no sound |
17:42.56 | [TK]D-Fender | carranca: through your loopback, go use another function like voicemail and see if it records,e tc |
17:42.59 | [TK]D-Fender | carranca: set up a small IVR and test DTMF, etc |
17:43.30 | carranca | [TK]D-Fender, ive tested a playback, thats 2011234 and it works correctly |
17:43.37 | *** join/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk) |
17:43.38 | carranca | [TK]D-Fender, thats the weird part |
17:43.47 | [TK]D-Fender | carraNot jsut playback, recording <- |
17:44.16 | carranca | [TK]D-Fender, i will try, give me a min ;) |
17:51.41 | *** join/#asterisk carranca (n=carranca@pampero.itba.edu.ar) |
17:54.19 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
17:56.37 | riddlebox | I dont think there is a ITSP that will ever port my number |
17:57.09 | carranca | [TK]D-Fender, recording and playback worked ok |
17:57.26 | lowtek | riddlebox: Who have you tried? |
17:58.13 | riddlebox | lowtek: everyone in the ITSP list from jbot |
17:58.31 | riddlebox | I even tried vonage just to see and they wont port it either |
17:58.56 | riddlebox | flexpulse |
17:59.18 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
17:59.35 | rene- | is there a way to increase the rx volume on an IAX trunk? |
17:59.42 | rene- | hello all |
18:00.22 | carranca | [TK]D-Fender, this is the new conf http://pastebin.com/ma7327bf |
18:00.44 | lowtek | riddlebox: I work for FlexPulse, we can usually port anything. Odd. |
18:01.59 | riddlebox | I msg'd the number to you maybe you can check but the site says no |
18:03.15 | *** join/#asterisk sasargen (n=chatzill@174-152-144-174.pools.spcsdns.net) |
18:03.16 | hardwire | USAC sent us a bill for 2 cents |
18:03.29 | lowtek | riddlebox: Nope, we can't either. Must be some off the beaten path clec. |
18:03.30 | hardwire | they spent like $0.50 just to send us the bill. |
18:03.40 | lowtek | riddlebox: Who's your phone company? |
18:03.41 | riddlebox | it is a number from Charter Communications |
18:03.52 | carranca | [TK]D-Fender, also tried voicemail and it works fine |
18:04.01 | lowtek | Do you know what rate center your area code is in? |
18:04.11 | riddlebox | nope |
18:04.48 | carranca | [TK]D-Fender, i think it has something to do with redirection |
18:05.06 | LemensTS | Hey is the guy who posted the flash phone in here from last night |
18:05.07 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
18:05.20 | Shizuo | Flash phone? |
18:05.21 | Shizuo | What? |
18:05.38 | LemensTS | Yea someone posted a link in here last night of one they made. |
18:05.50 | Shizuo | Was it ok? |
18:06.03 | LemensTS | It looked nice |
18:06.20 | Shizuo | Nice... |
18:06.34 | Shizuo | Does it work with my nice GS gateways? |
18:06.36 | LemensTS | Had a job for them if they wanted it. |
18:06.45 | LemensTS | ~gs |
18:06.46 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
18:07.00 | Shizuo | GS works for me |
18:07.04 | LemensTS | heh sorry lol |
18:07.05 | Shizuo | Never had a problem |
18:07.11 | Shizuo | Never used it asterisk, though |
18:07.27 | LemensTS | Im not sure I never used gs gateways. |
18:07.34 | Shizuo | They're nice |
18:07.47 | Shizuo | Their hybrid circuit outperforms most Cisco stuff I have |
18:07.57 | LemensTS | Whats it do |
18:08.18 | Shizuo | "The hybrid" is the circuit responsible for avoiding echoes at the interface |
18:08.36 | Shizuo | When you listen to the line, you'll also listen to whatever you're injecting at it |
18:08.51 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-ee8e9f46d0ff5b07) |
18:08.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
18:08.55 | Shizuo | So POTS handsets and equipments need to contain a "hybrid" |
18:09.10 | LemensTS | oh is it a big ata |
18:09.29 | Shizuo | It subtracts outcoming speech from the incoming data |
18:11.04 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
18:11.08 | Shizuo | LemensTS: Most cisco stuff lack an hybrid |
18:11.18 | Shizuo | So it needs to be performed in software |
18:11.24 | Shizuo | Which is not nice |
18:13.20 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-196-172-rrdg-esr-2.dynamic.isadsl.co.za) |
18:14.20 | maddog01 | having problem with call files. i create a file move it to /var/spool/asterisk/outgoing and asterisk dosn't process it. any ideas?? |
18:16.29 | *** join/#asterisk Assid (n=kvirc@unaffiliated/assid) |
18:16.42 | Alric | maddog01: Are you connected to the console when you move that file into the outgoing directory? |
18:16.42 | Assid | stupid machine |
18:16.59 | maddog01 | no |
18:17.15 | murdock_ut | murdock |
18:17.20 | maddog01 | not the asterisk console |
18:17.31 | Alric | maddog01: Try that, I believe it will show an error/notice if its rejecting the call file for some reason. |
18:17.42 | murdock_ut | wow, I can type my name. I'm so proud of myself. |
18:17.55 | maddog01 | thanks |
18:18.56 | coppice | almost all modern hybrids are implemented in software, like a short echo canceller. even FXO/FXS chips like the ones used on the digium cards, which appear to implement that simple DSP in hardware and probably doing it in small embedded processor |
18:20.00 | Assid | wants to move his desktp to a mac.. and soon |
18:21.04 | *** part/#asterisk viraptor (n=viraptor@87-194-164-154.bethere.co.uk) |
18:21.32 | Qwell | murdock_ut: hey, that's like 400 points on the SAT |
18:21.45 | *** join/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com) |
18:22.09 | murdock_ut | Qwell: Hey I can use all of the help I can take. Do you have any ideas on my one step parking issue with 1.6? |
18:23.00 | Assid | anyone wanna trade my core2quad 6600 - for a mac pro |
18:24.13 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
18:25.11 | *** join/#asterisk dieguito84 (n=diego@host118-190-dynamic.12-79-r.retail.telecomitalia.it) |
18:25.24 | Alric | Assid: How about a Mac Pro gutted and re-filled with PC components, running Vista? ;) |
18:25.56 | Assid | get away from me |
18:26.04 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
18:26.11 | tzafrir_laptop | mac pro can run Debian just fine. Though there are simpler alternatives |
18:27.17 | Assid | tzafrir_laptop: im just tired man.. really freaking tired.. it crashed on me right now.. i had soo many open browsers.. it just froze |
18:27.30 | Alric | Assid: You ever see the pics of the person that did that to a PowerMac when they first came out? |
18:27.33 | Assid | was updating a cisco 7960 phne |
18:27.43 | tzafrir_laptop | Assid, it == ? |
18:27.46 | Assid | Alric: no.. and honestly i wouldnt want to |
18:27.51 | Assid | tzafrir_laptop: xp :| |
18:28.12 | iEatChildren | does [DID_span_1] specify where my calls go when they come in? |
18:28.12 | Assid | in the middle of someone helping me.. getting this phone to work |
18:28.16 | Assid | can you imagine that.. |
18:28.19 | Assid | embarassing |
18:28.43 | Alric | Did it save your tabs at least? |
18:29.01 | Assid | Alric: npe.. hung.. nothing moved |
18:29.15 | Alric | Yuck |
18:29.24 | Assid | imagine if i dont save my codes everytime i edit them, or a db task |
18:29.32 | Assid | or even a CC transaction.. |
18:29.47 | *** join/#asterisk kotique (n=picachu@78.129.232.75) |
18:29.48 | Assid | sorry.. .im just annoyed |
18:30.06 | kotique | hey guys. How do I use C - like stares cases ? - |
18:30.11 | kotique | case _XXX: |
18:30.17 | kotique | case _XXXX: |
18:30.32 | kotique | playback(); |
18:30.38 | kotique | I'm talking about AEL. |
18:30.59 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
18:31.10 | LemensTS | <PROTECTED> |
18:31.12 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
18:31.20 | kotique | *stairs, not stares :) |
18:31.39 | Corydon76-dig | LemensTS: probably -addons not installed |
18:32.55 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
18:33.53 | maddog01 | Alric: i tried that it dosn't show anything in the console when i move the call file. |
18:35.29 | [TK]D-Fender | maddog01: check the TIMESTAMP |
18:35.46 | maddog01 | it's in the past i checked |
18:35.59 | maddog01 | and i checked the permissions |
18:36.14 | [TK]D-Fender | maddog01: remove the file, max out your debug and move the file back. |
18:37.20 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
18:37.47 | LemensTS | Corydon: do i need to restart asterisk after installing asterisk-addons |
18:38.11 | errr | I think so |
18:38.16 | [TK]D-Fender | LemensTS: Should be able to load the module live |
18:39.16 | LemensTS | Yep got it working now. thanks. |
18:39.25 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
18:39.27 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
18:40.10 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
18:41.26 | iEatChildren | can anyone tell me whats wrong here? exten = _XXXX,1,Goto(sip/2002|2002|1) |
18:41.32 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
18:41.36 | iEatChildren | keeps telling me invalid extention 2002 in the CLI |
18:42.11 | kannan | goto,context,priority,data |
18:42.13 | Qwell | sip/2002 isn't a context |
18:42.26 | kannan | sorry |
18:42.29 | loather | you probably want Dial(SIP/2002) instead |
18:42.34 | iEatChildren | ill try that |
18:42.38 | kannan | goto,context,exten,priority |
18:42.46 | iEatChildren | what happened is a user deleted something via gui and our asterisk guy is out of town |
18:42.48 | maddog01 | [TK]D-Fender: nothing again. I also check to see if the module is loaded and it dose show up when i use the show modules command |
18:43.20 | [TK]D-Fender | maddog01: Can be permissions, bad file, etc. Please pastebin EVERYTHING |
18:43.31 | loather | That exten line will also match any four-digit pattern, so if someone dials 2956 or 0987 it'll go to that extension as well |
18:43.40 | iEatChildren | so something like exten = _XXXX,1,Dial(SIP/2002) ? |
18:43.44 | loather | ... which might not be what you want |
18:44.11 | iEatChildren | its fine for now, im just putting on a band-aid fix untill i have time to redo the voice menu |
18:44.23 | loather | you also want => instead of = |
18:44.44 | iEatChildren | ok, ill try those things. thank you! |
18:44.49 | loather | welcome |
18:45.00 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
18:46.31 | *** join/#asterisk soa2ii (n=soa2ii@i59F56CE2.versanet.de) |
18:47.18 | *** part/#asterisk Shizuo (i=pato@200-171-49-211.dsl.telesp.net.br) |
18:47.34 | soa2ii | Hi there. I'm looking for people that are familiar with ISDN... maybe from germany so they coild tell me if this setup http://www.lagom.de/misc/setup.png would work (: |
18:49.28 | soa2ii | The VersatelBox is this here: http://www.festnetz24.com/images/versatelboxkonfiguration.jpg combined isdn splitter with two rj11, telephone, fax and rj45 for your internet. |
18:50.23 | soa2ii | So I simply want to plug my server into the second S0-Bus and make it a VoIP-ISDN-Gateway. |
18:51.28 | iEatChildren | loather - how would i make that transfer to a voice menu instead of a sip ext? |
18:51.34 | verywiseman | how connect analog phone to Rhino R24FXS PCI card? |
18:53.33 | [TK]D-Fender | verywiseman: with an RJ-21 patch panel or break-out box |
18:53.47 | [TK]D-Fender | verywiseman: And that is a fugly solution for FXS. |
18:54.31 | iEatChildren | nevermind loather...i got it woot woot! |
18:54.37 | iEatChildren | thanks again for your help earlier though |
19:06.20 | LemensTS | all you can do with the MYSQL cmd in the dialplan is query? no insert? |
19:06.55 | [TK]D-Fender | LemensTS: Yes, you can |
19:07.01 | murdock_ut | [TK]D-Fender: Do you use one step parking? |
19:07.19 | [TK]D-Fender | murdock_ut: Nope. |
19:07.32 | LemensTS | even in 1.2 |
19:08.32 | *** join/#asterisk ta^3 (n=tacvbo@189.146.186.167) |
19:09.49 | [TK]D-Fender | LemensTS: http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL |
19:11.33 | *** part/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net) |
19:11.43 | *** join/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net) |
19:12.28 | LemensTS | TK: yea thats what i was looking at, i only seen queries. then i get this in my cli: |
19:12.29 | LemensTS | -- Executing MYSQL("SIP/12345678-099e84d8", "INSERT questions SET questionnum=1, pollid=22") in new stack |
19:12.30 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
19:12.38 | LemensTS | Jan 7 14:09:58 WARNING[10338]: app_addon_sql_mysql.c:411 MYSQL_exec: Unknown argument to MYSQL application : INSERT questions SET questionnum=1, pollid=22 |
19:12.56 | LemensTS | I run that cmd in mysql just fine. |
19:13.47 | lowtek | LemensTS: post your MySQL() line from your dialplan |
19:14.55 | LemensTS | exten => s,n,MYSQL(INSERT questions SET questionnum=${questionnum}\, pollid=${pollid}) |
19:15.05 | LemensTS | maybe i dont have enough \ in there |
19:15.07 | lowtek | LemmensTS: You're missing some stuff. |
19:15.57 | lowtek | LemensTS: Should be somethign like: MySQL(Query resultid ${connid} INSERT\ questions\ SET\ questionnum=${questionnum}\, pollid=${pollid}) |
19:16.36 | lowtek | LemensTS: afaik you have to specify "Query" follwed by the resultid (even if there isn't a result) and the connection id you've opened in a previous MySQL() "connect" statement. |
19:17.57 | [TK]D-Fender | LemensTS: INSERT into table set field=value |
19:18.09 | *** join/#asterisk reneger (n=reneger@dslb-088-078-123-090.pools.arcor-ip.net) |
19:18.15 | [TK]D-Fender | LemensTS: improper base SQL syntax and improper escaping and MySQL() app syntax |
19:18.22 | [TK]D-Fender | LemensTS: Go read that page again |
19:19.53 | sasargen | does anyone know how to configure a Mitel 5302 ip phone to register with asterisk? |
19:22.55 | LemensTS | Thanks that worked. Confused myself. One last question, in php after i insert a reccord that has an auto_incrementing id, i can use mysql_insert_id() to find what that id was that it last inserted. Do you know of a way of doing this in the dial plan after i insert it? |
19:23.29 | lowtek | LemensTS: insert the uniqueid channel variable and use that. |
19:24.36 | maddog01 | trying to get call files working. I move the file to the outgoing folder and nothing happens. I have put a bunch of folder dumps / CLI info here (http://pastebin.com/d147ab9a3) can someone help me. |
19:25.49 | lowtek | maddog01: check your permissions on the file created, change them to something asterisk has permissions to r+w+x to before copying. |
19:25.53 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
19:25.56 | lowtek | maddog01: er.. moving, rather |
19:26.49 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
19:26.55 | maddog01 | i checked permissions there set to read asterisk:asterisk |
19:27.19 | lowtek | maddog01: Asterisk has to have permission to remove the file after it parses it. |
19:27.26 | StephenF | IM quoting a small office (16 phone, 5 POTS line) asterisk system what is everyone's favorite server hardware for this type of confg |
19:27.27 | jaytee | file: are you here? |
19:28.13 | rob0 | Steve, never skimp on hardware. |
19:28.48 | [TK]D-Fender | maddog01: * needs to be able to WRITE. Give ti FULL perms |
19:28.57 | maddog01 | k |
19:29.06 | StephenF | Thats my point what works well with asterisk, Dell, HP, etc... |
19:29.31 | Corydon76-dig | It's lmadsen's birthday, today. Everybody wish him a happy birthday! |
19:29.46 | rob0 | Name brands? Uh ... I don't buy those. I build my own. |
19:30.01 | StephenF | Also they currenty have what looks like an integrated T1 with voice and data. There is a box that splits the T1 into data and simulated POTS lines. What is that called? Would that be a PRI? |
19:30.05 | jaytee | I've got Asterisk running on a Dell PowerEdge 2950 with 4GB of RAM using a Digium TE212P dual port T1 card and 53 sip phones. it never breaks a sweat. |
19:30.16 | rob0 | Assemblers like Dell & HP are going to cut corners on things. |
19:30.37 | file | jaytee: moo |
19:30.38 | jaytee | StephenF, the "box" is called a T1 add/drop mux |
19:30.43 | StephenF | ahh ok |
19:30.45 | jaytee | file, quickie question |
19:31.04 | jaytee | I've got my main grammar file here: http://www.pastebin.ca/1302924 |
19:31.46 | jaytee | when I use speech the response is quick, when I use dtmf there is about a 4 to 5 second lag before the SpeechBackground app executes the option |
19:32.05 | StephenF | so if the customer already has that mux, I would need to spec an analog digium card like the AEX808E |
19:32.21 | maddog01 | [TK]D-Fender: asterisk dose have write permissions |
19:32.23 | StephenF | rob0 I see your point |
19:32.46 | file | jaytee: set the SPEECH_DTMF_MAXLEN dialplan variable to the maximum length of DTMF you want and it'll return immediately once it reaches it |
19:33.07 | rob0 | Note, I've never bought their server-grade hardware, so it might be worth more $$$ to save some time. |
19:33.39 | StephenF | rob0, thats kinda what I'm thinking. Plus the warranty support is helpful to maintain the device |
19:33.48 | jaytee | file, maximum length of DTMF? is that set in the grammar or the lumenvox.conf file? I'm confused. |
19:33.55 | file | jaytee: dialplan |
19:34.12 | jaytee | ah, it's a system global var? |
19:34.15 | file | jaytee: Set(SPEECH_DTMF_MAXLEN=2) before calling SpeechBackground, type in 2 DTMF digits and boom it'll return immediately |
19:34.23 | file | it's a dialplan variable |
19:34.24 | file | nothing special |
19:34.31 | rob0 | well, if you need warranty service, you (your customer) is screwed. They can't do anything with a Linux system. |
19:34.48 | jaytee | file, cool, I'll give that a shot. thanks!!! |
19:35.19 | StephenF | jaytee, So with that MUX in place I would need to spec an analog telephony card such as the AEX808E correct? |
19:36.03 | [TK]D-Fender | StephenF: Or jsut get a digital card and replace their MUX & data hardware and connect directly to the smartjack |
19:37.55 | StephenF | [TK]D-Fender right, Im not sure their provider would be happy with that. But anyway if I did what would their data network connect to then? If the smartjack connected to the digital tel card, where would the data connect? |
19:38.15 | [TK]D-Fender | StephenF: Your * box |
19:38.18 | StephenF | Do I use a 2 port card, one for smartjack connectivity, and the other for data network |
19:38.21 | StephenF | ahh |
19:38.27 | [TK]D-Fender | StephenF: UNIX has a long history of networking you know... |
19:38.30 | StephenF | lol |
19:38.50 | StephenF | ok, so I just need a 1 port card, and then * box connects to data network over standard NIC |
19:38.57 | StephenF | duh |
19:38.58 | eppigy | UNIX for networking? |
19:39.03 | eppigy | prepostrous |
19:39.18 | StephenF | who knew?! |
19:39.33 | StephenF | so in a sense I would be replacing the Mux with an * box |
19:40.04 | StephenF | very cool |
19:40.54 | jaytee | [TK]D-Fender, so with your proposal you're having him use a T1 card in * and zaptel to do the demux by defining the b channels for voice in one group and the data group as another? |
19:42.17 | [TK]D-Fender | jaytee: Individual timeslots, yes. However I'm pretty sure it isn't PRI, just FXSLS & data |
19:42.40 | [TK]D-Fender | jaytee: Whicht he card can bod off as an ethernet device and route accordingly. |
19:42.51 | [TK]D-Fender | bond* |
19:43.11 | jaytee | yeah, that's kinda where I figured you were going with that. smart choice and efficient as hell |
19:43.38 | [TK]D-Fender | jaytee: eggs in a basket as well but yes, efficient |
19:43.56 | [TK]D-Fender | jaytee: jaytee And better o the telephony side. |
19:44.18 | jaytee | don't know what brand of equipment their using but the Kentrox Add/Drop muxes are pricey as hell compared to a 2 port digium card. |
19:44.42 | StephenF | Mux is existing, provider is cbeyond |
19:45.10 | StephenF | though once their contract is up they will be dropping cbeyond |
19:45.47 | StephenF | So i think I will recommend the analog card since once they drop cbeyond they will be using us as their data provider and I doubt will want to pay for an PRI when they are only using 5 lines |
19:46.17 | [TK]D-Fender | StephenF: partial PRI w/ data is a great option. |
19:46.32 | [TK]D-Fender | StephenF: rAnalog sucks |
19:46.43 | [TK]D-Fender | StephenF: Mixing voice & data is cost-effective |
19:47.06 | StephenF | definetly, but this is kind of a weird situation. We are an ISP and they are moving into our building. So we can provide data with no local loop. Just a cat5 down to their suite |
19:47.39 | [TK]D-Fender | StephenF: If you're an ISP why not just go for an ITSP? |
19:47.51 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
19:48.07 | StephenF | Im contemplating that as well. We could do that |
19:48.17 | StephenF | Just not sure we are ready to take that on |
19:48.51 | [TK]D-Fender | StephenF: Levy againsth the cost of any card. then again against the woes of analog |
19:49.10 | timeshell | I found this useful to make oslec work with dahdi: http://www.freepbx.org/forum/freepbx/installation/asterisk-1-6-dahdi-oslec |
19:49.20 | jaytee | file, worked like a champ!!!! no more lag. where do i find this stuff documented? I've been using sample apps from lumenvox and just trying to make things work. |
19:49.49 | StephenF | I've never done an analog * box before, is configuration that much worse? Or are you talking about quality of analog lines? |
19:49.58 | LemensTS | lowtek: this works to find the last auto_incrementing id - exten => s,n,MYSQL(Query resultid ${connid} SELECT\ last_insert_id()) |
19:50.19 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
19:50.26 | cvnet | hello |
19:50.43 | lowtek | LemensTS: Assuming that another record from another call isn't inserted between the time you do your insert and the query. |
19:50.45 | *** join/#asterisk adminguru (n=atze@p57BD7B49.dip.t-dialin.net) |
19:50.47 | [TK]D-Fender | StephenF: Qualify, hassle, call setup delay, lack of DID functioning, line contention, etc. |
19:50.50 | lowtek | LemensTS: So it won't *really* work. |
19:52.15 | StephenF | [TK]D-Fender so what kind of ITSP setup would work best? |
19:52.27 | *** join/#asterisk km2 (n=x@68.161.155.223) |
19:52.27 | lowtek | LemensTS: Unless, of course, it's specific to that connection, in which case it might work. I wouldn't trust it, just insert the uniqueid and you won't even have to make another query. |
19:52.37 | StephenF | hosting the *, providing a sip connection to our *, or something even more complex? |
19:53.15 | StephenF | or just have an extablished ITSP provide their PSTN termination? |
19:53.39 | *** join/#asterisk alrs (n=lars@cpe-76-174-43-54.socal.res.rr.com) |
19:53.41 | LemensTS | lowtek: as soon as it inserts, the next line is checks for last_insert_id(). And i think it is doing it based on the current mysql connection. |
19:54.04 | LemensTS | ill have to test and see if i can screw it up. |
19:54.36 | lowtek | LemensTS: Well you can always fall back to inserting ${UNIQUEID} if it doesn't, good luck! |
19:54.57 | [TK]D-Fender | StephenF: Well this is for YOU, right? |
19:55.07 | jaytee | file, never mind that last question. I just finally found the motherlode on lumevox's site. couldn't find much 8 months ago. |
19:55.18 | StephenF | [TK]D-Fender what do you mean for ME? |
19:55.46 | Greek-Boy | lol |
19:55.56 | [TK]D-Fender | StephenF: nvm. Just pick an ITSP and run your own * box locally. |
19:55.59 | LemensTS | Yea I will keep that in mind, i should be able to do that too. Only reason i want to use the table id is if they end up doing this thru php in a browser, then i wont be able to ${uniqueid} but i can reference the last id still. |
19:56.03 | StephenF | we have our own * box, and we are using another ITSP for PSTN termination |
19:56.38 | StephenF | you said if we were an ISP, why not gor for an ITSP. Did you mean set-up this customer with an ITSP? |
19:56.44 | *** join/#asterisk maddog01 (n=minotaur@d221-91-175.commercial.cgocable.net) |
19:57.00 | [TK]D-Fender | StephenF: First you're talking analog off a T1 voice circuit, then PRI, not you already have an ITSP. Please clarify this mess |
19:57.12 | StephenF | lol ok |
19:58.00 | StephenF | I have a customer that currently has a T1 add/drop mux with 5 voice channels and the rest for data |
19:58.03 | StephenF | we are specing an * box for them |
19:58.26 | StephenF | They will be dropping their current T1 provider when the contract is up, and then using us for data connectivity |
19:59.02 | StephenF | (we are an ISP, they are moving into our building. This allows us to provide data connectivity without a local loop fee) |
19:59.38 | StephenF | So I was saying I dont think I would want to replace the Mux with an * box, because once they change providers they will most likely just order 5 POTS lines |
20:00.21 | StephenF | Unrelated to this customer, we also have our own * box that uses an ITSP for PSTN termination |
20:00.26 | StephenF | and thats the story of the day |
20:00.33 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
20:00.47 | [TK]D-Fender | StephenF: if your data is separate then no, PRI isn't probably a great way to go. |
20:00.59 | StephenF | yea prbly too expensive right |
20:01.13 | [TK]D-Fender | StephenF: Have they considered going right over to VoIP? |
20:01.53 | StephenF | that is an option but I really want this to be rock solid and I'm a little weary of doing that |
20:02.09 | iratik | I'm getting "funny" dtmf issues (like not recognizing digits pressed accurately) when playing an audio file and waiting for input from the remote extension .... this is on outbound calls where the remote party being called is entering digits .... the docs seem to indicate that relaxdtmf is a primarily a setting for tweaking problems with SIP, however this is on a PRI on a digium T1 card ... any ideas? |
20:02.16 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
20:02.37 | [TK]D-Fender | iratik: it is primarily for Zap, not SIP |
20:02.53 | [TK]D-Fender | iratik: and IS the first option to look at. Then look at your gains |
20:03.19 | *** join/#asterisk cuco (n=diego@DSL217-132-187-5.bb.netvision.net.il) |
20:03.46 | iratik | so if i'm getting inaccurate results ... go ahead and set relaxdtmf=1 ? |
20:03.52 | cuco | tzafrir_laptop, ping |
20:04.17 | [TK]D-Fender | iratik: relaxdtmf=yes |
20:04.41 | [TK]D-Fender | iratik: Don't go using numbers for these kinds of fields. Sets a bad precedent. |
20:04.42 | StephenF | [TK]D-Fender So i shall brave the woes of Analog |
20:04.52 | [TK]D-Fender | StephenF :why not... |
20:04.56 | iratik | sounds like relaxdtmf .. would make dtmf trigger more easily ... but shouldn't it be more strict? |
20:05.10 | StephenF | Sounds good to me, we'll get it done |
20:05.26 | StephenF | Thx for listening to the noob by the way ;) |
20:05.40 | tzafrir_laptop | cuco, hi |
20:06.35 | cuco | tzafrir_laptop, hi, i have a nice game for you. wanna play? :) |
20:07.39 | tzafrir_laptop | depends how nice it is |
20:07.39 | cuco | tzafrir_laptop, pmsg? |
20:08.23 | *** join/#asterisk af_ (n=getsmart@88-149-240-130.dynamic.ngi.it) |
20:09.50 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
20:09.50 | *** mode/#asterisk [+o denon] by ChanServ |
20:12.38 | [TK]D-Fender | iratik: does RELAX sound STRICT to you? |
20:12.57 | iratik | no ... thats the point |
20:13.09 | iratik | well.. if it is getting inaccurate results... it needs to be more strict about dtmf tones... not more releaxed |
20:14.26 | verywiseman | [TK]D-Fender, are you prefer "Grandstream HandyTone HT-286 ATA" for using with Asterisk? |
20:14.27 | eppigy | RELAXEDFITDTMF |
20:15.10 | eppigy | iratik: usually the results are innacyrate because it is not interpreting certain keypresses as valid dtmf tones |
20:15.15 | [TK]D-Fender | verywiseman: No. |
20:15.16 | eppigy | therefore you want its interpreation |
20:15.20 | eppigy | to be relxed |
20:15.24 | eppigy | relaxed |
20:15.38 | verywiseman | [TK]D-Fender, can u give me reasons ,pls? |
20:15.50 | iratik | well.. its interpreting them either inaccurately or not at all ... let me try it |
20:15.52 | [TK]D-Fender | verywiseman: History of being flakey shit. |
20:16.13 | eppigy | iratik: well there you go |
20:16.18 | eppigy | what i said verbatim |
20:17.05 | StephenF | any problems with asterisk and AMD processors? |
20:17.38 | verywiseman | [TK]D-Fender, what about sangoma PCI card? |
20:18.03 | [TK]D-Fender | verywiseman: before you go running around like a headless chicken, describe your precise NEEDS. |
20:19.14 | iratik | When using MixMonitor, then DTMF tones the caller pressed aren't heard in the recordings... is there anyway to make it so you can hear them? |
20:22.30 | *** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr) |
20:22.55 | verywiseman | [TK]D-Fender, i want to find suitable hardware for SOHO,medium business and large business |
20:24.05 | seb- | can Skype uses talk to Asterisk SIP servers? ...or are they locked out? |
20:24.07 | [TK]D-Fender | verywiseman: perhaps you do not understand the word PRECISE |
20:24.13 | seb- | users* |
20:24.20 | [TK]D-Fender | seb-: 3rd party only stuff now, all sucky |
20:24.24 | jjshoe | seb- closed beta going on right now for a connector |
20:24.25 | eppigy | TRABAJO |
20:24.37 | verywiseman | [TK]D-Fender, maybe , sory E is not my native language |
20:24.48 | jjshoe | skype had their own pbx on their website, for a few weeks, dunno whatever happened to it |
20:24.56 | [TK]D-Fender | verywiseman: DETAILS |
20:24.58 | seb- | jjshoe: what is cheapest/easiest way for people to connect to Asterisk? I was guessing Skype |
20:25.04 | Assid | okay so i managed to get a cisco 7960 on SIP, however i dont see it registering to asterisk.. sip debug doesnt show me a connection.. anyone here played with cisco+sip? |
20:25.11 | eppigy | seb-: xlite |
20:25.36 | jjshoe | seb- what eppigy said |
20:25.37 | jjshoe | http://www.prettymay.net/index.htm |
20:25.48 | seb- | eppigy: i believe that is a Windows app? then they just need to buy a plantronics headset and pray the docs are clear |
20:26.03 | khronos | <PROTECTED> |
20:26.06 | eppigy | they have windows and *nix |
20:26.45 | StephenF | what is everyone's opinion of Switchvox? any good? |
20:26.49 | seb- | eppigy: i'm doing an online calculus class..i hope the little students can figure out xlite set up then..thanks |
20:27.03 | jjshoe | StephenF why not download the free version and try it? |
20:27.11 | StephenF | I mean the hardware |
20:27.13 | jjshoe | seb- it's fairly straight forward |
20:27.28 | jjshoe | seb- you could of course ultimatly build an installer for it if you truly cared :P |
20:27.33 | [TK]D-Fender | StephenF: Its a PC. What is there to say? |
20:27.47 | StephenF | lol ok, has anyone used it? |
20:27.47 | seb- | jjshoe: in that case you've made my day!! buying a phone # from Teliax for conf calls was *butt* expensive |
20:28.03 | iratik | What is relaxdtmf set to by default? |
20:28.17 | StephenF | nvm way to expensive |
20:28.33 | seb- | jjshoe: yes! i could make a custom xlite that automatically connected to my SIP address! brilliant! |
20:28.42 | jjshoe | StephenF yeah, especially for a buisness phone system, cheap is always better! |
20:29.01 | StephenF | no but i can build a better dell box for half the price... |
20:29.04 | jjshoe | seb- or use one of many auto programs to install configs for it, or automate filling out, like autohotkey |
20:29.14 | [TK]D-Fender | ok, leaving early to go home. Later all. |
20:29.19 | StephenF | Cya |
20:29.28 | jjshoe | StephenF they provide you with more then just hardware. |
20:30.21 | verywiseman | [TK]D-Fender, i am now need fxs/fxo equipment from 1-250 fxs port and 1-48 fxo port , and there are 2 options (PCI card and ATA), and when i ask in this channel , somebody prefer ATA for small business |
20:30.27 | eppigy | seb-: if they hope to comprehend calculus im sure they will have no trouble with xlite setup |
20:32.11 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-970ebba76f676edd) |
20:32.30 | seb- | XLite is not open source...is there a quality open source SIP client? |
20:33.05 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
20:33.22 | LemensTS | if i have lots of s,n, lines, how can i send a GoTo(s,n) to a particular one? |
20:33.29 | LemensTS | ive always wondered that... |
20:33.49 | Alric | n ends up translating to a number. |
20:34.04 | LemensTS | oh reall?y so if its the 5th n, i can do GoTo(s,5) ? |
20:34.07 | Alric | Just manually count it out, or you can probably use labels or something. |
20:34.21 | LemensTS | awesome |
20:34.28 | jjshoe | seb- what does open source matter? where you hoping to modify the code? |
20:35.30 | mog | seb-, there is ekiga |
20:35.34 | Alric | LemensTS: I haven't bothered with labels, so I couldn't say if that really works, I've used the manual counting method in my Gotos. |
20:35.35 | mog | which is awesome |
20:36.29 | LemensTS | yea labels would help incase you insert a line in the middle of it |
20:36.50 | LemensTS | think they are s,n(labelhere), |
20:37.35 | *** join/#asterisk iceyp (n=icepick@60.234.68.250) |
20:37.38 | jjshoe | goto and labels.... awwwwwwwww. so cute! |
20:37.51 | iceyp | hey guys, does anyone know of a good free web based sip client that I can host on my web page |
20:38.06 | iceyp | I don't want a remote hosted product |
20:38.25 | LemensTS | jjshoe: yea... |
20:38.55 | *** join/#asterisk kerx (n=kerx@adsl-69-104-78-246.dsl.irvnca.pacbell.net) |
20:39.18 | kerx | Hi, is there any way to have SIPAddHeader() when you make a originate call from AMI? |
20:42.21 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
20:44.52 | Assid | i dropped my polycom handset once too many times.. it now has a screw loose inside or something |
20:45.34 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
20:50.00 | kerx | damn :-( |
20:50.08 | kerx | Polycom handset's are expensive |
20:50.21 | kerx | You have warranty on it? |
20:50.36 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
20:50.46 | Assid | nope :( |
20:50.53 | flujan | hello guys, i am creating a simple agi script to give it a try |
20:51.09 | kerx | To give what a try? |
20:51.10 | flujan | it needs to write in the stderr, so it will be redirected to asterisk |
20:51.13 | flujan | only it |
20:51.15 | flujan | a hello world |
20:51.19 | kerx | Oh, cool |
20:51.21 | flujan | but i am not having success |
20:51.39 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-b96d8a553644a429) |
20:53.04 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
20:54.13 | *** join/#asterisk sekil (n=sekil@80.93.247.26) |
20:57.03 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
20:57.06 | flujan | http://pastie.org/354993 |
20:57.10 | flujan | here goes the error message |
21:02.06 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
21:02.49 | iceyp | Anyone know of a web based sip client I can use on my webpage? |
21:04.16 | lowtek | iceyp: It would have to be java based or flash based so you could use sockets and access the computers mic. Probably better off just using a soft-phone. |
21:04.39 | iceyp | I'm looking for something to allow people to call me direct from our website |
21:04.46 | iceyp | not to make inbound / outbound calls |
21:04.56 | iceyp | just to dedicated numbers |
21:05.03 | lowtek | iceyp: Oh, so you want the ability to originate a call from a web page? Use php + an AMI call. |
21:05.25 | iceyp | umm, yeh possibly? |
21:05.36 | jjshoe | iceyp many of us do php + ami |
21:05.46 | jjshoe | "Enter your number and we'll call you!" |
21:06.07 | iceyp | i.e. people go to our contact us page, and click "Contact us now via VoIP" then when clicked, it enables their mic and speakers and they can talk to us |
21:06.11 | lowtek | iceyp: Yep, it's actually quite easy. PM me your email address and I'll give you a really good AMI wrapper I wrote a while back. |
21:06.50 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
21:06.55 | jasonwoot | woot |
21:06.57 | iceyp | brb ciggy |
21:07.05 | lowtek | iceyp: That's back to what I originally said. It would have to be java or flashed based and wouldn't work very well due to default security settings that prevent said practices on end-users computers. |
21:07.38 | lowtek | iceyp: Better off just initiating regular phone call. |
21:08.04 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
21:08.08 | Alric | Does the average user even have a mic on their computer statistically? |
21:08.21 | lowtek | Alric: I wouldn't think so, laptops maybe. |
21:08.22 | [TK]D-Fender | Alric: No. |
21:08.37 | Alric | That would have been my guess. |
21:09.03 | lowtek | iceyp: The problem taht will be the show stopper is that Java (not javascript) and Flash can't access the computer's resourcs or TCP/IP sockets without special permissions. |
21:10.05 | aiksa[LV] | lowtek: all of ar still waiting for pac.ifica ? |
21:10.16 | aiksa[LV] | all of us are * |
21:11.45 | iceyp | lowtek damn, it's just for family use, people that dont know how to install sip clients ;/ |
21:11.54 | lowtek | iceyp: skype? |
21:12.03 | iceyp | lowtek can you still email me your wrapper, would be good to take a look |
21:12.12 | iceyp | lowtek can skype dial sip:// urls? |
21:13.00 | jjshoe | iceyp no. |
21:13.01 | aiksa[LV] | iceyp: if only it could |
21:13.03 | aiksa[LV] | :P |
21:13.24 | aiksa[LV] | but there was a word a month or so ago that digium might have made deal with the devil |
21:13.46 | [TK]D-Fender | iceyp: Zoiper can be distributed pre-configured in a drop-in install mode |
21:14.05 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4) |
21:14.08 | [TK]D-Fender | iceyp: And supports IAX which can help in some cases |
21:14.18 | lowtek | iceyp: The short answer is that you're not going to be able to do browser based sip calls. Go to plan B, a softphone or maybe just an IM client if it's in-family stuff. Or Skype. |
21:14.21 | iceyp | [TK]D-Fender thanks let me take a look |
21:15.30 | iceyp | [TK]D-Fender the free version allows a pre-configuration ? |
21:15.47 | aiksa[LV] | iceyp: actually there is a config xml file |
21:15.54 | [TK]D-Fender | iceyp: Don't know all the fine points. Go look |
21:16.03 | aiksa[LV] | which contains everything that interface does and a little bit more |
21:16.29 | aiksa[LV] | the launch of the binary doesnt require the installation |
21:16.43 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
21:16.56 | Katty | woah |
21:17.01 | aiksa[LV] | but user will be prompted in windows to allow this application to access the network by default inbuilt windows firewall |
21:17.01 | iceyp | mmmm |
21:17.03 | Katty | [TK]D-Fender: sodium acetate sculptures |
21:17.25 | iceyp | aiksa[LV] you talk of zoiper? |
21:17.31 | lowtek | iceyp: In my family, I'd rather just pick up the cost vs trying to help them configure *anything* on their f*cked up computers and home networks. I just host a conference bridge and a toll-free for everyone to call in on. |
21:17.34 | aiksa[LV] | iEatChildren: yes |
21:17.55 | aiksa[LV] | iceyp: yes |
21:18.08 | iEatChildren | ty |
21:18.14 | iceyp | lowtek I have this in a number of countries though not all have toll free's, just trying to do the extra step :) |
21:18.16 | carranca | how can i hardcode dahdi to always make calls in ulaw? |
21:18.35 | aiksa[LV] | iceyp: additionaly - there was one russian comapy which did develop a prototype flash sip phone. flashphone.ru or something. |
21:19.06 | iceyp | aiksa[LV] so the xml is hosted on a web page, and the user just needs to input the XML URL? |
21:19.10 | [TK]D-Fender | Katty: Sodium tri-iodide > sodium acetate |
21:19.43 | aiksa[LV] | carranca: in /etc/zaptel.conf (or its dahdi.conf now): add this line alaw=25-86 |
21:19.45 | [TK]D-Fender | is Iron Chef of the Anarchist's Cookbook |
21:19.57 | aiksa[LV] | in your case that would be ulaw=chansfrom-chansto |
21:20.23 | aiksa[LV] | iceyp: no you just leave the ziped folder with all the files of an installed zoiper |
21:20.32 | aiksa[LV] | and the config file |
21:20.43 | *** join/#asterisk DarkRift (n=dark@65.92.169.136) |
21:20.47 | Katty | [TK]D-Fender: oh? does that also make dry ice? |
21:21.00 | iceyp | ahh and the user downloads the zip and then runs the binary out the zip |
21:21.07 | [TK]D-Fender | Katty: "Not Quite" |
21:21.12 | aiksa[LV] | back to a timing hacking |
21:21.20 | Katty | [TK]D-Fender: what's that do then? |
21:21.22 | carranca | aiksa[LV], you mean /etc/dahdi/system.conf or /etc/asterisk/chan_dahdi.conf? |
21:21.32 | [TK]D-Fender | Katty: GOOGLE :) |
21:21.35 | Katty | k |
21:22.01 | aiksa[LV] | carranca: the file which a while ago was named /etc/zaptel.conf not sure of the naming for the dahdi |
21:22.14 | aiksa[LV] | carranca: the one driver not asterisk related |
21:22.45 | Katty | [TK]D-Fender: is this just regular iodine? |
21:23.03 | carranca | aiksa[LV], ok :), its now /etc/dahdi/system.conf |
21:23.15 | Katty | [TK]D-Fender: not much documentation--wikipedia is not big on the practical application |
21:23.21 | [TK]D-Fender | Katty: No, a sediment-based contact explosive :) |
21:23.22 | aiksa[LV] | btw, [TK]D-Fender if a span in zttool is shown as "internally clocked" although has timing source priority of "1". Does that mean that telco doesnt provide the timing on this digital line at all? |
21:23.33 | Katty | [TK]D-Fender: ah, i see. |
21:24.02 | [TK]D-Fender | aiksa[LV]: I don't know zttool really |
21:24.04 | aiksa[LV] | sorry for HL you, but i thought you would probably now |
21:24.37 | aiksa[LV] | [TK]D-Fender: I have a suspicion that i dont get the timing from an E1 line and I am not sure how to check that |
21:24.39 | [TK]D-Fender | aiksa[LV]: pastebin your config |
21:24.53 | aiksa[LV] | a sec. |
21:24.55 | aiksa[LV] | ~pb |
21:24.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
21:24.59 | Sargun | Can someone translate the SIP tos_(audio/video/sip) fields in sip.con? |
21:25.03 | Sargun | Can someone translate the SIP tos_(audio/video/sip) fields in sip.conf? |
21:25.30 | FinboySlick | Sargun: To which language? |
21:27.14 | aiksa[LV] | [TK]D-Fender: http://pastebin.com/m4ec5db9e |
21:27.45 | aiksa[LV] | iceyp: exactly |
21:27.47 | Sargun | FinboySlick, nevermind, I found a document elsewhere. |
21:28.00 | [TK]D-Fender | aiksa[LV]: Indeed you are not taking timing from ANY of your ports |
21:29.06 | [TK]D-Fender | aiksa[LV]: span = PORT,TIMING,LBO. 0 = ACT as timing source. Bad for use with telcos |
21:29.27 | [TK]D-Fender | aiksa[LV]: 1= act as primary, 2,3,4, = secondary, tertiary, etc |
21:29.48 | [TK]D-Fender | aiksa[LV]: If all to telco : span=1,0,0,ccs,ami should be span=1,1,0,ccs,ami |
21:29.55 | aiksa[LV] | [TK]D-Fender: i want to take timing from that dynamic statement there |
21:29.56 | [TK]D-Fender | aiksa[LV]: span=2,2,0,ccs,ami |
21:30.07 | [TK]D-Fender | aiksa[LV]: as I've now shown you |
21:30.20 | *** join/#asterisk hakr (i=hakr@pdpc/supporter/active/hakr) |
21:30.23 | aiksa[LV] | [TK]D-Fender: all of the non dynamic spans ar not connected to telco but to a legacy pbx |
21:30.32 | aiksa[LV] | # |
21:30.33 | aiksa[LV] | dynamic=ethmf,eth1/00:50:c2:65:d3:92/0,31,1 |
21:30.33 | aiksa[LV] | # |
21:30.33 | aiksa[LV] | dynamic=ethmf,eth1/00:50:c2:65:d3:92/1,31,2 |
21:30.45 | [TK]D-Fender | aiksa[LV]: OH FFS, TDMEO |
21:30.51 | [TK]D-Fender | TDMOE |
21:30.54 | [TK]D-Fender | runs |
21:31.43 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
21:31.49 | aiksa[LV] | [TK]D-Fender: like run and hide? |
21:31.58 | aiksa[LV] | or like going to bed finally? |
21:32.13 | aiksa[LV] | explain: FFS ? |
21:32.48 | shinao1 | hi all... has anyone ever tried using the Cisco 3911 "SIP" phone with asterisk? |
21:33.15 | [TK]D-Fender | ~ffs |
21:33.16 | jbot | ffs is probably for f**k's sake, or for fine's sake. UCB's Fast File System |
21:33.39 | aiksa[LV] | i suspect the first .. |
21:34.19 | *** join/#asterisk Trionnis (n=rboggs@253.ptr.ifbyphone.com) |
21:35.47 | iratik | When using MixMonitor, then DTMF tones the caller pressed aren't heard in the recordings... is there anyway to make it so you can hear them? |
21:36.26 | aiksa[LV] | [TK]D-Fender: nevertheless is there a way to check if a span is being used as a timing source |
21:36.30 | aiksa[LV] | ? |
21:36.57 | [TK]D-Fender | aiksa[LV]: None of your fixed ones are, and I am not sure how TDMoE play into this |
21:38.53 | aiksa[LV] | [TK]D-Fender: fixed cannot be used as timing sources as they are not connected to telco |
21:39.01 | aiksa[LV] | only TDMoE is |
21:39.16 | aiksa[LV] | and according to documentation of ztd_ethmf |
21:39.39 | [TK]D-Fender | aiksa[LV]: As Isaid I have little to advise in configuring TDMoE. |
21:39.39 | aiksa[LV] | the last parameter in the "dynamic=" line refërs to timing priority |
21:39.55 | aiksa[LV] | [TK]D-Fender: ok. wont bother you then |
21:40.21 | aiksa[LV] | I just thought that there could be a tool which would show which span is used currently as timing source |
21:41.23 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-3f01ceaf57ff08af) |
21:43.12 | Trionnis | is anyone aware of a functional (with 1.4.21.2) backport of the bridging abilities? either app_bridge, or the built-in functions from 1.6? |
21:46.18 | Trionnis | the silence is deafening... ;-) |
21:46.31 | [TK]D-Fender | Trionnis: From what I remember having been mentioned about how they did it in 1.6 it took some major core changes the likes of which I don't believe couldbe back-ported.... |
21:47.01 | Trionnis | yeah, I kinda figured the 1.6 "version" would be a no-go |
21:47.08 | [TK]D-Fender | Trionnis: And yes, you wated a max of 3 minutes for an answer to an oddball question. Your patience is nothing less than ASTOUNDING |
21:47.25 | Trionnis | it is kinda saddening to see that the patch that enabled it back in 2005 didn't make it into 1.4 at any point |
21:48.13 | [TK]D-Fender | Trionnis: so what stops you from going to 1.6? |
21:48.34 | Sargun | Question: Does the RTP stream's source IP have to be the same as the SIP IP? |
21:48.38 | Trionnis | that it took 18 point releases of 1.4 to be considerable as production quality :) |
21:49.04 | jjshoe | Trionnis touche |
21:49.28 | aiksa[LV] | there is another "I want" feature of 1.6, I would really (I mean - REALLY) like to have in 1.4 |
21:49.38 | aiksa[LV] | Transfer event in AMI |
21:50.01 | aiksa[LV] | Trionnis: i suppose it could go in same changes bag with app_bridge |
21:50.39 | [TK]D-Fender | Sargun : Of course not. |
21:50.42 | aiksa[LV] | as the event model is totally redone as regards the old "Link" events. |
21:51.15 | *** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net) |
21:52.21 | Sargun | [TK]D-Fender, I assume in the SIP stream, the source is hinted at. |
21:52.58 | [TK]D-Fender | Sargun : the entire point of SIP is to setup RTP and and handle call flow |
21:54.05 | jasonwoot | I thought the entire point of SIP was to ruin my life |
21:54.29 | [TK]D-Fender | jasonwoot: no, thats in the "Other Cool Features" column |
21:54.31 | aiksa[LV] | jasonwoot: it's his alter ego |
21:54.37 | Trionnis | dunno about ruining my life, but it's certainly caused its fair share of migraines |
21:55.07 | *** part/#asterisk adminguru (n=atze@p57BD7B49.dip.t-dialin.net) |
21:55.38 | Sargun | [TK]D-Fender, Well, I was wondering if I can do layer 3 routing for SIP. |
21:55.42 | aiksa[LV] | Anyone fiddling with Sip is much like Oracle admins - they dont need wife to get guaranteed sexual life. |
21:55.59 | Sargun | [TK]D-Fender, Err, I mean RTP. So, I can send RTP via a different source IP. |
21:56.17 | SQLDarkly | My users keep getting dropped from my conferences.... I have looked at messages but I see nothing at the time of the drop. What could cause this? Its happened 4 times in 2 days |
21:58.18 | [TK]D-Fender | Sargun : Might be an idea... if it wasn't Layer 4 |
21:58.52 | Sargun | [TK]D-Fender, Well, I can set the ToS to SIP to N, and then route everything with the ToS value of N through line 2. |
22:00.04 | [TK]D-Fender | SQLDarkly: I'm sorry, could you be a little more vague please? |
22:00.15 | SQLDarkly | I can if you like ;) |
22:00.50 | SQLDarkly | My apologies. I am runnign 1.4.22 on 2 boxes. I have RT Arch setup and Meetme is pulling its rooms from a DB |
22:02.08 | SQLDarkly | Once the meetme is started people can join the conference but random people seem to drop |
22:02.24 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
22:02.30 | SQLDarkly | I do not see anything in the messages log as to the cause. |
22:02.42 | SQLDarkly | I was not monitoring the console during the drop |
22:03.04 | jasonwoot | SQLDarkly: happens to my meetme's constantly with zap callers, but not with SIP |
22:04.16 | SQLDarkly | There must be a cause.... |
22:05.19 | aiksa[LV] | SQLDarkly: not even anything in debug log? |
22:05.32 | aiksa[LV] | verbose sometimes is not verbose enough |
22:05.41 | aiksa[LV] | off for a smoke |
22:05.53 | *** join/#asterisk Trido (i=trido@ppp178-168.static.internode.on.net) |
22:13.47 | aiksa[LV] | back |
22:19.27 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
22:25.49 | Assid | does polycom 301 do hinting? |
22:26.40 | errr | I think so |
22:26.47 | [TK]D-Fender | AssiiViewable in Buddies, but not on line-keys |
22:27.27 | Trionnis | ok, slightly different track... anyone know if the Bridge application/manager command will work if you give it the Call-ID of each leg, or does it require the actual channel name? |
22:27.28 | *** part/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com) |
22:27.57 | verywiseman | [TK]D-Fender, i am now need fxs/fxo equipment from 1-250 fxs port and 1-48 fxo port , and there are 2 options (PCI card and ATA), and when i ask in this channel , somebody prefer ATA for small business |
22:28.05 | Assid | [TK]D-Fender: so same thing as a directory? |
22:28.41 | [TK]D-Fender | Trionnis: Bridge bridges CHANNELS. Call-id is a PROTOCOL-SPECIFIC thing. This has been abstracted |
22:29.11 | [TK]D-Fender | Assid: similar |
22:29.27 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
22:29.35 | [TK]D-Fender | verywiseman: AUdioCodes MP-124 <----- |
22:29.35 | Assid | theres more? |
22:29.49 | [TK]D-Fender | Assid: More what? |
22:30.13 | Assid | isnt the only difference between hinting and directory is online keys? |
22:31.06 | [TK]D-Fender | Assid: huh? Buddies screen just lists the contacts you have and their status |
22:31.23 | [TK]D-Fender | Assid: And I have no idea what you're getting at with "online keys" |
22:31.37 | Assid | like online/offline (registered/not registered) ? |
22:31.59 | Assid | "on-line" keys |
22:32.48 | [TK]D-Fender | Assid>"on-line" keys <- drop this term |
22:32.49 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-b2bceb54c5fa3021) |
22:33.04 | [TK]D-Fender | Assid: and you will see their STATUS |
22:33.29 | Assid | okay like if they are on call and such? |
22:33.43 | [TK]D-Fender | Assid: Yes |
22:34.05 | Assid | sweet.. i need to learn this.. but do OTHER buddies have to have polycoms? |
22:34.20 | [TK]D-Fender | Assid: No |
22:34.50 | *** join/#asterisk ManxPower (n=manxpowe@router.asteriasgi.com) |
22:35.09 | Assid | brilliant.. everyone else on ATA anwyays |
22:35.19 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
22:35.41 | Assid | [TK]D-Fender: oh, is it possible to check for the status of a phone on another asterisk server? |
22:35.56 | Assid | like i have 2 boxes, in 2 locations |
22:41.05 | [netman] | ~nat |
22:41.06 | jbot | somebody said nat was Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
22:41.06 | aiksa[LV] | hmm I was wondering perhaps someone did a backport of new manager events to 1.4.22 ? |
22:41.06 | [netman] | ~ports |
22:41.06 | jbot | methinks ports is http://www.debian.org/ports/, or http://www.isi.edu/in-notes/iana/assignments/port-numbers, or the FreeBSD ports system etc etc, or http://www.portforward.com/routers.htm |
22:41.16 | *** join/#asterisk sucituanbo (n=full@c-24-21-121-148.hsd1.wa.comcast.net) |
22:41.51 | *** join/#asterisk pittstains (n=frank@mx1.distributivenetworks.com) |
22:43.08 | pittstains | I have a question about passing variables from the AMI to the dialplan: is it possible? If so, how? Here's what I'm trying without success: |
22:43.12 | SQLDarkly | would MOH still be played with zap/pseudo even if telephony hardware is installed? |
22:44.52 | pittstains | in the AMI... |
22:44.52 | pittstains | Action: Originate |
22:44.52 | pittstains | [other necessary params] |
22:44.52 | pittstains | Variable: FOO=bar |
22:45.53 | pittstains | in the dialplan: |
22:45.53 | pittstains | exten => 0,n,Verbose(Test: ${FOO}) |
22:48.44 | *** join/#asterisk Jembanin (n=Ben@193.103-84-212.ippool.ndo.com) |
22:50.00 | pittstains | what i get when i run it is "Test: " instead of "Test: bar" |
22:50.23 | Katty | yeah we should probably see some pastebin |
22:50.25 | Katty | and some cli output |
22:50.28 | eppigy | no |
22:50.29 | Katty | mmmkay |
22:50.33 | eppigy | let me read your thoughtd |
22:50.36 | eppigy | thoughts |
22:50.40 | Katty | eyes eppigy |
22:50.43 | Katty | WHAT AM I THINKING |
22:50.45 | eppigy | 8[] |
22:51.13 | Katty | taps fingers |
22:51.21 | eppigy | hold please |
22:51.51 | Katty | http://mfrost.typepad.com/photos/uncategorized/2008/03/16/overslurp.jpg |
22:52.08 | pittstains | what do you need me to show you? |
22:52.37 | eppigy | you are currently wondering what is taking me so long to come up with a witty response? |
22:52.41 | eppigy | you are currently wondering what is taking me so long to come up with a witty response |
22:52.57 | drmessano | HA |
22:53.01 | drmessano | Now thats funny |
22:53.01 | eppigy | YEAH SON |
22:53.07 | drmessano | A cat without a caption |
22:53.15 | drmessano | OLD SKOOL |
22:53.40 | Katty | eppigy: http://pastebin.ca/1303076 |
22:53.45 | Katty | eppigy: what am i thinking? |
22:53.49 | drmessano | That cat is all like "I just got licked".. not like "ZOMG, HE R TASTED MEH" |
22:54.24 | eppigy | TRUE |
22:54.27 | eppigy | TIME |
22:54.28 | eppigy | TO |
22:54.29 | eppigy | DIP |
22:54.30 | eppigy | OUT |
22:54.34 | eppigy | yeah it is |
22:54.58 | eppigy | i just ordered take out |
22:55.02 | Katty | hot |
22:55.05 | eppigy | so i gotta give them about 10 minutes |
22:55.21 | Katty | yeah i'm on hold. |
22:55.25 | Katty | taps fingers |
22:55.34 | eppigy | KILL EVERYTHING |
22:55.40 | Deeewayne | waves to Katty |
22:55.46 | Katty | deeeeewayneee!!!! |
22:55.51 | Katty | hugs Deeewayne |
22:56.07 | eppigy | imitates a third wheel |
22:56.07 | Deeewayne | gives Katty a Happy New Year hug |
22:56.11 | Katty | :> |
22:56.16 | Katty | happy mew years to you too! |
22:56.37 | Katty | i learned so much today. |
22:56.46 | Katty | bioluminescent dana octopus squid |
22:56.52 | Katty | and sodium acetate hot ice. |
22:56.59 | pittstains | hmmmm, seems to be working now |
22:57.18 | aiksa[LV] | i just did a thing I thought is never possible! |
22:57.23 | pittstains | the only thing i changed was to change my variable name from THIS_IS_A_TEST to TEST |
22:57.34 | pittstains | i guess maybe underscores in variable names are bad |
22:57.58 | aiksa[LV] | I managed to trick zaptel to choose correct timing device by calling ztcfg twice in a row with two seperate configuration files |
22:58.04 | aiksa[LV] | yeah! :) |
22:58.21 | Katty | woo! time to go |
22:58.22 | Katty | darts |
23:00.11 | aiksa[LV] | if only tzafrir_laptop was here, he would say I am nuts |
23:00.13 | aiksa[LV] | :P |
23:01.23 | eppigy | pittstains: underscores denote inheritable variables |
23:01.32 | eppigy | i dont know if that has something to do with i t |
23:02.51 | pittstains | inheritable variables?!?! never heard of it -- time to reread the manual! |
23:02.54 | ix33 | is going from zaptel -> dahdi difficult or am i just retarded? |
23:03.53 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
23:06.22 | aiksa[LV] | ix33: it depends on what modules you are using |
23:08.14 | drmessano | ix33: Its very easy.. There 162 steps you need to follow in readme1.txt thru readme14.txt, but beyond thats it all gravy |
23:09.03 | *** part/#asterisk Alric (n=Alric@masq.hyperusa.com) |
23:09.32 | cvnet | anyone know of any softphone where you can change the sip port? |
23:09.56 | SQLDarkly | No match Their Call ID: 17f9@10.43.160.24 Their Tag 0x17f9-1owxe1 Our tag: as5908261c | any idea what that message means. THis is in my debug log at the point of drop |
23:10.04 | jasonwoot | most that I've used... cvnet |
23:10.06 | jasonwoot | zoiper |
23:10.31 | cvnet | jasonwoot thanks |
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23:12.18 | aiksa[LV] | drmessano: not to mention that some modules havent ported to dahdi yet |
23:12.25 | cvnet | i changed the port for sip to 2222 on *, yet i can not connect to it, I have the firewall off, is there way to find out if that port is open? |
23:12.48 | drmessano | Which ones? |
23:13.47 | _ShrikE | cvnet: nmap |
23:13.59 | cvnet | nmap is a unix command? |
23:14.38 | _ShrikE | yes, you may need to install it though |
23:14.54 | cvnet | yap nmap? |
23:15.08 | cvnet | i have it installed |
23:19.07 | cvnet | if i want to use a different por than 5060 for sip, is there any rules ? |
23:19.22 | jasonwoot | nmap localhost |
23:19.47 | jasonwoot | sip.conf must be configured to bind to that port however |
23:20.09 | jasonwoot | running a tcpdump on the appropriate interface will tell you if that port is responding |
23:20.18 | cvnet | thanks |
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23:22.33 | cvnet | hum, when i do nmap, and sipconf binds to 5060 it doesnt show it there |
23:22.44 | cvnet | yet i can connect via softphone |
23:23.06 | aiksa[LV] | drmessano: not the ones from digium of course |
23:23.14 | aiksa[LV] | third party modules |
23:23.23 | drmessano | Obviously |
23:23.29 | drmessano | Still doesnt answer |
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23:23.53 | aiksa[LV] | xpd_usb |
23:24.00 | aiksa[LV] | which relies on bristuff |
23:24.07 | aiksa[LV] | but I might be wrong |
23:25.26 | aiksa[LV] | cvnet: there are two meanings of a port in sip clients |
23:25.52 | aiksa[LV] | i hope this is all ABC to you, but stil not to confuse |
23:26.26 | aiksa[LV] | the SIP port option in Zoiper can realte to either a) The port on which zoiper is listening to b) to which port of the SIP server to connect to |
23:27.18 | *** join/#asterisk MrNaz (n=mrnaz@124-168-62-116.dyn.iinet.net.au) |
23:27.21 | aiksa[LV] | drmessano: I just finished a day long of butsex with zaptel :)) but in the end i managed to break its spirit ;))) |
23:28.15 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
23:28.57 | jjshoe | can sox "display/print" what "volume" a sound file is? |
23:29.06 | jjshoe | for example, when trying to normalize all your audio files to be the same level |
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23:31.22 | aiksa[LV] | jjshoe: sorry cant answer this one |
23:34.03 | drmessano | So.. google is working on building its own routers |
23:34.14 | drmessano | They are using their own switches already |
23:34.27 | drmessano | Pretty soon: Google will make it's own users |
23:34.39 | drmessano | Soylent green <-- |
23:35.24 | beek | drmessano: you said "make", not "eat" its own users. |
23:37.29 | [TK]D-Fender | beek: We're relatively sure that SoylentCo didn't eat their own product ;) |
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23:44.43 | cvnet | aiksa[LV] in Zoiper I only see one port option, Sip Option -> Port, I thought thats the port it connects to the server. have you ever tried to change the SIP port and have a softphone connected to ? |
23:48.17 | aiksa[LV] | cvnet let me open the configuration menu of my zoiper |
23:48.31 | cvnet | thanks |
23:48.59 | cvnet | when i click on Advance Option then i get the Sip Option |
23:49.03 | aiksa[LV] | i have that under protocol options |
23:49.17 | cvnet | ya and sip option right? |
23:49.20 | aiksa[LV] | thats a port Zoiper listen for SIP packets on |
23:49.28 | cvnet | that is the port which it connects to server right? |
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23:50.27 | aiksa[LV] | if you wanted to change a port on the server it connects to, i suppose you could simply add it with server.host.tld:5061 in specific account options |
23:50.31 | cvnet | so its not the port it connects to server? hum then why would it be 5060 by default |
23:50.44 | cvnet | hum, let me try taht |
23:50.49 | aiksa[LV] | cvnet: because 5060 is a default SIP port |
23:50.59 | aiksa[LV] | Zoiper is acting as both server and client |
23:51.11 | cvnet | ok let me try that |
23:52.18 | cvnet | oo boy |
23:52.32 | cvnet | thanks, I waisted 3 days to figure it out, man I feel dumb |
23:52.41 | cvnet | aiksa[LV] thanks a bunch |
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