| 00:16.37 | *** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 00:16.42 | *** join/#asterisk Corydon76-home (i=five@pdpc/supporter/sustaining/Corydon76-home) |
| 00:16.42 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 00:17.38 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-63-246.lns2.syd6.internode.on.net) |
| 00:23.07 | *** join/#asterisk rsd99 (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net) |
| 00:23.36 | rsd99 | does anyone know if there is a configuration utility out there for asterisk configs? |
| 00:23.38 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 00:23.45 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 00:23.48 | JT | i use vi myself |
| 00:24.29 | rsd99 | i was hoping there would be something out there that generated the config files. i just don't know where to start |
| 00:25.02 | rob0 | Used to be, "make samples", but now I think "make install" does that. |
| 00:25.47 | rob0 | IOW, the default config files give you a pretty good start. |
| 00:26.24 | JT | ~thebook |
| 00:26.24 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:26.27 | JT | is quite useful too |
| 00:28.04 | rsd99 | just a little overboard for me. i am just doing this for something to do at home. a simple ivr where someone dials in, and chooses an extension and it rings one of my ip phones |
| 00:28.51 | rsd99 | i have the calls working internally. |
| 00:31.58 | JT | so how is the book overboard? |
| 00:32.05 | JT | it applies just fine to at home |
| 00:32.15 | *** join/#asterisk _DAW (n=chatzill@adsl-222-41-108.msy.bellsouth.net) |
| 00:34.35 | [TK]D-Fender | rsd99, go to www.trixbox.org and use that. If you have any questions they have their own IRC channel & support forums. |
| 00:35.15 | Qwell | [TK]D-Fender: Who are you, and what have you done to [TK]D-Fender? |
| 00:35.31 | mihinomenest | someone gave up the ghost. |
| 00:36.11 | [TK]D-Fender | Qwell[], Preemptive strike to rid us of a person who doesn't actually want t learn anything. |
| 00:36.19 | JT | Strom_M: how much longer are you there for? |
| 00:36.21 | [TK]D-Fender | Qwell[], that is ENTIRELY me :) |
| 00:36.25 | Strom_M | I leave saturday |
| 00:37.59 | JT | Strom_M: ah, i thought you were just doing a plane change |
| 00:38.17 | Strom_M | no no; i'm in melbourne for a week |
| 00:38.31 | Strom_M | i only change planes in sydney |
| 00:39.02 | Jon335 | Does anyone know of a site that sells VoIP DECT phones in the US? |
| 00:42.14 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 00:42.27 | docelmo | anyone know anything about automon? |
| 00:43.03 | JT | Strom_M: ah ok |
| 00:43.11 | JT | Strom_M: business or fun? |
| 00:48.56 | *** join/#asterisk Strom_C (n=strom@60-241-200-26.static.tpgi.com.au) |
| 00:52.21 | [TK]D-Fender | BBIAB |
| 00:52.27 | *** join/#asterisk SwK (n=SwK@m090e36d0.tmodns.net) |
| 00:55.03 | plla | JT: thanks, it worked. |
| 00:55.26 | *** join/#asterisk Strom_C (n=strom@60-241-200-26.static.tpgi.com.au) |
| 00:55.28 | plla | 12 hours and the pri hasn't failed. |
| 00:55.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:56.35 | JT | plla: all you changed was the span line? |
| 00:56.52 | plla | yep |
| 00:57.11 | JT | nice |
| 00:57.16 | JT | good to see it helped |
| 00:57.23 | plla | Still it makes me wonder how has it been working in the other setup for 4 months. |
| 00:57.34 | JT | chance |
| 00:58.59 | *** join/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader) |
| 01:00.07 | spaceinvader | is it possible to connect an client to an iax channel from the console? |
| 01:00.41 | spaceinvader | i.e. something like dial but that affects a remote sip connection and not the console |
| 01:02.43 | *** join/#asterisk GlobeTrotter (n=eric@ip22-28-10-190.ct.co.cr) |
| 01:02.45 | JT | i really don't understand what you're asking |
| 01:02.47 | *** join/#asterisk Caplain (n=shayne@adsl-75-45-253-93.dsl.sfldmi.sbcglobal.net) |
| 01:02.55 | samy_b1 | hello |
| 01:03.16 | Caplain | whats a good fxo fxs card? |
| 01:03.24 | spaceinvader | JT: i have a client connected via sip |
| 01:03.28 | GlobeTrotter | hi i get this error when i start asterisk |
| 01:03.36 | GlobeTrotter | -/usr/lib64/asterisk/modules/chan_sip.so: undefined symbol: ast_osp_terminate |
| 01:03.37 | samy_b1 | can some one tell me why soem times DISA works and some time don't |
| 01:03.45 | spaceinvader | JT: the ui wont accept a IAX/ addr as its sip |
| 01:04.02 | spaceinvader | JT: can i somehow initiate the IAX call from the console and route it to that sip client? |
| 01:04.08 | samy_b1 | like 3 of the times i call from my cell i get it working and one not] |
| 01:04.18 | samy_b1 | is realy strange and i don't see any errors |
| 01:04.29 | samy_b1 | in the erros logs files |
| 01:04.30 | JT | spaceinvader: that makes no sense, why wouldn't you just use sip? |
| 01:04.50 | samy_b1 | any one head that issue too ? |
| 01:06.00 | spaceinvader | JT: i want to make a call to an IAX channel, but its only a one-off so i dont want to make an extension for it |
| 01:06.43 | JT | i assume you mean an iax channel somewhere other than your network |
| 01:07.02 | spaceinvader | yes |
| 01:08.04 | JT | no, i don't know an easy way to do that |
| 01:08.09 | JT | an iax softphone would be easy |
| 01:08.53 | Strom_C | why not make a temporary extension that you delete later? |
| 01:08.56 | Strom_C | or would that be too easy? :) |
| 01:09.29 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 01:11.22 | JT | Strom_C: so you here on business? |
| 01:11.45 | Strom_C | yes |
| 01:12.48 | GlobeTrotter | i get this error when i try to start astereisk ::load_modules: Unable to open modules directory /usr/lib/asterisk/modules |
| 01:13.33 | _VoiceMeUp_COM | ls -la /usr/lib/asterisk/modules/*.so |
| 01:13.37 | _VoiceMeUp_COM | anything in there ? |
| 01:13.54 | _VoiceMeUp_COM | if no modules ./configure --enable-shared |
| 01:13.55 | _VoiceMeUp_COM | and redo |
| 01:14.35 | JT | Strom_C: do you have to change planes in sydney, or just get off and reboard the same flight? |
| 01:14.36 | GlobeTrotter | where do i run this command? |
| 01:14.42 | GlobeTrotter | there is nothing in there |
| 01:14.54 | GlobeTrotter | whre do i run ./configure --enable-shared |
| 01:14.55 | _VoiceMeUp_COM | in asterisk src dir |
| 01:15.01 | _VoiceMeUp_COM | how you get asterisk ? rpm ? |
| 01:15.03 | _VoiceMeUp_COM | or src ? |
| 01:15.21 | _VoiceMeUp_COM | if rpm .. then type updatedb & |
| 01:15.50 | GlobeTrotter | BEcd |
| 01:15.56 | GlobeTrotter | bussiness edition cd |
| 01:16.25 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:16.28 | _VoiceMeUp_COM | once done locate pbx_functions.so |
| 01:16.29 | Strom_C | JT: it's a completely new plane in sydney |
| 01:16.32 | _VoiceMeUp_COM | and youll find where htey are |
| 01:16.42 | _VoiceMeUp_COM | then check asterisk.conf fo rlibdir and see if they match |
| 01:16.48 | _VoiceMeUp_COM | im out luck to you |
| 01:17.12 | JT | Strom_C: so would you have time to meet at the airport (would it even be possible? heh) |
| 01:17.13 | _VoiceMeUp_COM | but you paid 1000 for this , so it includes support i hear.. you should have an 800 support # to bitch to |
| 01:17.40 | _VoiceMeUp_COM | <PROTECTED> |
| 01:17.44 | _VoiceMeUp_COM | that means DIR not there |
| 01:17.46 | _VoiceMeUp_COM | btw |
| 01:17.53 | _VoiceMeUp_COM | not that it cant find the modules |
| 01:17.56 | _VoiceMeUp_COM | so my bad |
| 01:18.09 | _VoiceMeUp_COM | still locate pbx_functions.so |
| 01:18.46 | Strom_C | JT: um, i'd have to check my schedule |
| 01:19.05 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:19.05 | JT | Strom_C: i'm thinking it might not even be possible |
| 01:19.13 | JT | if you're in the secure area |
| 01:19.26 | Strom_C | yeah |
| 01:19.48 | JT | if it's the international terminal, you need to clear customs to get in, and you need a boarding pass for that :/ |
| 01:19.55 | JT | i did that last week |
| 01:19.57 | JT | was in japan |
| 01:21.28 | GlobeTrotter | astmoddir => /usr/lib/asterisk/modules is what is in asterisk.conf.. but the files are in astmoddir => /usr/lib64/asterisk/modules |
| 01:22.14 | *** join/#asterisk mindCrime (n=chatzill@65.190.188.124) |
| 01:25.33 | Strom_C | JT: well, you could always pop down to melbs for dinner or somesuch :) |
| 01:25.37 | Strom_C | how long is the drive? |
| 01:26.21 | JT | 8.5 hours |
| 01:26.50 | JT | that's with a max of an hour break, obeying the speed limit mostly :) |
| 01:27.06 | JT | it's cheaper to fly for one person travelling, due to fuel costs |
| 01:27.32 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:27.32 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:27.41 | GlobeTrotter | yeah i paid but the support is now closed |
| 01:27.43 | GlobeTrotter | :( |
| 01:30.56 | *** join/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com) |
| 01:31.15 | Cyber-Dogg | I have zaptel and asterisk setup on my system... but I'm having a couple isses |
| 01:31.34 | Cyber-Dogg | I have zaptel.conf, zapata.conf and extensions.conf all setup the way I want... |
| 01:32.01 | Cyber-Dogg | i do not have a dial tone on my phones though... and when I load up asterisk... I don' thave any zap commands available |
| 01:32.05 | Cyber-Dogg | any thoughts of what to check? |
| 01:32.13 | Cyber-Dogg | lsmod shows that zaptel is loaded... |
| 01:33.09 | JT | is the card's driver loaded? |
| 01:33.25 | Cyber-Dogg | I think so... |
| 01:33.46 | Cyber-Dogg | wct4xxp and wctdm are both loaded according to lsmod |
| 01:33.54 | Cyber-Dogg | but don't I need wctfxo too? |
| 01:34.30 | JT | err |
| 01:34.40 | JT | you only need the driver for your card loaded |
| 01:34.45 | Cyber-Dogg | oh ok |
| 01:34.51 | Cyber-Dogg | well i think it is then... I have a 400p |
| 01:35.03 | JT | you have 2 drivers loaded |
| 01:35.20 | Cyber-Dogg | should I kill the wctdm then? |
| 01:38.57 | Cyber-Dogg | how do I unload it? |
| 01:39.39 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:41.46 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 01:43.05 | Cyber-Dogg | in my modules.autoload.d I only have zaptel and wct4xxp |
| 01:43.10 | Cyber-Dogg | so I'm not sure how to undo the other one |
| 01:43.51 | JT | rmmod |
| 01:44.13 | Cyber-Dogg | ok... let me see if that helped any |
| 01:44.54 | Cyber-Dogg | that breaks it worse... |
| 01:44.59 | Cyber-Dogg | I think I need wctdm |
| 01:45.54 | Cyber-Dogg | when I do ztcfg -v the channel map looks right... so I think everything is ok with my channels |
| 01:48.06 | *** part/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com) |
| 01:50.11 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 01:52.03 | *** join/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com) |
| 01:52.14 | tengulre | nobody here? |
| 01:52.16 | Cyber-Dogg | I am |
| 01:52.27 | Cyber-Dogg | but I'm pretty worthless... :-) |
| 01:52.31 | Cyber-Dogg | I'm trying to get help myself |
| 01:53.57 | [TK]D-Fender | www.drphil.com |
| 01:54.07 | Cyber-Dogg | he's a crock... |
| 01:57.10 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 01:59.45 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 01:59.45 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 02:00.43 | *** join/#asterisk ehaupt_ (n=ehaupt@unaffiliated/ehaupt) |
| 02:02.50 | *** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp) |
| 02:03.02 | Cyber-Dogg | so... anyone have any ideas why I'm having problems? |
| 02:03.14 | Cyber-Dogg | my mapping looks right... |
| 02:03.21 | Cyber-Dogg | but no dialtone... |
| 02:04.02 | *** join/#asterisk jmacz (n=jmacz@190.24.103.191) |
| 02:05.25 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 02:07.50 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 02:08.13 | Cyber-Dogg | when I am in the cli... shouldn't I have some "zap" commands available to me? |
| 02:08.20 | Cyber-Dogg | liek... zap show status |
| 02:09.08 | *** join/#asterisk flujan (n=flujan@201-42-103-137.dsl.telesp.net.br) |
| 02:09.51 | *** part/#asterisk flujan (n=flujan@201-42-103-137.dsl.telesp.net.br) |
| 02:10.40 | JT | yes, unless chan_zap failed to load |
| 02:11.03 | *** join/#asterisk lmoreira (n=lmoreira@201009076233.user.veloxzone.com.br) |
| 02:11.39 | *** join/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
| 02:12.03 | *** part/#asterisk steve___ (n=steve@kit-dhcp1.porchlight.ca) |
| 02:12.21 | Cyber-Dogg | chan_Zap.... how do I chekc that? |
| 02:13.13 | *** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net) |
| 02:13.15 | JT | there's probably some error that prevented it from loading |
| 02:13.24 | Cyber-Dogg | where can I check to see? |
| 02:13.50 | Cyber-Dogg | dmesg doesn't show anything... |
| 02:13.57 | *** part/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net) |
| 02:14.14 | *** join/#asterisk SwK (n=SwK@m055e36d0.tmodns.net) |
| 02:15.01 | JT | the startup of asterisk |
| 02:15.14 | Cyber-Dogg | is ther ea log file for that? |
| 02:16.04 | JT | if you have enabled full log, yes |
| 02:16.19 | Cyber-Dogg | and where might that log file be? |
| 02:17.29 | Cyber-Dogg | when I type /etc/init.d/asterisk start it shows !! so I'm assuming thre is an error |
| 02:18.23 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 02:18.37 | JT | where all asterisk logs are kept /var/log/asterisk |
| 02:19.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:20.18 | Cyber-Dogg | usr/lib/asterisk/modules/chanzap: file not found |
| 02:20.51 | JT | is this asterisk from source or a package? |
| 02:21.30 | Cyber-Dogg | emerge in gentoo |
| 02:21.34 | Cyber-Dogg | source I believe |
| 02:23.04 | JT | would seem you compiled it in the wrong order |
| 02:23.24 | Cyber-Dogg | oh... so... I need to do zaptel... then asterisk I assum? |
| 02:23.25 | JT | compile zaptel, then asterisk |
| 02:23.27 | JT | yes |
| 02:23.29 | Cyber-Dogg | ok |
| 02:23.44 | Cyber-Dogg | so... since have both already installed... can I just redo asterisk? |
| 02:24.26 | JT | i guess so |
| 02:24.34 | Cyber-Dogg | ok... here goes nothing |
| 02:25.11 | *** join/#asterisk n00dle (n=ccraft@ip-249-27.springsips.com) |
| 02:27.19 | *** join/#asterisk flujan (n=flujan@201-42-103-137.dsl.telesp.net.br) |
| 02:29.01 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 02:33.25 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 02:33.39 | Cyber-Dogg | ok... I did a rebuild... |
| 02:35.09 | Cyber-Dogg | now chan_zap.so is ther |
| 02:35.21 | Cyber-Dogg | should I do a rboot? |
| 02:35.33 | Cyber-Dogg | or is stopping asterisk and zaptel... then restarting them enough? |
| 02:37.06 | JT | no need to rebook |
| 02:37.09 | JT | reboot |
| 02:37.14 | JT | you didn't change kernel or hardware |
| 02:37.17 | Cyber-Dogg | nope |
| 02:37.25 | Cyber-Dogg | well... I restart zaptel |
| 02:37.35 | Cyber-Dogg | and then I tried to restart asterisk... but I don't think it is... |
| 02:37.40 | Cyber-Dogg | I can't connect to the CLI |
| 02:37.51 | Cyber-Dogg | unable to connect |
| 02:38.09 | Cyber-Dogg | I checked the log and there isn't anything there |
| 02:42.43 | *** join/#asterisk ^rocket^ (n=rocket@c-71-235-48-164.hsd1.ct.comcast.net) |
| 02:43.39 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 02:44.59 | *** join/#asterisk logicwrath (n=some@c-68-60-121-112.hsd1.mi.comcast.net) |
| 02:45.16 | xheliox | NOTICE[19527]: chan_sip.c:5250 process_sdp: No compatible codecs, not accepting this offer! |
| 02:45.35 | xheliox | I'm seeing this a lot lately -- is there anyway for it to be more verbose? |
| 02:45.45 | xheliox | E.g. The IP it's rejecting? |
| 02:46.59 | *** join/#asterisk nephfl (i=nephilim@wsip-70-184-144-158.ga.at.cox.net) |
| 02:49.06 | *** join/#asterisk SwK_ (n=SwK@m095e36d0.tmodns.net) |
| 02:49.43 | snuffy22 | sip debug ;) |
| 02:50.08 | snuffy22 | only problem is then u get the kitchen sink as well |
| 02:50.28 | xheliox | yeah |
| 02:50.32 | xheliox | it's only happening every few hours |
| 02:50.52 | xheliox | I've allowed all codecs.. and I'm still getting that, so I think it's pretty bogus, I'm just curious :) |
| 02:53.20 | *** join/#asterisk rsd99 (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net) |
| 02:53.42 | rsd99 | when i go to listen to a voicemail, all i hear is static |
| 02:53.48 | *** join/#asterisk Bryan93108 (n=chatzill@131.muf30.hrfr.wswdc01r18.dsl.att.net) |
| 02:54.21 | Bryan93108 | can anyone tell me if SRTP is usable in Asterisk yet, with the Linksys ATAs? |
| 02:55.32 | JT | xheliox: do you have g.729 licenses? |
| 02:56.11 | Bryan93108 | no but will obtaini them if needed (kinda new to SRTP) -- whom do I license it from? |
| 02:56.48 | JT | Bryan93108: i wasn't talking to you there :) |
| 02:56.54 | JT | unless you're xheliox |
| 02:56.55 | Bryan93108 | doh |
| 02:57.40 | [TK]D-Fender | Bryan93108, No. |
| 02:57.44 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-37-61.pskn.east.verizon.net) |
| 02:58.15 | rsd99 | i just finished setting up voicemail. when i go to play a new vm, all i hear is static |
| 02:58.18 | rsd99 | any ideas? |
| 02:58.24 | ^rocket^ | I installed asterisk opn Xubuntu, and port 5060 isn't open when I do an nmap scan, what am I doing wrong? |
| 02:58.26 | xheliox | JT: Hmm. I might not on that box... |
| 02:58.46 | Strom_C | ^rocket^: why are you running asterisk on a box with x windows? |
| 02:58.58 | JT | xheliox: perhaps a client is requesting a codec with no transcoder available |
| 02:59.08 | JT | which'd most likely be g.729 |
| 02:59.11 | ^rocket^ | it's running on an iMac, someone gave me a boot disk for Macppc |
| 02:59.20 | Strom_C | ..... |
| 02:59.27 | xheliox | Yeah, I'm willing to accept that ;) Checking now... |
| 02:59.27 | rsd99 | is the mac running OSX |
| 02:59.28 | ^rocket^ | I had poor results booting other distros |
| 02:59.44 | ^rocket^ | rsd99 no |
| 03:00.07 | ^rocket^ | It's Linux muffin 2.6.15-26-powerpc #1 Fri Sep 8 19:51:33 UTC 2006 ppc GNU/Linux |
| 03:00.09 | rsd99 | just curious |
| 03:00.20 | [TK]D-Fender | ^rocket^, SIP uses *UDP*, not *TCP*. You'd have to adjust your scan. |
| 03:00.53 | rsd99 | i used to tun macppc back in the day. ;-) |
| 03:00.56 | xheliox | Hmm, yeah, I have g729 licenses installed, good thought though :) |
| 03:01.01 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-18-14-219.nycmny.east.verizon.net) |
| 03:01.25 | JT | g.723, iLBC, or a few others could be culprits |
| 03:01.37 | xheliox | just should be nice to know where it's coming from |
| 03:01.50 | xheliox | should/would |
| 03:02.03 | ^rocket^ | [TK]D-Fender: nmap -PU shows no port 5060 open |
| 03:02.19 | ^rocket^ | I can't connect a P2000W to it |
| 03:02.41 | *** join/#asterisk Cyber-Dogg (n=jphelps@24-178-240-97.dhcp.stls.mo.charter.com) |
| 03:02.46 | Cyber-Dogg | ok... I'm back again... still no more progress |
| 03:02.54 | JT | a P2000W, yes, that |
| 03:03.01 | Cyber-Dogg | chan_zap exists... but I can't load asterisk |
| 03:03.23 | ^rocket^ | nor my IAXy |
| 03:03.32 | JT | Cyber-Dogg: is zaptel.conf and zapata.conf setup? |
| 03:03.35 | Cyber-Dogg | yes |
| 03:03.47 | [TK]D-Fender | ^rocket^, your failure to properly setup a specific device does not mean much... |
| 03:03.47 | JT | did you run ztcfg? |
| 03:03.49 | Cyber-Dogg | and ztcfg -v yields the channel map appropriately |
| 03:04.11 | Cyber-Dogg | well.. ztcfg -vv |
| 03:04.13 | ^rocket^ | well, shouldn't a port scan show open ports? |
| 03:04.25 | ^rocket^ | they have in the past with other setups I did |
| 03:04.36 | ^rocket^ | I had no problems with asterisk on debian or BSD |
| 03:04.41 | JT | ^rocket^: check netstat |
| 03:05.10 | Cyber-Dogg | zapata.conf is setup as well |
| 03:05.17 | ^rocket^ | JT: I do see open ports with tha |
| 03:06.30 | JT | so clearly you have a firewall/network issue |
| 03:06.50 | ^rocket^ | JT: k |
| 03:06.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 03:06.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 03:07.49 | Cyber-Dogg | I am supposed to edit the zapata.conf in /etc/asterisk right? |
| 03:09.56 | [TK]D-Fender | JT, nothing is clear right now. we have no idea if his configs are sane |
| 03:11.40 | ^rocket^ | I'm gonna reboot and see if that helps |
| 03:11.48 | JT | [TK]D-Fender: well he said that netstat had the sip port as open, if he was right about this, the issue is networking related :) |
| 03:12.05 | ^rocket^ | I had two different IPs listening on UDP 5060 |
| 03:12.18 | Cyber-Dogg | I'm certian that zaptel.conf and zapata.conf are right |
| 03:12.26 | Cyber-Dogg | the issue has to be with asterisk somewhere... |
| 03:12.27 | [TK]D-Fender | JT, or the scan was done wrong. |
| 03:12.34 | Cyber-Dogg | or some other config issue |
| 03:12.49 | JT | [TK]D-Fender: yes that's possible, but assume he also can't connect with some sip stuff |
| 03:12.52 | ^rocket^ | nmap -sU shows only DHCP listening |
| 03:13.46 | [TK]D-Fender | Cyber-Dogg, never assume your configs are right and DEFINITELY never think for a second they we assume it. |
| 03:13.59 | Cyber-Dogg | LOL... ok |
| 03:14.26 | [TK]D-Fender | Cyber-Dogg, first rule... if everything was right... it would WORK. So clearly somethings ^&%#ed up. |
| 03:14.36 | Cyber-Dogg | right... |
| 03:14.57 | Cyber-Dogg | well... zaptel.conf is pretty straight forward... |
| 03:14.58 | [TK]D-Fender | Cyber-Dogg, Next tip : Never nag about a problem without providing a DETAILED pastebin including everything relevent to the problem. |
| 03:15.16 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:15.33 | JT | third rule: too many dots make the baby jesus cry |
| 03:15.48 | [TK]D-Fender | Cyber-Dogg, next tip : Never try to explain why you think your configs are right without SHOWING THEM. Again it is MY automatic assumption that you have in all likelyhood screwed up EVERY file you may have gotten your hands on :) |
| 03:15.52 | *** join/#asterisk iratsu (n=iratsu@modemcable090.239-59-74.mc.videotron.ca) |
| 03:16.29 | Cyber-Dogg | alright alright alright,,, I'll go make a pastebin |
| 03:16.45 | [TK]D-Fender | Cyber-Dogg, good, because I was running out of rules :p |
| 03:16.57 | Cyber-Dogg | I was hoping I hadn't done much more wrong... |
| 03:16.59 | JT | commas are basically dots for the purposes of my rule ;) |
| 03:17.02 | Cyber-Dogg | ;-) |
| 03:17.10 | Cyber-Dogg | LOL sorry it's a habit |
| 03:18.20 | nephfl | Don't forget the tip that since people in IRC are not paid to help, they have no incentive not to be an ass, so always expect to get what you pay for and consider yourself lucky when you get more. (not to say there aren't a ton of very helpful people in irc, but there are alot of asses too) |
| 03:19.40 | nephfl | Shoot, even when you pay for support you get mixed results. |
| 03:20.00 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 03:20.44 | Cyber-Dogg | http://pastebin.com/932351 |
| 03:22.02 | Cyber-Dogg | as mentioned, ztcfg displays my channel map appropriately but after I execute asterisk, I am unable to connect to the CLI |
| 03:22.11 | Cyber-Dogg | and the log has nothing in it |
| 03:22.16 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:22.25 | blitzrage | zup hombres |
| 03:22.55 | [TK]D-Fender | Cyber-Dogg, how are you starting *? |
| 03:23.05 | Strom_C | Cyber-Dogg: what happens when you start asterisk by typing "asterisk -cvvvvvvvg"? |
| 03:23.34 | [TK]D-Fender | Cyber-Dogg, indeed, start it exactly as shown above |
| 03:24.04 | Cyber-Dogg | hey! now that freaking helpful |
| 03:24.12 | Cyber-Dogg | I wish I would have known that before :-) |
| 03:25.31 | Cyber-Dogg | a few notices |
| 03:25.34 | Cyber-Dogg | 2 warnings |
| 03:25.44 | blitzrage | does it die at some point? |
| 03:25.58 | Cyber-Dogg | yes, loading module chan_oss.so |
| 03:26.56 | *** join/#asterisk apardo (n=deal@26.144.217.87.dynamic.jazztel.es) |
| 03:27.18 | Cyber-Dogg | it just says failed |
| 03:27.20 | Zion800 | Does anyone have any experience setting up Broadvoice with Asterisk? |
| 03:27.53 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 03:28.51 | [TK]D-Fender | Cyber-Dogg, add "noload => chan_oss.so" to modules.conf |
| 03:29.22 | Cyber-Dogg | where is the modules.conf? |
| 03:29.48 | Cyber-Dogg | n/m |
| 03:29.48 | JT | where all your asterisk configuration files are |
| 03:29.51 | [TK]D-Fender | Cyber-Dogg, same folder as the rest of *'s configs |
| 03:30.25 | JT | [TK]D-Fender: keep your windowsisms away ;) |
| 03:30.43 | Cyber-Dogg | ok, I added that |
| 03:31.05 | Cyber-Dogg | retry the asterisk -cvvvvvvvvvg |
| 03:31.06 | Cyber-Dogg | ? |
| 03:31.11 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:31.18 | JT | that'd be the idea |
| 03:31.27 | JT | i'm sure you can work out what these steps are doin |
| 03:31.29 | JT | doing |
| 03:31.33 | Cyber-Dogg | yes |
| 03:31.55 | Cyber-Dogg | well the reason I asked was because I had already tried the asterisk -cvvvvvvvvg and it still says the same thing |
| 03:32.03 | Cyber-Dogg | so I was hoping there was an additional step yet ;-) |
| 03:35.12 | [TK]D-Fender | Cyber-Dogg, pastebin the failure so we can see |
| 03:35.18 | [TK]D-Fender | Cyber-Dogg, and your modules.conf |
| 03:35.37 | Cyber-Dogg | etc/modules.conf right? |
| 03:35.49 | JT | no |
| 03:35.52 | JT | <PROTECTED> |
| 03:36.08 | Cyber-Dogg | well... that would be a problem |
| 03:36.21 | Cyber-Dogg | I don' thave a modules.conf ther |
| 03:36.44 | Cyber-Dogg | I have one in /etc/asterisk though |
| 03:36.47 | Cyber-Dogg | err... /etc |
| 03:37.24 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 03:37.26 | Cyber-Dogg | http://pastebin.com/932362 |
| 03:37.51 | nowork | hi, in asterisk CLI, what command I can check if the call is proxy mode or not? thanks |
| 03:38.10 | Strom_C | Cyber-Dogg: pastebin.ca seems to be sucking less than pastebin.com, fyi :) |
| 03:38.25 | JT | Cyber-Dogg: wrong modules.conf |
| 03:38.32 | JT | Cyber-Dogg: remove the line from that file |
| 03:38.35 | Cyber-Dogg | I did |
| 03:38.47 | JT | it sounds like you didn't make the sample asterisk configurations |
| 03:39.10 | Cyber-Dogg | doesn't ring a bell :-) |
| 03:39.37 | JT | if there's nothing in /etc/asterisk, it's likely you didn't make them |
| 03:39.39 | [TK]D-Fender | JT : I was right behind you on that :) |
| 03:39.47 | Cyber-Dogg | I have some files, just not that one |
| 03:40.17 | [TK]D-Fender | Cyber-Dogg, backup the configs you know you did by hand (extensions.conf, sip.conf, zapata.conf, etc....) and do a "make samples" |
| 03:40.18 | JT | sounds defective |
| 03:40.47 | Cyber-Dogg | where do I need to be to do the make samples? |
| 03:40.50 | Cyber-Dogg | I'm on gentoo |
| 03:40.54 | nowork | hi,TK>.. in asterisk CLI, what command I can check if the call is proxy mode or not? thanks |
| 03:41.03 | Strom_C | Cyber-Dogg: asterisk source directory... |
| 03:41.06 | [TK]D-Fender | nowork, * is NOT a proxy. |
| 03:41.44 | nowork | TK..hm, I just don't want my client see my SIP DID provider 's ip address. |
| 03:42.06 | nowork | TK..I don't know what term to express this.. |
| 03:42.12 | [TK]D-Fender | nowork, "canreinvite=no" |
| 03:42.18 | Cyber-Dogg | would that be /usr/portable/net-misc/asterisk |
| 03:42.24 | Cyber-Dogg | cause it doesn't wrok ther |
| 03:42.27 | Cyber-Dogg | it says no target |
| 03:42.32 | [TK]D-Fender | nowork, use gratuitously throughout sip.conf |
| 03:43.04 | JT | Cyber-Dogg: if you can't work out gentoo, i suggest you download the source from asterisk.org |
| 03:44.19 | [TK]D-Fender | Packaged * = ASS |
| 03:44.27 | nowork | TK..thank you.. |
| 03:44.52 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:44.53 | JT | [TK]D-Fender: it was compiled in gentoo from a gentoo package |
| 03:45.24 | [TK]D-Fender | JT : should have been labeled "Fruit Loops". |
| 03:45.40 | JT | heh |
| 03:47.21 | Cyber-Dogg | I'm trying to find it |
| 03:47.38 | JT | asterisk.org > download |
| 03:47.59 | rob0 | Cleverly hidden there! |
| 03:48.51 | Strom_C | totally impossible to find! |
| 03:50.24 | Cyber-Dogg | eh... I just made my own modules.conf and it works now LOL |
| 03:51.10 | JT | rofl, i guess |
| 03:52.33 | *** join/#asterisk bmg505 (n=leon@196.209.177.223) |
| 03:52.37 | Cyber-Dogg | woot |
| 03:52.58 | Cyber-Dogg | ok! |
| 03:53.14 | Cyber-Dogg | so, should I use SIP or IAX for two systems? |
| 03:53.33 | JT | up to you |
| 03:53.50 | Cyber-Dogg | which would you recommend? |
| 03:54.10 | JT | depends on what you're trying to do |
| 03:54.35 | Cyber-Dogg | just be able to call from one system to the other and use voicemail across them too |
| 03:56.18 | blitzrage | nicer to use SIP across all of them, but it's a bit of a pain with SIP |
| 03:56.25 | blitzrage | (at least until TFoT2 comes out) |
| 03:56.33 | rob0 | If doing NAT, IAX is easier. |
| 03:56.34 | blitzrage | since i documented that scenario this time around |
| 03:56.42 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 03:56.59 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:57.13 | rob0 | And both are pretty well covered at the Wiki ... Asterisk+dual+servers |
| 03:57.16 | JT | if doing a big multiserver cluster, SIP has advantages |
| 03:57.27 | JT | like being able to use OpenSER |
| 03:57.37 | Cyber-Dogg | just two |
| 03:57.45 | Strom_C | blitzrage: is a draft of TFoT2 available, or will we just have to wait? :) |
| 03:58.00 | blitzrage | just have to wait at this point -- we're in copyedit mode right now |
| 03:58.07 | Strom_C | alright |
| 03:58.23 | blitzrage | I have about 500 pages to go through (and is about 2/3s of the book :)) |
| 03:58.38 | Cyber-Dogg | well, I'm goign to save that task for tomorrow |
| 03:58.48 | Cyber-Dogg | I really appreciate the help and the hints and tips too! :-) |
| 03:59.03 | Cyber-Dogg | I'll be sure to use the pastebin in the future |
| 03:59.26 | Cyber-Dogg | ok time to crash... |
| 03:59.31 | Cyber-Dogg | needed to get that last ... in there ;-) |
| 03:59.34 | Cyber-Dogg | g'night all |
| 03:59.37 | blitzrage | night |
| 04:01.37 | Strom_C | tomorrow afternoon!! |
| 04:01.43 | Zion800 | strom! |
| 04:01.52 | Strom_C | hi? |
| 04:01.58 | Zion800 | its michael! |
| 04:02.04 | Strom_C | oh hey! :) |
| 04:02.14 | Zion800 | u still in au? |
| 04:02.26 | Strom_C | yeah |
| 04:02.36 | Zion800 | hows ur class? |
| 04:03.52 | Strom_C | going well |
| 04:04.10 | Zion800 | r they smart? |
| 04:05.32 | rue_mohr | does asterisk support call waiting lights on analog phones? |
| 04:05.47 | [TK]D-Fender | rue_mohr, Yes |
| 04:05.55 | rue_mohr | I been told its done with dtmf tones while on hook... |
| 04:05.57 | rue_mohr | cool |
| 04:06.11 | [TK]D-Fender | rue_mohr, More specifically Digium's analog cards and jsut about every ATA I've ever seen |
| 04:06.27 | rue_mohr | have to dig up some docs for that and mod my phones |
| 04:06.30 | [TK]D-Fender | rue_mohr, Oh wait.. CW lights... well thats the PHONE's job |
| 04:06.46 | Strom_C | rue_mohr: it's not DTMF tones |
| 04:06.49 | Strom_C | it's FSK data bursts |
| 04:06.55 | rue_mohr | no, message in voicemail thing |
| 04:06.56 | Strom_C | Zion800: yeah, they're having a good time |
| 04:07.03 | rue_mohr | didn't mean call waiting |
| 04:07.11 | rue_mohr | what you thought I meant was what I did |
| 04:07.19 | rue_mohr | hmm |
| 04:07.25 | Strom_C | rue_mohr: the analog card sends an FSK data burst that says "turn the MWI on" |
| 04:07.31 | Strom_C | how the phone actually handles that is up to the phone |
| 04:07.37 | rue_mohr | same fsk as call display? |
| 04:07.44 | Strom_C | well, similar |
| 04:07.47 | Strom_C | but not identical :) |
| 04:07.54 | rue_mohr | same baud rate? |
| 04:07.59 | Strom_C | yeeesssssss |
| 04:08.10 | Strom_C | just different data encoded |
| 04:08.15 | rue_mohr | same freq for 1/0? |
| 04:08.19 | rue_mohr | yea |
| 04:08.21 | rue_mohr | hmm |
| 04:08.35 | rue_mohr | might have a use for the parts from those call displays after all |
| 04:08.39 | rue_mohr | oh wait |
| 04:08.41 | rue_mohr | idea |
| 04:08.54 | rue_mohr | you dont have to RING to send call display data do you |
| 04:08.55 | Strom_C | rue_mohr: jesus, can you please stop pressing enter every three words? |
| 04:09.05 | rue_mohr | sorry |
| 04:09.12 | [TK]D-Fender | rue_mohr, Ditch that analog crap and buy a decent SIP phone :) |
| 04:09.26 | rue_mohr | ok, whats your credit card number again? |
| 04:09.55 | Strom_C | 4417 0131 1555 2368 |
| 04:09.57 | blitzrage | I DON'T WANT TO MEET YOUR MOM |
| 04:10.05 | rue_mohr | I have a bunch of call display modules, I'm thinking with a few tweeks, I could send messages with them |
| 04:10.22 | Strom_C | "call display" -- you must be canadian |
| 04:10.26 | *** join/#asterisk centrex (n=mythtv@c-68-62-167-203.hsd1.al.comcast.net) |
| 04:10.31 | rue_mohr | ? |
| 04:10.34 | Strom_C | here in the rest of the universe, we call that "caller ID" |
| 04:10.40 | rue_mohr | yea that |
| 04:10.58 | rue_mohr | how do i do this enter key reduction thing? |
| 04:11.06 | rue_mohr | I know, I go to bed |
| 04:11.46 | [TK]D-Fender | blitzrage, I JUST WANT |
| 04:11.52 | blitzrage | ! ! ! |
| 04:11.59 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:12.01 | [TK]D-Fender | z0mg |
| 04:12.06 | blitzrage | anyone know what the MS Exchange replacement software is for Linux? |
| 04:12.13 | blitzrage | z0mghawt |
| 04:12.32 | kiscokid | blitzrage Postfix? |
| 04:12.38 | [TK]D-Fender | blitzrage, Ximian Evolution |
| 04:12.48 | [TK]D-Fender | blitzrage, Oh, exchange, not Outlook |
| 04:12.55 | blitzrage | exchange :) |
| 04:12.57 | kiscokid | depends on what functions of Exchange you want |
| 04:13.11 | blitzrage | well, I'm just curious to look |
| 04:13.12 | [TK]D-Fender | blitzrage, There is always OpenExchange. |
| 04:13.16 | blitzrage | aha |
| 04:13.27 | *** join/#asterisk justdave_ (n=dave@unaffiliated/justdave) |
| 04:13.30 | blitzrage | I'm just curious of the list of features it supports right now |
| 04:13.35 | [TK]D-Fender | blitzrage, semi-free or SomethingOrAnother.... |
| 04:13.53 | [TK]D-Fender | don't fully recall.... |
| 04:14.04 | blitzrage | eh? |
| 04:14.09 | [TK]D-Fender | Head office... heck MY office poos on OSS these days... |
| 04:14.21 | blitzrage | ah |
| 04:14.22 | kiscokid | also check out Postfix mail server |
| 04:14.45 | blitzrage | what I'd really like is push mailing to my cell phone (Nokia E61i) |
| 04:15.11 | JT | kiscokid: postfix is great for smtp, it is not an exchange replacement, however |
| 04:15.40 | kiscokid | doesn't postfix do imap and pop3? |
| 04:16.09 | blitzrage | and I'm not even using Exchange now (thank god), I use Google for my Calendar and Email... just looking at some other things for interest |
| 04:16.16 | blitzrage | although I should go to bed ... got a flight in the morning |
| 04:16.17 | blitzrage | night all! |
| 04:16.21 | JT | kiscokid: no |
| 04:16.33 | JT | imap and pop3 are also NOT exchange replacements |
| 04:17.00 | JT | postfix can work together with stuff which does imap and pop3 access |
| 04:17.42 | kiscokid | ok, I stand corrected |
| 04:18.27 | JT | i'm thinking you've never used outlook/exchange |
| 04:18.43 | rob0 | I know what he's talking about but can't think of the name now. |
| 04:19.11 | rob0 | Someone (novell?) built a groupware suite based on Postfix. |
| 04:19.37 | kiscokid | I'm trying to understand the physical wiring aspects of my current Norstar/Nortel PBX and what I would need to replace the PBX with *... |
| 04:19.47 | JT | i think there have been a couple of solutions to try and replace exchange functionality |
| 04:20.36 | rob0 | Suits seem to crave it. Geeks don't seem to care enough to make a free software solution. |
| 04:20.43 | kiscokid | currently have 5 analog lines with "hunt" and one DID analog trunk (which provides access to 20 extensions) |
| 04:21.24 | JT | kiscokid: i'd be tempted to replace all the lines with a fractional PRI |
| 04:21.50 | kiscokid | If I looked at the wiring of my PBX would I expect to find 5 cables terminated by RJ-11 plugs ? Would there be one for the DID? |
| 04:21.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:22.20 | JT | depends, the more professional stuff is terminated by punchdown blocks |
| 04:22.30 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 04:22.37 | kiscokid | well there is a punchdown block |
| 04:23.25 | rob0 | Zimbra! |
| 04:23.58 | kiscokid | what about the DID? Does it have a separate cable? |
| 04:24.48 | kiscokid | or does the DID just control which extensions can be called on the 5 analog lines? |
| 04:25.11 | JT | since it's analogue, probably seperate |
| 04:25.17 | JT | as analogue has crap signalling |
| 04:25.30 | JT | distinctive ringing could also do the trick, but it's hackish |
| 04:26.39 | kiscokid | don't think we're using distinctive ringing since we have 20 extensions |
| 04:28.43 | JT | does each extension have a different did? |
| 04:29.20 | kiscokid | not sure I understand the question. Each extension has a different number |
| 04:29.58 | JT | then the answer is yes |
| 04:30.09 | JT | if that's analogue, i have no idea how it's done |
| 04:30.14 | JT | maybe some strange line signalling |
| 04:30.35 | [TK]D-Fender | JT, DTMF on initial connect prior to bridge |
| 04:30.37 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 04:30.46 | JT | [TK]D-Fender: nasty |
| 04:30.56 | JT | [TK]D-Fender: does it have a proper name? and does ast support it? |
| 04:30.58 | [TK]D-Fender | JT : ANALOG ;) |
| 04:31.13 | kiscokid | how would fractional PRI help? |
| 04:31.20 | JT | analog is a horrible abomination of "analogue" :) |
| 04:31.25 | [TK]D-Fender | JT : yeah * would support it. basically your IVR has got to grab it fast |
| 04:31.31 | JT | kiscokid: digital signalling is simple to work with and reliably |
| 04:31.54 | [TK]D-Fender | JT : it's called a homonym ;) |
| 04:32.42 | Strom_C | homophones are words which sound alike but suck each other off in the men's restroom |
| 04:34.09 | [TK]D-Fender | kiscokid, like I told you yesterday, diconnect 1 line from the PBX and hook an analog line on and LISTEN. |
| 04:34.41 | kiscokid | ok, what would I listen for? DTMF? |
| 04:34.52 | JT | jesus |
| 04:34.54 | [TK]D-Fender | kiscokid, exactly |
| 04:35.27 | Strom_C | how to work with asterisk |
| 04:35.28 | Strom_C | step 1: |
| 04:35.32 | Strom_C | turn your brain on |
| 04:36.02 | kiscokid | Fender: what would I have to connect to the line? An analog phone? |
| 04:36.05 | rob0 | I *knew* I forgot something. |
| 04:36.14 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 04:36.20 | waKKu | hi folks.. good evening |
| 04:36.46 | waKKu | can someone explain to me: wtf a new exten i had create doesnt work with Playback app ? |
| 04:37.20 | waKKu | this says me that doesnt locate carried-away-by-monkeys in any format... (i use this file in ALL my dialplan and works fine) |
| 04:37.20 | rob0 | Because you did something wrong. :) |
| 04:37.26 | waKKu | no rob0 ;/ |
| 04:37.38 | rob0 | paste the line here |
| 04:37.45 | waKKu | ok.. pastebin |
| 04:37.52 | rob0 | just one line, can put it here |
| 04:38.03 | rob0 | exten => |
| 04:38.40 | rob0 | exten => 0,1,Playback(tt-monkeys) |
| 04:39.37 | rob0 | That's the name, "tt-monkeys". Do you have one called "carried-away-by-monkeys"? |
| 04:39.54 | waKKu | rob0 http://pastebin.ca/578061 |
| 04:40.06 | waKKu | i already try another file too |
| 04:40.07 | rob0 | My "carried-away-by-monkeys" was eaten by weasels. |
| 04:40.15 | [TK]D-Fender | kiscokid, Yes |
| 04:40.41 | waKKu | the only playback line is: exten => 19062007,2,Playback(health-center) |
| 04:40.57 | waKKu | 2 lines down has other playback that not work too |
| 04:41.36 | JT | try Playback(tt-monkeys) instead? |
| 04:41.48 | [hC] | waKKu: by the looks of it that sound file does not exist or has incorrect permissions in your sounds directory |
| 04:41.49 | waKKu | no.. but try beep, good, goodbye |
| 04:41.59 | waKKu | let me check |
| 04:42.05 | kiscokid | Fender: this is an unknown area for me. When I plug the phone into this line will it act like a regular analog line like I have at home? Will I get dialtone? |
| 04:42.08 | [hC] | waKKu: which should be /var/lib/asterisk/sounds by default i believe |
| 04:42.44 | [hC] | you can check what directory its using in asterisk.conf, look for 'astvarlibdir' it should be /var/lib/asterisk |
| 04:42.54 | [hC] | then the sounds directory is 'sounds' after that |
| 04:42.58 | [hC] | and asterisk needs to be able to read the file. |
| 04:43.20 | JT | kiscokid: yes, just plug it in already |
| 04:44.17 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 04:44.46 | kiscokid | JT: well, I'm not at work so I can't. Also, I have to figure out how to make a cable that goes from a punchdown block to an rj-11 socket |
| 04:45.40 | waKKu | yeah. |
| 04:45.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:45.47 | JT | kiscokid: it's only 2 wires |
| 04:47.06 | kiscokid | JT: ok lets say I get it connected to a phone. Then I call a DID with my cellphone. What will happen with this analog phone? Will it ring? When I pcik it up will I hear the DTMF? |
| 04:47.51 | JT | that's what you'll find out |
| 04:48.12 | JT | if it's all the one group of lines, you'll probably want to try line 1 |
| 04:48.51 | *** join/#asterisk gardo (n=gardo@121.97.198.153) |
| 04:48.55 | kiscokid | JT: I guess you mean line 1 of the hunt group |
| 04:50.11 | JT | yes |
| 04:50.31 | JT | i shouldn't need to spell out every little obvious detail :) |
| 04:51.11 | kiscokid | JT: If I hear the DTMF would that imply that a digium or other analog card would properly handle the DID ? |
| 04:52.10 | kiscokid | (hopefully once I learn about this I'll be able to help other folks on here) |
| 04:52.55 | JT | kiscokid: [TK]D-Fender was saying you should drop it into an IVR and the IVR would be able to pick up the dtmf, if that's how it's done |
| 04:53.13 | JT | i'd be inclined to get rid off the analogue crap and replace it with digital myself |
| 04:53.18 | JT | s/off/of/ |
| 04:53.47 | JT | i really hate analogue phone lines for pbxes, especially if it's for a business |
| 04:54.29 | Strom_C | for small numbers of lines, ISDN BRI is the way to go if you're not in north america :) |
| 04:56.12 | JT | Strom_C: fractional 10ch PRI is cheaper than about 5 channels (2-3BRIs) worth of connectivity here, if you can get an Optus PRI :) |
| 04:56.27 | Strom_C | ah, ok |
| 04:56.31 | JT | but yes |
| 04:56.42 | Strom_C | ....I have an Optus SIM |
| 04:56.44 | JT | BRI is an excellent filler for the lower numbers |
| 04:56.47 | Strom_C | their customer service is crap |
| 04:56.55 | JT | heh |
| 04:56.55 | Strom_C | for prepaid anyway :) |
| 04:57.09 | JT | they're better than telstra, for pri |
| 04:57.19 | JT | telstra is the only one offering bri, hence the expense |
| 04:57.26 | Strom_C | from what I'm told, Telstra is the GTE of Australia |
| 04:57.29 | waKKu | damn.. |
| 04:57.45 | JT | not sure what gte is, but the answer is probably yes :) |
| 04:57.48 | waKKu | this is using /usr/share/asterisk/sounds instead of /var/lib/asterisk/sounds (defined varlibdir = /var/lib/asterisk) |
| 04:58.10 | JT | also, call costs through telstra are substantially higher than optus |
| 04:58.19 | waKKu | the only entry for /usr/share/asterisk is agidir = /usr/share/asterisk/agi on asterisk.conf |
| 04:58.21 | waKKu | weird |
| 04:58.28 | Strom_C | waKKu: you installed from a package, didn't you |
| 04:58.29 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 04:58.42 | waKKu | Strom_C dunno.. wasnt me |
| 04:58.47 | Strom_C | lame |
| 04:59.03 | JT | Strom_C: gte? |
| 04:59.13 | waKKu | but i guess no.. /usr/src/asterisk* exists |
| 04:59.20 | Strom_C | JT: General Telephone & Electronics |
| 04:59.30 | Strom_C | JT: the crappy north american phone company |
| 04:59.30 | JT | ah |
| 04:59.55 | JT | were they an incumbent provider? |
| 04:59.58 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 05:06.14 | Strom_C | they were the independent telco |
| 05:06.22 | Strom_C | s/the in/the largest in/ |
| 05:07.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:08.33 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 05:09.34 | JT | Strom_C: telstra was the national government run telco, but has gradually been privatised |
| 05:09.44 | Strom_C | yes, I know :) |
| 05:10.14 | JT | ok :) |
| 05:11.59 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 05:13.32 | JT | Strom_C: optus do fractional PRIs starting at AUD$20/ch/mo, minimum 10 channels |
| 05:13.39 | JT | it's quite competitive :) |
| 05:18.54 | *** join/#asterisk iratsu (n=iratsu@modemcable090.239-59-74.mc.videotron.ca) |
| 05:25.50 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:27.33 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 05:28.04 | Splat | JT, you do know you can get BRI and PRI from Verizon in Australia too don't you? I don't know how they compare to Optus pricing.. but they are another option I like their local call rate.. since it's cheaper then the cheapest VoIP rates I've ever found.. heh |
| 05:28.43 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 05:35.19 | JT | Splat: bri, it's probably resold telstra bri |
| 05:35.37 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 05:35.55 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 05:37.04 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 05:37.40 | *** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net) |
| 05:37.53 | RyanW | what does "Got a FRAME_CONTROL (5)" mean ? |
| 05:40.30 | waKKu | argh.. i cant use System(); ... i solve two possible problems: asterisk is running with -U asterisk, i give a shell bash to user asterisk - and i set permissions to asterisk runs these shellscript... can someone help me with it ? |
| 05:41.06 | JT | Splat: cheaper than the cheapest voip rates, with what sort of minimum spend? :) |
| 05:41.17 | waKKu | this shows me on CLI running... but scripts just make a "echo teste > /tmp/works.log" |
| 05:41.40 | waKKu | and doesnt work.. i ran it manually and works ok |
| 05:41.49 | inv_arp[work] | hmm how to slow this registration down... sip.inphonex.com:5060 6762000 20 Registered |
| 05:41.59 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 05:42.08 | inv_arp[work] | every 20sec.. |
| 05:43.53 | *** part/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 05:45.46 | waKKu | forget.. again.. |
| 05:48.30 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 05:52.47 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 05:58.02 | *** join/#asterisk saftsack (n=oliver@p54A7F622.dip.t-dialin.net) |
| 06:13.41 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 06:16.30 | *** join/#asterisk saftsack (n=oliver@p54A7CD63.dip.t-dialin.net) |
| 06:18.16 | *** join/#asterisk Nuitari (n=Nuitari@142.46.207.230) |
| 06:18.44 | Nuitari | Hi |
| 06:19.10 | Nuitari | Is there a way to have SIP hints w/o limiting the number of calls someone can have? |
| 06:24.31 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 06:26.20 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:27.49 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 06:29.24 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 06:30.11 | Splat | jt: on a BRI around $62.52 +GST per month with 8.8 cent local calls and the qualifier was the cheapest VoIP rates that *I* have found.. the main thing was I needed the lines and they had better rates then Telstra so I ended up only needing to put STD calls through VoIP.. heh |
| 06:30.45 | Nuitari | naughty... |
| 06:31.04 | JT | naughty? |
| 06:31.17 | Nuitari | STD... |
| 06:31.17 | JT | Splat: that's telstra resold |
| 06:31.27 | JT | Splat: expensive line charges |
| 06:31.37 | JT | Nuitari: subscriber trunk dialling |
| 06:32.49 | creativx | or... |
| 06:32.50 | creativx | sexually.. |
| 06:32.52 | creativx | ehh nevermind |
| 06:32.59 | Nuitari | finally someone gets it |
| 06:33.18 | Splat | JT: I figured as much.. for the simple fact that no one other then telstra actually has anything in the Collinsvale exchange.. so that left Verizon as a better deal then telstra.. heh |
| 06:33.21 | Nuitari | is there any way to make sip presence work in 1.4 w/o limiting how many calls someone can do ? |
| 06:33.26 | JT | i knew what you were implying, it just wasn't very funny :P |
| 06:33.47 | JT | Splat: i assume you don't need that high a call volume |
| 06:33.54 | JT | since it's just rebilled |
| 06:34.16 | JT | Splat: optus is still king of line rental costs, if you can get them |
| 06:34.52 | Splat | nah.. that was basically just so I could play with asterisk on isdn.. heh |
| 06:35.21 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:35.32 | JT | Splat: 10chs with optus is cheaper than 6 with telstra |
| 06:36.44 | Splat | yeah.. most things are cheaper then telstra.. it's just that optus doesn't really have anything outside of the CBD in Hobart.. |
| 06:37.34 | JT | ah damn |
| 06:38.02 | JT | you could chuck a pri into a datacentre there, but then you're betting on your Internet link |
| 06:38.35 | JT | Splat: what are the costs like for other verizon calls? |
| 06:38.43 | *** join/#asterisk saftsack (n=oliver@p54A7D363.dip.t-dialin.net) |
| 06:39.47 | Splat | I wouldn't need a PRI in my house.. well not anytime soon.. maybe if I had 3 teenage girls in the house I'd need a PRI to be sure I can make or receive a phone call.. :P |
| 06:40.57 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 06:41.12 | JT | ah, i didn't know this was just for your house :P |
| 06:41.50 | IOscanner | I am having an issue with asterisk if I just run ps -ef or ls -al on a large directory the audio stream breaks up. |
| 06:41.56 | IOscanner | I tried to renice to -20 |
| 06:42.00 | IOscanner | still does the same thing |
| 06:42.09 | IOscanner | any ideas what is causing this |
| 06:42.15 | IOscanner | I have plenty of memory |
| 06:42.18 | IOscanner | and CPU |
| 06:43.32 | webman | IOscanner: what filesystem and HDD are you using, and what parameters do you start asterisk with? |
| 06:43.44 | Splat | JT, didn't have anywhere else to play with it at the time.. of course now there's on in the office where I supposedly work but haven't been to for ages.. heh and I got rid of the one I had here.. for now.. until I have a real use for one.. heh |
| 06:44.06 | IOscanner | Using sata II |
| 06:44.15 | flenders | splat: I have 7 POTS line with tesltra, which cost us 34.50/line |
| 06:44.17 | IOscanner | Segate 250 .NCQ |
| 06:44.28 | flenders | splat: a 10ch PRI with optus is 200 |
| 06:44.34 | IOscanner | -f |
| 06:45.06 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:45.09 | webman | IOscanner: try to add -p to asterisk startup.... you still didn't specify what filesystem you use ... |
| 06:45.34 | IOscanner | ext3 |
| 06:45.35 | IOscanner | sorry |
| 06:45.37 | IOscanner | what is -p |
| 06:46.08 | Splat | yeah.. $100 a month cheaper then Verizon for 10 channels.. what are their local call rates like? |
| 06:46.13 | webman | gives asterisk realtime priority (if you start it as root), so if asterisk screwwwws up then your whole box can be hosed |
| 06:46.58 | webman | but it should also give better performance for asterisk, which is usually what you want :) |
| 06:47.46 | IOscanner | nope still the same thing |
| 06:49.29 | webman | IOscanner: how many files in the directory |
| 06:50.04 | IOscanner | 50 or so |
| 06:50.09 | webman | flenders: any idea what optus call their PRI service, I can't see it on the website |
| 06:50.10 | IOscanner | even when I start top |
| 06:50.39 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 06:50.40 | IOscanner | pretty much anything I do on the command line causes the audio stream to have problem |
| 06:50.48 | JT | Splat: depends on your deal, but around 10-13ex GST untimed |
| 06:50.49 | IOscanner | I am using ZAP lines only |
| 06:50.54 | IOscanner | I have 8 lines in this box |
| 06:50.59 | JT | Splat: Multiline |
| 06:51.02 | IOscanner | so it is not a bandwidth issue |
| 06:51.17 | IOscanner | I thought it might be, but it seem to be on the system level. |
| 06:51.36 | webman | IOscanner: 50 isn't a lot ... I have a directory with over 5000 files, and don't have a problem (but I do use reiserfs ) |
| 06:51.44 | JT | webman: i meant that for you, Multiline |
| 06:51.46 | IOscanner | nope |
| 06:51.53 | IOscanner | I have never seen this before |
| 06:52.02 | IOscanner | I am running 1.2.18 |
| 06:52.06 | webman | IOscanner: possibly CPU related then.... or IRQ problems (missing IRQ's) |
| 06:52.26 | webman | IOscanner: what kernel version? have you checked with zttest ? |
| 06:52.33 | webman | JT: thanks.... |
| 06:52.42 | IOscanner | 2.6.18 |
| 06:52.53 | IOscanner | nope maybe I will try that |
| 06:53.03 | IOscanner | I am rebuilding the kernel for full 64bit support |
| 06:53.25 | justdave | since asterisk 1.4.x has this nice menuselect thing now, what file do I need to move from one source tarball to the next when upgrading to preserve my config choices? |
| 06:54.46 | IOscanner | zttest shows all 100% and a few 99.98% |
| 06:54.50 | webman | IOscanner: something is being starved of resources, you need to find out which resource, either disk IO, memory, CPU, or something, and then fix it.... I've never had that problem, so I'm not entirely sure how to fix it.... |
| 06:55.07 | webman | IOscanner: what card are you using? |
| 06:55.24 | IOscanner | TDM400p |
| 06:55.39 | IOscanner | 2 with 4 modules each FX) |
| 06:55.45 | IOscanner | FXO |
| 06:56.09 | webman | IOscanner: try removing one card and see what happens |
| 06:56.33 | IOscanner | I can't the box is not with me. |
| 06:56.37 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:56.37 | IOscanner | I am doing this from remote |
| 06:57.08 | webman | IOscanner: I thought you said it was all ZAP channels, how do you connect to the ZAP if you are remote? |
| 06:57.11 | JT | webman: to get a quote, basically ring optus business direct, punch in your post code, and speak to the channel partner for your area |
| 06:57.34 | IOscanner | I am testing with a single IAX client |
| 06:57.42 | IOscanner | same results with a local phone |
| 06:57.58 | webman | JT: am waiting to speak to them now... thanks.... I mainly just want to move the line rental to them, as all my calls are either preferred or prefix dialled to another company already |
| 06:58.27 | JT | webman: how many lines do you have? |
| 06:58.41 | webman | JT: 10 channel PRI with telstra at the moment |
| 06:58.52 | JT | webman: ah, $365/mo? |
| 06:59.06 | webman | JT: something like that |
| 06:59.23 | JT | webman: optus pris have free installation on 24mo contracts |
| 06:59.37 | JT | and they will probably give you a credit if you negotiate one |
| 06:59.47 | flenders | JT: optus hates you mate |
| 06:59.48 | flenders | :D |
| 06:59.52 | JT | the main disadvantage is the lead time |
| 06:59.57 | JT | flenders: :P |
| 07:00.06 | flenders | but on his case, PRI is already there |
| 07:00.10 | webman | JT: can they "take" a telstra PRI? |
| 07:00.31 | JT | webman: it'd need to be reterminated to their own exchange gear for the line rental savings |
| 07:00.42 | JT | i'm sure they can rebill, you just won't save on line rentall |
| 07:01.26 | flenders | JT: why the cabler who installs the E1s for optus is from telstra? |
| 07:01.50 | webman | oh well, I can just wait for the new install I suppose and then cancel the telstra one later |
| 07:01.59 | flenders | optus equipament on telstra exchange? |
| 07:02.02 | JT | flenders: it's telstra copper |
| 07:02.07 | JT | telstra own all the copper |
| 07:02.23 | JT | they just tag the line really, from what i can tell |
| 07:02.44 | JT | flenders: yes, optus equipment inside the telstra exchange |
| 07:02.54 | flenders | gotcha |
| 07:02.57 | JT | webman: well optus will port your numbers for free |
| 07:03.02 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 07:03.24 | webman | JT: good, gotta keep my numbers, it ends in 0000 which is nice :) |
| 07:03.37 | JT | heh |
| 07:03.37 | flenders | JT: optus told me today that if I wanted to switch our ADSL2 lines to them, we wouldn't pay for line rental on those lines |
| 07:03.45 | JT | it can take up to 10 weeks though, webman |
| 07:03.49 | JT | from signing of contract |
| 07:03.51 | JT | to porting |
| 07:04.02 | JT | telstra drag their feet on porting |
| 07:04.07 | KpoH | is it correct to use chan_local in IVR? |
| 07:04.13 | KpoH | to forward calls |
| 07:04.17 | KpoH | like this |
| 07:04.20 | KpoH | exten => 1,1,Dial(Local/7094@local/n|30|tTm) |
| 07:04.38 | KpoH | @local is one of my contexts |
| 07:05.11 | KpoH | i did so because i need to bill users for callforward |
| 07:05.16 | webman | JT: yeah, I really don't mind about the timeframe, as long as actual downtime would be fairly minimal..... I've been paying lots for a couple of years now, so a few more months probably won't matter |
| 07:05.33 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 07:05.34 | KpoH | but it seems this is not working |
| 07:05.52 | webman | JT: neway, optus say they will callback within 48 hours.... so I'll see what they have to offer :) |
| 07:06.03 | flenders | KpoH: why don't you use account codes? |
| 07:06.36 | flenders | webman: after 5PM |
| 07:06.38 | KpoH | flenders: what you mean? Set(CDR(accountcode))? |
| 07:06.44 | JT | webman: 5mins down time for each phone number |
| 07:06.51 | JT | they port them one at a time |
| 07:06.57 | flenders | KpoH: yeah, wouldn't it work for you? |
| 07:07.17 | flenders | then on your Master.cvs file you would get all calls related to that account code |
| 07:07.46 | webman | JT: hmmm, a 100 number range, thats 500 minutes... looks like it will take them all night |
| 07:07.53 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 07:08.09 | *** join/#asterisk jmacz (n=jmacz@190.24.97.36) |
| 07:08.12 | JT | webman: maybe they can more a 100number range faster |
| 07:08.18 | JT | i was talking about individual numbers |
| 07:08.24 | JT | move |
| 07:08.45 | flenders | our porting (100 numbers) will take minutes |
| 07:08.51 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 07:08.57 | flenders | it'll happen next thursday night |
| 07:09.01 | webman | JT: yeah, I'd like to think so, but I'm sure telstra will manage to slow things down or stuff something up :) |
| 07:09.13 | flenders | start at 5PM and they reckon it'll be finished by 5:30PM |
| 07:09.14 | JT | flenders: were they on bri? |
| 07:09.18 | flenders | yeah |
| 07:09.19 | webman | flenders: you are in the process of moving to optus? |
| 07:09.33 | flenders | webman: yeah, 4 BRIs to 1 PRI |
| 07:09.40 | JT | flenders and I both have multiple sites getting optus pris in ;) |
| 07:09.45 | JT | <3 optus |
| 07:09.53 | JT | ok, maybe that's a bit strong there :P |
| 07:10.43 | webman | I wonder if optus can move my optus cable analog service onto the same PRI (101 numbers on the PRI) ?? |
| 07:10.43 | flenders | JT: have you heard from them on yours? |
| 07:11.23 | JT | flenders: next couple of weeks is what i was told |
| 07:11.28 | JT | webman: yes they can |
| 07:11.51 | flenders | webman: they can port any number to the PRI |
| 07:11.56 | JT | they move up to 1 number per channel for free, then you have to start paying for 100 number ranges |
| 07:11.58 | flenders | well, not mobiles, I think |
| 07:12.11 | JT | flenders: heh, that'd be awesome if they could move mobiles ;) |
| 07:12.19 | webman | kewl, I can dump my old analog line as well :) |
| 07:12.48 | flenders | JT: yeah, I was already pretty happy that they can change the ownership of mobile numbers at anytime, so you can have you personal number on the company account |
| 07:13.07 | JT | webman: i'd always keep one as a backup |
| 07:13.17 | JT | remember a pri only runs over 1 or 2 pairs |
| 07:13.20 | webman | hmmm, I wonder if that means they will charge me to port the extra 90 numbers from my 100 number range.... |
| 07:13.26 | JT | if you lose it, buy buy phones |
| 07:13.32 | KpoH | realy strange...why IVR instead of some reaction write's in cli -- Attempting native bridge of SIP/7325928432-b7222120 and SIP/USA Route-160-082de1b0 |
| 07:13.34 | *** join/#asterisk corpcomp (n=IceChat7@125-238-120-174.broadband-telecom.global-gateway.net.nz) |
| 07:13.40 | JT | webman: $36.50inc gst, 100number range |
| 07:13.40 | KpoH | somethink wrang with DTMF |
| 07:13.53 | KpoH | but what it chould be? |
| 07:14.07 | JT | s/buy buy/bye bye/ |
| 07:14.18 | webman | JT: true... though I have a cable that runs direct from the exchange into the office... (they ran out of pairs in the street, so they ran a new cable for me) |
| 07:14.36 | JT | a lot of buildings have that, webman |
| 07:14.38 | JT | ours does |
| 07:14.44 | JT | they still manage to f*ck it up |
| 07:14.54 | JT | idiot techs at the exchange dejumpering shit |
| 07:15.20 | JT | i've had if happen at least twice |
| 07:15.36 | JT | also, POTS is the most reliable thing in a blackout |
| 07:16.12 | flenders | JT: PRIs arent? |
| 07:16.17 | JT | flenders: nup |
| 07:16.25 | flenders | JT: why not? |
| 07:16.29 | JT | especially since most of us stick them in power hungry PCs |
| 07:16.38 | JT | because they aren't powered from the exchange |
| 07:16.43 | webman | JT: well, if we really lost the PRI, analog lines wouldn't help, would just use mobile phones :) we don't have very high call volumes anyway |
| 07:16.51 | JT | and you can't use simple telephones on them |
| 07:16.59 | flenders | JT: ok, but as long as you're on a decent UPS, you're fine, no? |
| 07:17.07 | JT | webman: i'm thinking more for an emergency situation too |
| 07:17.21 | JT | webman: also if you have a fax, or modems, it's easier |
| 07:17.33 | JT | flenders: i bet your ups can't keep up with a telstra exchange |
| 07:17.55 | flenders | JT: at the travel agent, we'll have a TDM400 with FXS modules on it for fax and eftpos |
| 07:18.06 | flenders | JT: I bet it too |
| 07:18.10 | JT | banks of 2V @ 500-1000Ah lead acid batteries |
| 07:18.11 | webman | JT: I want to move all my fax to iaxmodem and hylafax |
| 07:18.19 | JT | and backup generators that kickin within minutes |
| 07:18.27 | JT | flenders: hopefully it will work.... |
| 07:18.40 | flenders | JT: I can't see why it wouldn't |
| 07:18.50 | JT | haven't heard eftpos success stories yet |
| 07:18.53 | JT | in theory it should |
| 07:19.06 | flenders | eftpos on FXS? |
| 07:19.06 | JT | but i assume everything doesn't work with asterisk until proven otherwise |
| 07:19.07 | *** join/#asterisk dijungal (n=chatzill@209.59.110.5) |
| 07:19.07 | webman | flenders: I use a TDM card with a fax machine (outbound) + eftpos, works pretty well |
| 07:19.35 | flenders | webman: couple of FXS modules? |
| 07:19.51 | webman | I configured a new context for the eftpos/fax so that you didn't need to dial 9 to get a external dialtone :) |
| 07:20.10 | flenders | over here we don't dial anything to dial out |
| 07:20.19 | webman | flenders: yeah, I think it is one FXO and 3 x FXS |
| 07:20.23 | *** join/#asterisk SwK_ (n=SwK@user-69-73-37-99.knology.net) |
| 07:20.35 | dijungal | has anyone successfully connected a inbound H.323 to asterisk? |
| 07:20.39 | flenders | people like to dial like they do at home or their mobiles |
| 07:20.45 | JT | flenders: also, if anything in your setup fails, bye bye phones |
| 07:20.51 | JT | analogue phone = very simple |
| 07:20.52 | flenders | JT: true |
| 07:21.06 | JT | px based pbx + csu/dsu + digital = complex ;) |
| 07:21.14 | JT | s/px/pc/ |
| 07:21.32 | webman | flenders: but how do you deal with internal extension numbers and escaping them from the public dialplan/numbering scheme? |
| 07:21.36 | flenders | a single pots line wouldnt hurt |
| 07:21.37 | flenders | I know |
| 07:21.58 | flenders | webman: different contexts? |
| 07:22.18 | webman | flenders: but your users only exist in a single extension |
| 07:22.30 | flenders | all public dialplans have to have at least 8 digits |
| 07:22.31 | webman | s/extension/context/ |
| 07:22.39 | *** join/#asterisk Diablus (n=bth@217.115.85.18) |
| 07:22.46 | dijungal | i am trying to connect the H.323 calls from a cisco 3661 router to an asterisk box... i know asterisk can interconnect with cisco but it's not working out for me... any ideas? |
| 07:22.53 | webman | flenders: so you rely on a digittimeout |
| 07:22.55 | JT | webman: you don't need outbound dialling prefixes if your internal extensions don't interfere with pstn numbers |
| 07:22.58 | flenders | include the outgoing context |
| 07:23.15 | JT | flenders: 13XXXX is 6 digits, not 8+ |
| 07:23.23 | JT | 000 is 3 digits |
| 07:23.25 | webman | JT: but how can you be sure that you won't conflict |
| 07:23.26 | flenders | damn you JT! |
| 07:23.42 | JT | webman: you only allow certain prefixes out to pstn |
| 07:23.43 | flenders | those ones are on the dialplan |
| 07:23.49 | webman | there are also some weird special numbers like 1xxx |
| 07:24.03 | flenders | 1234? |
| 07:24.08 | JT | webman: emergency, 13/1300, 1800, local, national, international |
| 07:24.12 | webman | flenders: yeah, that crap |
| 07:24.23 | JT | webman: yes but if your staff don't need to dial them, don't add them to the pbx dialplan |
| 07:24.29 | flenders | no one ever complained here. |
| 07:24.29 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:24.42 | flenders | and you know they cost you money too? |
| 07:24.42 | JT | webman: and you can be sure if you use the national numbering plan |
| 07:24.50 | JT | i always consult the national numbering plan |
| 07:24.53 | flenders | so, they can google for the number they want |
| 07:24.56 | JT | it's what the telcos refer to |
| 07:25.49 | flenders | JT: damn, the other guys are late for the meeting |
| 07:25.52 | webman | how about 1802288 or 12552 or 1223 |
| 07:26.08 | JT | webman: do they do anything useful? |
| 07:26.11 | JT | flenders: ah well |
| 07:26.18 | flenders | I don't even know what they're for! |
| 07:26.23 | webman | one of those is the telstra number for reporting ADSL faults |
| 07:26.36 | flenders | webman: we don't use telstra ADSL |
| 07:26.37 | JT | add it to the dialplan then ;) |
| 07:26.41 | webman | 1223 is directory assistance (I think local call cost/free if it still works) |
| 07:26.57 | flenders | webman: google/yellow pages |
| 07:27.12 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 07:27.22 | JT | webman: most businesses these days don't allow unrestricted raw access to the pstn to staff |
| 07:27.28 | webman | anyway, the question remains, what prefix is not used by the PSTN.... AFAIK, every number has a use |
| 07:27.32 | JT | webman: anyway, you can write override prefixes too |
| 07:27.40 | JT | to send stuff raw |
| 07:28.01 | JT | webman: national numbering plan, everything is detailed there |
| 07:28.13 | webman | JT: we don't allow raw access... well, except for me :) |
| 07:28.17 | JT | you have to match by prefix anyway if you want to implement any form of least cost routing |
| 07:29.09 | webman | JT: I read that ages ago... from memory, there wasn't an available range.... thats why I am wondering how you have managed it? |
| 07:29.23 | mvanbaak | heya all |
| 07:29.25 | mvanbaak | app_queue.c: No one is answering queue |
| 07:29.30 | mvanbaak | what does that mean ? |
| 07:29.41 | JT | webman: i use 0 for an outside line personally |
| 07:29.49 | JT | but it doesn't give raw pstn access |
| 07:29.50 | webman | mvanbaak: that there is someone in the queue who is not getting answered |
| 07:29.59 | JT | for that you need to dial a special prefix and type in a pin :) |
| 07:30.16 | flenders | JT: funny that on the national numbering plan, they drop the '0' from most numbers |
| 07:30.32 | JT | flenders: ? |
| 07:30.34 | flenders | and say '9 digit number' for a sydney local number, for example |
| 07:30.37 | mvanbaak | webman: this user will stay in the queue waiting for an agent to pickup ? |
| 07:30.41 | webman | JT: OK, so that is basically the same as here... btw, does "000" work, or do you need "0000" |
| 07:30.41 | JT | ah |
| 07:30.45 | Shaun2222 | anybody use the iaxy s101 unit, can i program it with iaxyprov from a remote network? |
| 07:30.48 | JT | webman: 0000 |
| 07:30.58 | JT | webman: everyone knows you need 0 for an outside line |
| 07:31.07 | JT | it's the same as hitting a line key |
| 07:31.08 | flenders | 2 9 (NDC)9 digits9 digitsGeographic number for fixed network telephone serviceMetro: Sydney |
| 07:31.11 | JT | we use an isdn key system too |
| 07:31.16 | mvanbaak | I just want to know wether this can be a problem |
| 07:31.20 | mvanbaak | like loosing calls |
| 07:32.08 | webman | mvanbaak: it depends on your configuration (timeout options configured) and how long they are left in the queue, I think the default is for them to wait forever |
| 07:32.34 | webman | JT: what about a guest/visitor? |
| 07:32.39 | mvanbaak | webman: thanks |
| 07:32.41 | JT | webman: hmm? |
| 07:32.55 | mvanbaak | they will stay forever in that queue indeed (if I look at the configs) |
| 07:33.06 | JT | 0 is the standard in australia for getting an outside line |
| 07:33.10 | mvanbaak | $some_company hired me to find the fuckups in their asterisk setup |
| 07:33.13 | JT | they can hit a line key instead anyway |
| 07:33.29 | mvanbaak | some polish dudes configured this machine but they are missing many calls and they have some other trouble |
| 07:33.40 | mvanbaak | this logline I never saw on any of our own boxen |
| 07:33.43 | mvanbaak | that's why I'm asking |
| 07:33.55 | webman | JT: some people have asked me to use 9 as the prefix .... some want 0.... derpends on what they had before usually |
| 07:34.09 | JT | 1heh |
| 07:34.32 | JT | you can't make 0 then 00 call 000 anyway |
| 07:34.36 | JT | 0011 is the idd code |
| 07:35.32 | webman | JT: yeah, I think the way around that one was to setup a 'internal' extension that required a password, before you could access DISA to dial your 0011 numbers :) |
| 07:35.40 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 07:35.57 | JT | there may be some other 00 prefixes too |
| 07:36.17 | JT | the solution is to not let idiots use your phone system :) |
| 07:36.31 | webman | jt: AFAIK, all the 00xx prefixes are for international access .... |
| 07:36.49 | JT | there used to be 0055, maybe there's other non idd codes |
| 07:37.15 | JT | our phone system is heaps easy to use compared to say japan |
| 07:37.30 | webman | JT: personally, I might argue that if you are too dumb to dial 0000 or 9000 in an emergency, then you don't deserve to survive (darwin theory) but the problem is I might be the one in trouble and needing someone else to call 000 for me :( |
| 07:38.03 | JT | heh, they can use a mobile surely |
| 07:38.15 | JT | or you have instructions next to your phones on getting an outside line |
| 07:38.17 | webman | JT: I think the other codes like 0055 were converted to 1500 xxx xxx or something... |
| 07:38.24 | JT | 1900 mostly |
| 07:38.53 | webman | yeah, those too... but I always ban those on the telstra side as well :) |
| 07:38.57 | JT | in japan to call overseas, you need to dial 011 <carrier access code 010 |
| 07:39.13 | JT | so to use NTT, 011 033 010 + country code + number |
| 07:39.32 | JT | it's harder to work out when the error messages are in japanese ;) |
| 07:40.28 | webman | JT: but wouldn't the PBX 'auto-dial' the carrier access code and the 010 for you? |
| 07:40.46 | JT | webman: pay phones aren't on a pbx :/ |
| 07:41.00 | JT | which is the type of phone a foreigner is likely to try |
| 07:41.11 | webman | JT: payphones?? who still uses those things :) |
| 07:41.20 | webman | JT: true |
| 07:41.28 | *** join/#asterisk angryuser (n=aster@df01t2-213-44-148-16.d4.club-internet.fr) |
| 07:41.55 | JT | webman: umm, most australian mobile phones will not work in japan |
| 07:42.34 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 07:42.40 | JT | i brought 2 different phones, i knew they would not work |
| 07:42.43 | JT | and they didn't :P |
| 07:44.59 | snuffy22 | long as they are tri-band generally u wont have any problems |
| 07:45.12 | JT | tri band gsm would be completely useless |
| 07:45.31 | JT | it would need to be 3G AND have roaming to japan enabled by your carrier before you leave |
| 07:46.16 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 07:46.19 | zeeesh | hi all |
| 07:48.14 | *** join/#asterisk CelticSoul (n=CelticSo@bne75-1-81-57-10-55.fbx.proxad.net) |
| 07:53.21 | JT | (you can't actually buy a phone sim in japan as a non resident) |
| 07:53.54 | *** join/#asterisk vgster (n=vgster@h146106.navonline.net) |
| 08:02.05 | dijungal | anyone has any experience with ooh323 on asterisk |
| 08:02.19 | dijungal | i'm trying to get asterisk to pickup incoming H323 calls |
| 08:02.48 | dijungal | but when the calls come in the CLI does not even budge... no debug... it's like asterisk didn't even see the call |
| 08:02.51 | Strom_C | the H stands for Headache |
| 08:03.27 | JT | only in astland ;) |
| 08:03.41 | dijungal | now a tcpdump shows me that the call packets are actually getting to the box... asterisk just sits there like a big dummy doing nothing to take the call :) |
| 08:04.30 | dijungal | i've been wondering over the net for days... looking at all different h323 configurations... still no help |
| 08:05.31 | JT | asterisk really sucks with H.323 |
| 08:06.05 | dijungal | oooh that's great news... i know that... but i also know in certain cases it works |
| 08:06.45 | *** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net) |
| 08:10.12 | dijungal | so i guess that's it.. "asterisk really sucks with H.323" |
| 08:12.03 | *** join/#asterisk yassaccan (n=yassacca@admin146.hgo.se) |
| 08:12.33 | sergee | dijungal: i have asterisk in production converting SIP to H.323... no problems, |
| 08:13.15 | dijungal | i need to do that.. SIP <-> H.323 for outgoing and opposite for incoming |
| 08:13.36 | dijungal | i'm more interested in the incoming right now... H.323 -> SIP |
| 08:14.06 | sergee | dijungal: which instance of h323 did you try? |
| 08:14.23 | dijungal | ooh323 |
| 08:14.30 | dijungal | the one that come with asterisk-addons |
| 08:14.55 | dijungal | asterisk-addons-1.2.6_1.2.18-1 |
| 08:15.25 | sergee | dijungal: chan_h323 from asterisk (trunk or 1.4) works well, it was fixewd by PCadach |
| 08:15.51 | sergee | dijungal: asterisk-trunk/channels/h323/ |
| 08:16.39 | dijungal | how do i get that installed? |
| 08:17.36 | sergee | dijungal: cd /usr/src |
| 08:17.54 | walhala | io |
| 08:17.55 | dijungal | uhuh |
| 08:19.05 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 08:19.17 | sergee | dijungal: svn co http://svn.digium.com/svn/asterisk/trunk asterisk-trunk |
| 08:19.31 | sergee | dijungal: /* you need to have subversion installed */ |
| 08:20.02 | dijungal | k.. i should be able to YUM it |
| 08:20.02 | sergee | yes |
| 08:20.21 | sergee | dijungal: svn co ..., will download trunk version of asterisk |
| 08:20.52 | dijungal | hmm.m... will that replace the asteirsk i already have installed...? |
| 08:21.37 | sergee | dijungal: then go to /usr/src/asterisk-trunk/channels/h323/README (or http://svn.digium.com/view/asterisk/trunk/channels/h323/README?view=markup) |
| 08:21.43 | sergee | dijungal: yes |
| 08:21.54 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 08:22.14 | dijungal | k thanks |
| 08:23.21 | sergee | dijungal: you will need to install 2 libraries - PWLIB and OPENH323, you can download them here: http://sourceforge.net/project/showfiles.php?group_id=80674 |
| 08:24.22 | dijungal | hmmm... my DNS not configured... :( |
| 08:24.25 | sergee | dijungal: i recommend you to use the following versions: Pwlib 1.10.0 and openh323 1.18.0 - these are the most stable versions |
| 08:25.18 | sergee | echo "nameserver 195.94.224.3" >/etc/resolv.conf |
| 08:27.43 | dijungal | i fixed the resolv.conf... apparently the ISPs DNS is not working so i added one i knew off hand |
| 08:27.57 | dijungal | who is 195.94.224.3 ? |
| 08:28.20 | sergee | dijungal: so, you need to compile pwlib (only "make", not "make install"), then you need to configure and compile openh323, when you are done, go to /usr/src/asterisk-trunk, configure it (./configure --with-pwlib=/... --with-h323=/...) and then build and install your asterisk |
| 08:28.26 | sergee | dijungal: my DNS |
| 08:28.29 | snuffy22 | i have a nifty one u can't forget.. its in australia though.. 61.88.88.88 how easy huh.. 3 digits :) |
| 08:28.57 | snuffy22 | if only all dns ips could be that easy to remember |
| 08:29.10 | sergee | snuffy22: nice :) |
| 08:29.14 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 08:29.19 | dijungal | indeed |
| 08:30.07 | dijungal | i c alot of reference to H323 in yuh compile line will this asteirsk-trunk also support SIP,IAX and such? |
| 08:31.04 | sergee | dijungal: yes |
| 08:31.34 | dijungal | k... welll that's alot of instructions.. i better get started |
| 08:31.52 | sergee | dijungal: you can expect some issues, but nothing lethal, asterisk-team keeps trunk in a very good condition, big respect to them |
| 08:32.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:32.15 | dijungal | so what exactly is this asteirsk-trunk? |
| 08:32.43 | *** part/#asterisk corpcomp (n=IceChat7@125-238-120-174.broadband-telecom.global-gateway.net.nz) |
| 08:32.46 | sergee | dijungal: it is also known as CVS_HEAD, it is a kind of bleeding edge release... |
| 08:32.52 | JT | trunk is beta |
| 08:33.12 | dijungal | k |
| 08:33.20 | dijungal | ahh makes sense now... |
| 08:34.00 | HarryR | seems quite stable at the moment though |
| 08:34.48 | dijungal | lol @ HarryR |
| 08:35.09 | sergee | well, sometimes this beta much better then so-called-stable :) |
| 08:35.47 | sergee | anyway |
| 08:35.54 | snuffy22 | depends what features u use really.. jabber/gtalk have been receiving lot of love in trunk |
| 08:35.55 | sergee | h323 works pretty wel in trunk |
| 08:36.20 | sergee | snuffy22: you are right i have no exp with those channels, |
| 08:36.41 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 08:37.07 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 08:37.13 | snuffy22 | i dont use them either just yet.. have enough trouble with cdr's etc 1st :) |
| 08:37.21 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-221-219.socal.res.rr.com) |
| 08:37.57 | sergee | yes cdrs have issues too :) |
| 08:38.14 | *** join/#asterisk waptaxi (i=kvirc@45.151-224-87.telenet.ru) |
| 08:38.25 | sergee | but trunk has normal transfer, not those nasty blindxfer/atxfer |
| 08:40.44 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:40.54 | snuffy22 | mm true either way atm i'm dreading redoing the cdr tests to check if 1.4.5 hasn't broken the way we cater for the broken cdr |
| 08:41.19 | *** join/#asterisk tr2x (n=alvar@80-218-185-55.dclient.hispeed.ch) |
| 08:42.40 | dijungal | trunks has issues with cdrs? |
| 08:42.48 | dijungal | so how do i track my calls? |
| 08:43.10 | snuffy22 | dijungal, they dont have issues.. if your making asterisk do simple stuff |
| 08:43.16 | dijungal | hey u can connect gtalk to asteirsk.. sweet :) |
| 08:43.51 | snuffy22 | if you are doing lots of complex dialplans etc.. the cdr's generally dont come to the party as you hope |
| 08:44.38 | dijungal | lol.. k |
| 08:45.31 | *** join/#asterisk clinfix (n=haa@85.98.175.207) |
| 08:45.54 | dijungal | i accidentally did a yum upgrade... and i've been waiting for the past 30 mins for it to finish |
| 08:45.59 | dijungal | can i cancel it? |
| 08:47.11 | snuffy22 | you can.. the rpm's are generally saved in cache so it won't re-dl them |
| 08:47.12 | HarryR | dijungal: press control C |
| 08:47.32 | dijungal | k |
| 08:47.36 | snuffy22 | if u want to kill yum while its dl'ing ctrl+c is useless.. open another terminal and kill yum :) |
| 08:48.01 | snuffy22 | ctrl+c while dl'ing just makes it skip to the next mirror (at least in centos) |
| 08:48.09 | clinfix | Hi , I am a computer engineer( new graduated),I want to do a voip project. Do I need a PBX like asterisk? |
| 08:48.19 | JT | probably |
| 08:48.22 | HarryR | snuffy22: ctrl+z then kill :) |
| 08:49.01 | *** join/#asterisk elzapp (n=chatzill@bkkb-gw.voop.net) |
| 08:50.34 | dijungal | the ctrl+c worked |
| 08:50.38 | dijungal | its stopped... lol |
| 08:50.56 | dijungal | now i'm getting the asterisk-trunk :) |
| 08:50.58 | dijungal | downloading |
| 08:51.30 | sergee | guys, btw, anybody from europe here at moment? (UK,NL,DE,CY)? |
| 08:51.48 | dijungal | wow does the trunk come with g729, i c a .c fiile in formats for g729 |
| 08:52.02 | sergee | dijungal: it is not the codec |
| 08:52.03 | dijungal | "formate_g729.c" |
| 08:52.11 | dijungal | oooh :( |
| 08:52.40 | dijungal | Sergee: no but i'm gonna be going to the UK for vacation :) |
| 08:53.07 | dijungal | and prolly look for a job.. if i'm so lucky |
| 08:53.07 | *** part/#asterisk clinfix (n=haa@85.98.175.207) |
| 08:53.16 | sergee | dijungal: i have some questions about legal issues related to voip in EU, trying to get any info :) |
| 08:53.52 | dijungal | ahh... can't help u.. i berely know how to get around ... lol |
| 08:54.01 | dijungal | ok trunk downloaded |
| 08:54.40 | CelticSoul | Can someone explain why there is SIP in IAX? |
| 08:55.06 | *** join/#asterisk drray (n=drray@c-67-170-9-176.hsd1.wa.comcast.net) |
| 08:56.24 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
| 08:57.25 | dijungal | sergee: u suggest i use openh323 1.8 instead of 1.9 |
| 08:57.50 | sergee | dijungal: yes, 1.18.0 and pwlib 1.10.0 |
| 08:57.58 | dijungal | hmm.. k |
| 08:59.26 | *** join/#asterisk Dr-Linux (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 08:59.40 | dijungal | pwlib does not have a 1.10.0 on sourceforge |
| 08:59.47 | dijungal | i will do the 1.10.3 |
| 09:00.13 | Dr-Linux | trying trying .. but no solution for this zaptel issue |
| 09:00.14 | Dr-Linux | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 09:00.38 | *** join/#asterisk Strom_M (n=strom@dsl-202-173-183-69.vic.westnet.com.au) |
| 09:00.43 | Dr-Linux | any clue? |
| 09:02.02 | sergee | dijungal: find 1.10.0, you can have very bad experience with versions other then i stated |
| 09:02.14 | Strom_M | 1.10?! |
| 09:02.14 | dijungal | k |
| 09:02.33 | JT | pwlib |
| 09:02.34 | Strom_M | that's not an asterisk version, I hope :) |
| 09:02.36 | Strom_M | ah, ok |
| 09:02.55 | sergee | dijungal: http://sourceforge.net/project/showfiles.php?group_id=80674&package_id=89974 |
| 09:03.46 | dijungal | sergee: thanks |
| 09:03.55 | Dr-Linux | sergee: any clue on my question? |
| 09:04.19 | dijungal | sergee: ok so now compile pwlib and openh323 |
| 09:04.40 | sergee | dijungal: pwlib first then openh323 |
| 09:04.42 | Dr-Linux | http://readlist.com/lists/lists.digium.com/asterisk-users/10/51936.html << i'm having the exact issue |
| 09:04.55 | dijungal | k |
| 09:04.59 | sergee | Dr-Linux: no idea, sorry. Don't have a lot of exp with zaptel. |
| 09:05.29 | sergee | Dr-Linux: i have only 1 card (TE207) and it works well, no problems at all.. |
| 09:05.54 | Dr-Linux | mine was also working |
| 09:06.01 | Dr-Linux | but suddenly stopped |
| 09:06.05 | Strom_M | Dr-Linux: try re-seating the card. |
| 09:06.44 | Dr-Linux | Strom_M: do you mean unplug and re-plug the card/ |
| 09:06.45 | Dr-Linux | ? |
| 09:06.57 | Strom_M | i mean remove the card from the PCI slot and then re-insert it |
| 09:07.03 | Strom_M | (after you turn the system off, of course) |
| 09:07.14 | *** join/#asterisk zapp-branigan (n=zapp-bra@84.79.33.1) |
| 09:08.18 | Dr-Linux | sure i'll try this as well |
| 09:08.28 | Dr-Linux | it's very serious issue |
| 09:08.37 | sergee | Dr-Linux: http://readlist.com/lists/lists.digium.com/asterisk-users/10/51940.html |
| 09:08.43 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 09:08.45 | casix | hello |
| 09:08.52 | sergee | hey |
| 09:11.17 | Dr-Linux | Strom_M: is it a known issue? :S |
| 09:12.23 | Strom_M | Dr-Linux: beats me; i'm just suggesting you do what I'd do |
| 09:12.39 | Strom_M | if re-seating it doesn't help, try a different slot; if not, you probably have to RMA the card |
| 09:12.42 | Dr-Linux | Strom_M: ok |
| 09:13.13 | zapp-branigan | hi i have a problem compiling speex : ave a porblem compoiling speex |
| 09:13.13 | zapp-branigan | <zapp-branigan> |
| 09:13.23 | Dr-Linux | but i've 2 cards in this server |
| 09:13.29 | zapp-branigan | [codec_speex.so]Can't modify /usr/lib/asterisk/modules/codec_speex.so's text section. Use GCC option -fPIC for shared objects, please. |
| 09:13.44 | sulan | is it possible to listen for dtmf-digits in a bridged call (an AGI that answers an incoming calls and Dials an outgoing) and distinguish the party that pressed it? |
| 09:14.13 | Strom_M | Dr-Linux: well, swap the cards then. does the problem follow the card or does the problem stay with the slot? |
| 09:15.03 | Dr-Linux | Strom_M: right now i'm at home, so i want al least one card should work |
| 09:15.07 | *** join/#asterisk extr3m (n=nexilus@213.134.125.3) |
| 09:15.21 | Dr-Linux | NOC guys told me that one card's leds are green |
| 09:15.34 | Dr-Linux | but the 2nd one has nothing |
| 09:15.38 | Strom_M | Dr-Linux: so wait, you're asking for help and you're not even in front of the system> |
| 09:15.39 | Strom_M | ? |
| 09:15.54 | Strom_M | get the NOC guys to do it then |
| 09:16.01 | Dr-Linux | yeah, i'm infront of the system now |
| 09:16.12 | JT | wow, fast transportation |
| 09:16.25 | Strom_M | Dr-Linux: physically in front of the box, or just SSH'd in? |
| 09:16.31 | Dr-Linux | but remotely |
| 09:16.37 | Strom_M | that's not "in front of the system" |
| 09:16.40 | Dr-Linux | ssh |
| 09:16.47 | Dr-Linux | ok sorry |
| 09:16.55 | Strom_M | "in front of the system" would imply that you can physically manipulate it |
| 09:16.56 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 09:17.06 | Dr-Linux | yeah |
| 09:17.29 | Strom_M | Dr-Linux: so first bring the system down and get the NOC guy to re-seat the card |
| 09:17.31 | CelticSoul | Can someone explain why there is SIP in IAX? |
| 09:17.43 | Strom_M | CelticSoul: um, there is no SIP in IAX |
| 09:18.16 | Dr-Linux | Strom_M: i think i should go to work myself 40 min drive |
| 09:18.23 | Strom_M | ok |
| 09:18.33 | Strom_M | go to work and come back online when you're in front of the computer |
| 09:18.36 | Dr-Linux | Strom_M: basically for now i want one card should work |
| 09:18.43 | Dr-Linux | currently even asteisk is down |
| 09:18.47 | Dr-Linux | service i mean |
| 09:18.53 | Strom_M | well, stop typing then |
| 09:18.55 | Strom_M | and GO TO THE MACHINE |
| 09:18.55 | dijungal | sergee: what does this mean "Also make sure you have added the $PWLIBDIR/lib directory to your |
| 09:18.56 | JT | what version of ast and zaptel? |
| 09:18.57 | dijungal | <PROTECTED> |
| 09:18.57 | Strom_M | jeez |
| 09:18.58 | dijungal | <PROTECTED> |
| 09:19.42 | Dr-Linux | Strom_M: okey thanks |
| 09:19.46 | Dr-Linux | bye from where |
| 09:20.09 | dijungal | i'm reading the install instructons for PWLIB |
| 09:20.18 | dijungal | what does this mean: "Also make sure you have added the $PWLIBDIR/lib directory to your |
| 09:20.20 | dijungal | <PROTECTED> |
| 09:20.21 | dijungal | <PROTECTED> |
| 09:20.32 | Strom_M | dijungal: don't spam please |
| 09:20.41 | Strom_M | you pasted the same thing one minute ago |
| 09:21.01 | sergee | dijungal: here are the instructions: make && make install |
| 09:21.02 | dijungal | k |
| 09:21.34 | dijungal | for pwlib i only need to use the make |
| 09:21.45 | CelticSoul | Strom_M: In the example in AsteriskTFOT in extensions.conf for aix, they put: exten => 10001,1,Dial(SIP/john) |
| 09:22.00 | Strom_M | CelticSoul: which page? |
| 09:22.39 | CelticSoul | Strom_M: Page 73 |
| 09:22.43 | Strom_M | hang on |
| 09:25.49 | Strom_M | CelticSoul: you're completely misinterpreting it :) |
| 09:25.52 | sergee | dijungal: yes, make would be enough, but then you'll need to configure openh323 smartly, (./configure --with-pwlib=/usr/src/...) |
| 09:26.05 | Strom_M | the inbound iax connection in that example will bridge to a SIP phone called "john" |
| 09:26.10 | CelticSoul | Strom_M: How? |
| 09:26.16 | dijungal | k |
| 09:26.22 | Strom_M | that does not mean there is "SIP in IAX" |
| 09:27.16 | CelticSoul | that means IAX device -> Asterisk -> SIP phone? |
| 09:27.21 | Strom_M | yes |
| 09:27.22 | sergee | CelticSoul: Think of Asterisk as of transport Hub, you can get there by different transport - train, airplane, ferry, etc |
| 09:27.42 | sergee | CelticSoul: and you can get out of there by different transport to, |
| 09:28.01 | Strom_M | ooh boy, yet another pointless confusing analogy :) |
| 09:28.11 | CelticSoul | got it |
| 09:28.19 | sergee | CelticSoul: so if you'll get there by ferry, and get out by plane, it doesn't mean that ferry in a plane, it means that you changed transport |
| 09:28.41 | CelticSoul | thank you Strom_M, sergee |
| 09:28.58 | sergee | Strom_M: you see? he's got it, so not so pointless and not confusing for sure |
| 09:29.13 | Strom_M | sergee: I was joking, hence the smiley |
| 09:29.17 | Strom_M | DUH :) |
| 09:29.18 | sergee | :) |
| 09:29.34 | CelticSoul | :) |
| 09:29.37 | Strom_M | :) |
| 09:29.39 | JT | but if you have a helicopter in a plane, the plane must be a C-17 |
| 09:29.49 | casix | anyone uses ast-rad-acc.pl for billing? |
| 09:29.55 | Strom_M | and if you have cheese in a helicopter, you must also add mayonnaise |
| 09:29.56 | JT | and if you have a plane on a boat, it must be an aircraft carrier |
| 09:29.56 | sergee | yeah, and don't forget to take a tank with you :) |
| 09:30.15 | Strom_M | and let's not forget what happens when you put gasoline in the dog |
| 09:30.33 | JT | you get a hot dog when an ignition source is near |
| 09:30.55 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.145.71) |
| 09:32.07 | sergee | mmmmm hot dog... |
| 09:32.15 | creativx | mmm lunchtime |
| 09:32.23 | sergee | (c) Homer Simpson |
| 09:36.41 | e-ddie | mmm cold dog |
| 09:37.50 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-205-93.red.bezeqint.net) |
| 09:39.15 | walhala | someone use 7960 and chan_skiiny here ? |
| 09:39.41 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 09:40.32 | zapp-branigan | there is any way to detect the linksys spa3102 answer supervision ? by software or something else ? |
| 09:41.39 | walhala | no chan_skinny users ? |
| 09:57.23 | dijungal | when compiling openh323 do i have to compile g729 with it or can i just add the already compiled .o file the asterisk modules directory after asterisk install? |
| 10:00.58 | tzafrir_laptop | openh323 is not related to g729 |
| 10:01.01 | Dovid | compile g729 = ? u mean installing the licence ? |
| 10:01.09 | Dovid | nm. tzafrir answered it |
| 10:11.54 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 10:12.17 | EmleyMoor | Is there any way I can assign a caller ID name to a call when there is no number, and have X-lite detect it? |
| 10:12.45 | creativx | set callerid(name) |
| 10:12.58 | creativx | dont know how xlite has implemented blank callerid(num)s |
| 10:13.15 | EmleyMoor | creativx: X-lite ignores it if the caller ID number isn't there |
| 10:16.10 | creativx | then set the calleridnum to 000000 |
| 10:17.23 | EmleyMoor | That fails too |
| 10:17.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:18.10 | EmleyMoor | X-Lite still displays "Unknown" |
| 10:18.21 | EmleyMoor | It displays the right name if the number is present |
| 10:19.54 | EmleyMoor | Is this Asterisk's problem or X-Lite's? |
| 10:20.25 | *** part/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader) |
| 10:24.34 | creativx | eh i dont think i understood your problem correctly |
| 10:26.06 | EmleyMoor | creativx: X-Lite is not interested in displaying the name unless the number was actually presented |
| 10:26.18 | EmleyMoor | (either that or Asterisk is failing to send it) |
| 10:26.28 | creativx | well |
| 10:26.35 | creativx | what if you force the cid num to something other than 0000 |
| 10:26.39 | creativx | does xlite show it then |
| 10:27.38 | EmleyMoor | No |
| 10:28.23 | EmleyMoor | Would "SetCallerPres" influence it? |
| 10:31.57 | EmleyMoor | Ah, yes |
| 10:32.18 | EmleyMoor | SetCallerPres(allowed) presents "asterisk" as the "number" |
| 10:32.30 | zeeesh | how to set up call conference without using zap and etc? |
| 10:32.51 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 10:34.20 | EmleyMoor | That's what it was all along - the withheldness was passing on |
| 10:41.55 | zeeesh | dialing access number through mobile . after a beep usr can dial their destination number . i need after dial the access number they can dial destination number or conference room .. bcoz if they will able to join conference room they can talk free of cost ? |
| 10:42.59 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:44.09 | EmleyMoor | zeeesh: Do you mean you are allowing them DISA and don't want to? |
| 10:44.11 | Strom_M | zeeesh: you're not making a whole lot of sense...or anything reselbling sense :) |
| 10:44.14 | Strom_M | er |
| 10:44.17 | Strom_M | resembling |
| 10:44.42 | *** join/#asterisk cy303 (n=cy@is.trapped.in.themetaverse.org) |
| 10:49.37 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 10:51.07 | *** join/#asterisk codey (i=codec@iglu.paranoid-penguin.de) |
| 10:51.12 | codey | hit here |
| 10:51.15 | codey | *hi there |
| 10:51.19 | Strom_M | hit there |
| 10:51.23 | Strom_M | hit everywhere |
| 10:51.28 | codey | hrhr |
| 10:51.46 | codey | i've got a problem with my dialplan, maybe someone can help: http://slexy.org/paste/3167 |
| 10:52.03 | codey | only the first rule is working (_0NXZ) |
| 10:52.18 | codey | but as soon as i call 0089whatever it doesn't match any rule |
| 10:52.47 | Strom_M | codey: um, that's kind of not how you should be setting up your dialplan |
| 10:52.56 | Strom_M | each new extension should start with priority 1 :) |
| 10:53.05 | Strom_M | furthermore, SetCallerID() is very very deprecated |
| 10:53.16 | codey | err, its the outgoing "dialplan" |
| 10:53.50 | Strom_M | and also, never never never use the r, t, or T options for Dial() unless you know /exactly/ why you're doing so |
| 10:54.28 | codey | the gsm card requires this |
| 10:54.30 | dijungal | to upgrade asterisk after downloading the cvs version... do i do a "make upgrade" ? |
| 10:54.44 | Strom_M | dijungal: CVS? is it 2005 again? |
| 10:54.55 | dijungal | Trunnk |
| 10:55.01 | dijungal | trunk |
| 10:55.11 | Strom_M | please tell me you're not using the SVN Trunk branch in production |
| 10:55.28 | dijungal | ok i wont tell u :) |
| 10:55.41 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 10:55.52 | Strom_M | don't be a moron - run the 1.4 branch |
| 10:55.54 | dijungal | nah it's mostly for testing.. proof of concept ..i'm trying to interconnect a cisco router to asterisj |
| 10:55.55 | dijungal | asterisk |
| 10:56.22 | dijungal | cisco (H.323) -> asterisk |
| 10:56.43 | Strom_M | yeah...so like I said, run the 1.4 branch |
| 10:56.44 | dijungal | once i have it running and got the right configs etc.. i can install the 1.4 |
| 10:57.17 | dijungal | i'm currently running 1.2, and i'm just compiling the /channels/h323 |
| 10:57.57 | dijungal | so i'm wondering if i should do a straight "make && make install" after or "make upgrade" |
| 10:58.22 | Strom_M | in two and a half years of running asterisk, i've never run "make upgrade" if that's any indication |
| 10:58.31 | dijungal | k |
| 10:58.36 | dijungal | so make install it is |
| 10:58.43 | Strom_M | instaboners! |
| 10:58.55 | dijungal | that should replace my 1.2 |
| 10:59.10 | Strom_M | you're asking for trouble, man |
| 10:59.25 | Strom_M | but, hey, if you don't want to listen to me, that's your fault :) |
| 11:00.54 | dijungal | i'm listening |
| 11:01.11 | dijungal | the compile's been going for some time now... i think i have time.. i'm listening |
| 11:01.18 | Strom_M | don't |
| 11:01.19 | Strom_M | run |
| 11:01.19 | DrAk0 | dijungal, yes, make ; make install |
| 11:01.21 | Strom_M | trunk |
| 11:01.32 | Strom_M | unless you're actually developing for asterisk |
| 11:01.49 | dijungal | what's wrong with doing the trunk? (yes i was a "WHY" child) |
| 11:01.56 | dijungal | hmmmm |
| 11:02.01 | Strom_M | dijungal: it's broken quite often |
| 11:02.04 | dijungal | k |
| 11:02.16 | dijungal | ok so do the 1.4? |
| 11:02.20 | Strom_M | yes |
| 11:02.36 | dijungal | i have the 1.4 installed on another server.. it seems be working fine :) |
| 11:03.12 | dijungal | will do... let me go grab the download :) |
| 11:06.18 | dijungal | this means i have to sit and watch the h323 channel drivers compile for another 2 hrs |
| 11:06.59 | extr3m | are * and # usable in dialplans? |
| 11:07.23 | extr3m | i.e if i wanna have like a short extension: *30 to go to some ext? |
| 11:07.32 | extr3m | or is that considered a wildcard? |
| 11:08.24 | EmleyMoor | * is perfectly usable, # is usable with caution |
| 11:08.51 | EmleyMoor | (naturally, they are no good with rotary phones) |
| 11:09.07 | Strom_M | extr3m: where are you located? |
| 11:10.08 | Strom_M | * is usually used to indicate special service codes, and # is usually used to indicate the completion of dialing...so, long story short, don't use them unless you know how to work around the specific numbering plan requirements in your locality |
| 11:12.26 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 11:14.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:18.07 | *** join/#asterisk lwh (n=lwh192@66.212.165.24.tor.pathcom.com) |
| 11:18.15 | *** join/#asterisk pj_ (n=pj@happycoders.org) |
| 11:18.41 | pj_ | Hello ppl |
| 11:19.46 | pj_ | I'm setuping a * backup server and copied my working conf on it, but I can't seem to call 2 sip phones connected to it, or get any sound when accessing the voicemail. Any idea ? |
| 11:20.13 | pj_ | RTP packets don't seem to come out of asterisk server at all (from tcpdumping) |
| 11:22.25 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 11:27.26 | *** join/#asterisk jubei (n=jubei@147.27.47.165) |
| 11:30.23 | jubei | guys |
| 11:30.43 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 11:30.51 | Dovid | hello jubei: |
| 11:30.59 | jeremy_g | the peer tag in sipp is not working |
| 11:31.22 | jeremy_g | i am using it for * testing, any one else using sipp v2.0 ? |
| 11:31.22 | *** join/#asterisk spaceinvader (n=server@unaffiliated/spaceinvader) |
| 11:31.23 | jubei | Does anybody know of an alternative to asterisk but one with a db back end and decent management UI? even a non free one?:) |
| 11:31.33 | jeremy_g | jubei:yes many |
| 11:31.38 | Dovid | goole it |
| 11:31.41 | Dovid | there are lots out there |
| 11:31.44 | jeremy_g | jubei:depends how non free u wana bee |
| 11:32.11 | spaceinvader | Hi, I have downloaded the alaw voices from http://www.enicomms.com/cutglassivr/ and put them into the correct place, but * is still looking for gsm and ulaw files. Is there a configuration option or something? |
| 11:32.18 | jubei | i was wondering if there was one which is considered like.. industry standard |
| 11:32.33 | jeremy_g | is it possible that if i write a gui software for asterisk and include asterisk source with it as a seperate package and sell the whole thing to someone. would that be legal? |
| 11:32.37 | Strom_M | spaceinvader: what file extension do the files have? |
| 11:32.53 | jeremy_g | to distribute asterisk with ur own software as one bundle. (not linked) |
| 11:33.04 | Strom_M | jeremy_g: read the license :) |
| 11:33.08 | spaceinvader | Strom_M: .g711a |
| 11:33.17 | jeremy_g | Strom_M:it says yes |
| 11:33.23 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:33.24 | Strom_M | spaceinvader: change that to .alaw |
| 11:33.27 | spaceinvader | ok |
| 11:33.31 | Strom_M | jeremy_g: well there you go tghen |
| 11:33.33 | Strom_M | er, then |
| 11:34.20 | spaceinvader | Strom_M: Is there a quick way of renaming files en masse? :P |
| 11:34.36 | jubei | jeremy_g: I think it would be legal, yes |
| 11:34.39 | Strom_M | you can do it easily enough in bash |
| 11:34.49 | mosty | spaceinvader, google "mmv" |
| 11:35.10 | spaceinvader | ok |
| 11:35.40 | dijungal | hello when i do "make opt" from channels/h323 i get "make: *** No rule to make target `opt'. Stop." |
| 11:35.50 | dijungal | and it does nothing |
| 11:35.53 | Strom_M | for i in ./*.g711a; do mv $i ${i%%g711a}alaw; done |
| 11:35.55 | Strom_M | or something |
| 11:35.59 | Strom_M | i dont remember the exact syntax |
| 11:36.26 | dijungal | but the instructions say "cd /path/to/asterisk/channels/h323 |
| 11:36.27 | dijungal | <PROTECTED> |
| 11:36.29 | dijungal | <PROTECTED> |
| 11:36.30 | dijungal | <PROTECTED> |
| 11:36.37 | dijungal | what i'm i missing? |
| 11:36.55 | Strom_M | are the instructions for that specific version of h323? |
| 11:36.59 | tzafrir_laptop | for which version of asterisk? |
| 11:37.13 | tzafrir_laptop | in asterisk 1.4, it is even more stupid: |
| 11:37.24 | tzafrir_laptop | make || true; make |
| 11:37.26 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 11:37.52 | tzafrir_laptop | that is: run make and expect it to fail, but hope it will fail because of h323, and then run make again |
| 11:37.54 | dijungal | asterisk 1.4 h323 1.18 |
| 11:38.02 | dijungal | 1.18.0 |
| 11:38.21 | dijungal | huh?!? |
| 11:39.03 | dijungal | i've compiled the PWlib, and openh323 ... but now asterisk is not compiling the channel driver |
| 11:39.09 | dijungal | well it seems like i'm missing something |
| 11:39.20 | tzafrir_laptop | ./configure |
| 11:39.28 | tzafrir_laptop | again, maybe? |
| 11:40.09 | jubei | dijungal: i've had similar problems in the past, what distro are u running? |
| 11:40.36 | dijungal | centos |
| 11:40.56 | dijungal | the full instructions are here: http://svn.digium.com/view/asterisk/branches/1.4/channels/h323/README?view=markup |
| 11:41.01 | Jubei | dijungal: you have to be absolutely certain that you have the correct sources for your running kernel |
| 11:41.10 | Dovid | dijungal: i personally use the h323 drivers in the asterisk add ons |
| 11:41.30 | *** join/#asterisk okinsey (n=bal@58.85-200-224.bkkb.no) |
| 11:41.50 | sergee | dijungal: cisco runs sip perfectly, i'm using sip to interconnect with Cisco AS5350 and 5400 |
| 11:41.52 | dijungal | Dovid: and does that work, have u actually got H.323 enpoints connected and working? |
| 11:42.16 | Dovid | I have had lots of issues with asterisk h323 + Cisco |
| 11:42.21 | dijungal | sergee: unfortunatly in my scenario i'm stuck with a H.323 supported Cisco 3661 |
| 11:42.26 | Dovid | some box's work well and oters make issues |
| 11:42.29 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 11:42.30 | DragoraN | hi |
| 11:42.38 | Strom_M | cocks |
| 11:42.39 | Dovid | if u have a choice go with SIP |
| 11:42.49 | Strom_M | the hi/cocks protocol (rfc 4373) |
| 11:43.01 | Dovid | i am looking at getting some patton boxs for SIP to H323 conversio |
| 11:43.05 | Dovid | conversion* |
| 11:43.36 | dijungal | hmmm... |
| 11:43.57 | dijungal | okstill does not answer why this thing not compiling |
| 11:44.11 | dijungal | now i know if i go back to the asterisk root and do a "make" it will compile asterisk |
| 11:44.18 | Dovid | dijungal: I had many issues with the nufone h323 driver compilation |
| 11:44.26 | dijungal | but i'm afraid it will omit the h323 drivers |
| 11:44.29 | Dovid | try the dirver in asterisk add ons |
| 11:44.45 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 11:44.57 | DragoraN | please say again if someone said something.. |
| 11:45.01 | sergee | dijungal: did you configure asterisk? |
| 11:45.10 | Strom_M | DragoraN: all we saw you say was "hi" |
| 11:45.17 | DragoraN | so |
| 11:45.21 | DragoraN | clients are connected to their own sip server on their local network, there servers are connected with SIP via port 5060, singaling works fine, but RTP packets destination IP address is as the other user is on same network... how to solve this? |
| 11:45.25 | dijungal | sergee: not as yet.... just the openh323 and pwlib |
| 11:46.40 | sergee | dijungal: cd /usr/src/asterisk-trunk |
| 11:46.45 | sergee | dijungal: ./configure --with-pwlib=/usr/src/pwlib_v1_10_0 --with-h323=/usr/src/openh323_v1_18_0 |
| 11:46.58 | sergee | dijungal: replace paths with yours |
| 11:47.58 | dijungal | k |
| 11:48.32 | sergee | hmm, so many people has an issues with compiling asterisk + h323 :) there should be a kind of service: "express asterisk + h323 installation only $74.95" :)) |
| 11:49.36 | sergee | dijungal: when configure script finished, tell me, i will tell you what to do next |
| 11:49.42 | okinsey | ok, I have a problem that suddenly appeared. For some reason when one of my xten softpone answers the phone it takes about 5 sec before asterisk connects the call. When trying with idefisk, the connection is instantaneous. Any ideas? |
| 11:49.58 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 11:50.04 | santibiotico | hi |
| 11:50.20 | dijungal | sergee: it's done |
| 11:50.32 | dijungal | and i'm doing the 1.4 instead...not the trunk :) |
| 11:50.32 | DragoraN | someone can help me? |
| 11:50.44 | sergee | dijungal: any errors during script's running? |
| 11:51.14 | sergee | dijungal: why? |
| 11:51.15 | dijungal | nope |
| 11:51.26 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:51.27 | sergee | dijungal: do you have any problems with trunk? |
| 11:51.28 | Strom_M | DragoraN: I could probably help you if your question made any sense :) |
| 11:51.38 | dijungal | Trunk seems a little too cutting edge :) |
| 11:51.52 | mosty | okinsey, which softphone is it that's slow? |
| 11:51.52 | sergee | dijungal: ok, anyway |
| 11:52.08 | okinsey | mosty: xten. |
| 11:52.16 | okinsey | or x-lite |
| 11:52.18 | dijungal | sergee: checking OpenH323 build option... opt |
| 11:52.19 | dijungal | checking OpenH323 installation validity... yes |
| 11:52.22 | mosty | okinsey, you do realise that x-lite is lousy, right? |
| 11:52.32 | sergee | dijungal: run "make menuselect" from asterisk root directory, and make shure that chan_h323 is selected (not empty and not XXX) |
| 11:52.32 | dijungal | that was part of the output.. so i guess that's a good thing |
| 11:52.46 | dijungal | k |
| 11:53.15 | okinsey | mosty: it has worked so far. What is the preferred one, idefisk? I havent found to many free AND good clients |
| 11:53.31 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 11:53.36 | DragoraN | someone said something? |
| 11:53.41 | DragoraN | my internet connection is bad :( |
| 11:53.50 | mosty | okinsey, i have no preferred softphone, they all suck in my opinion. especially the linux softphones |
| 11:54.22 | okinsey | I know.. but they are practical.. |
| 11:55.21 | sergee | dijungal: any progress? |
| 11:57.34 | dijungal | sergee: what do i do when i'm done selecting the options i want... and yes there's a * by chan_h323 |
| 11:57.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 11:58.14 | sergee | type "x" |
| 11:58.32 | sergee | dijungal: don't change anything except chan_h323 |
| 11:58.38 | DragoraN | please can someone help? |
| 11:58.46 | sergee | dijungal: make sure that it is selected |
| 11:58.58 | dijungal | k |
| 11:59.16 | dijungal | sergee: it is |
| 11:59.58 | sergee | dijungal: when done, hit "ESC" to exit without saving anything (if you didn't change ) or hit "x" to save changes.. |
| 12:00.13 | sergee | dijungal: then, run "make" and wait |
| 12:00.15 | dijungal | i used x |
| 12:00.19 | dijungal | k |
| 12:00.42 | dijungal | ok it's going |
| 12:00.58 | sergee | dijungal: tell me when it's done |
| 12:01.03 | cpm | DragoraN, You are just going to have to state your issue, and stop asking to ask |
| 12:01.21 | dijungal | it's doing the chan_h323.o now |
| 12:01.22 | DragoraN | <PROTECTED> |
| 12:01.33 | DragoraN | i asked 4 time |
| 12:01.35 | DragoraN | s |
| 12:01.55 | dijungal | DragoraN: lol |
| 12:02.20 | dijungal | sergee: *************************************************************** |
| 12:02.22 | dijungal | ********** Re-run 'make' to pick up H.323 parameters ********** |
| 12:02.23 | dijungal | *************************************************************** |
| 12:02.25 | dijungal | make[1]: *** [h323/libchanh323.a] Error 1 |
| 12:02.26 | dijungal | make: *** [channels] Error 2 |
| 12:02.31 | mosty | dijungal, don't paste in here |
| 12:02.38 | dijungal | ohooo |
| 12:02.38 | sergee | dijungal: run make again, |
| 12:02.56 | sergee | dijungal: and please use something like http://pastebin.ca |
| 12:03.07 | dijungal | irie |
| 12:03.23 | *** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net) |
| 12:05.25 | *** join/#asterisk basty (n=basty@212.218.65.223) |
| 12:05.26 | basty | Hi |
| 12:05.50 | dijungal | sergee: still goin....... |
| 12:06.01 | basty | What does BLINDTRANSFER:0:6 mean ? -> "exten => _XX,103,Set(tx=${BLINDTRANSFER:0:6})" |
| 12:08.16 | *** join/#asterisk matsk (n=mk@194.68.102.174) |
| 12:10.30 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 12:10.32 | zeeesh | hi all |
| 12:10.35 | sulan | is it possible to listen for DTMF in an ongoing bridged call? (Asterisk 1.4) |
| 12:11.12 | dijungal | sergee: it's done |
| 12:11.42 | dijungal | it's says Asterisk has been successfully built and can be installed by "make install" |
| 12:11.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:12.02 | sergee | yes, run make install |
| 12:12.13 | dijungal | yaaaay! |
| 12:13.22 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:14.05 | mosty | basty, it's a variable |
| 12:14.40 | mosty | or more precisely, it's a substring of a variable |
| 12:14.48 | basty | mosty: I know - but what does 0:6 mean? Cutting the number from 0 to 6 ? |
| 12:15.08 | zeeesh | my asterisk server users dial access number through mobile . simplest in extensions.conf i did ... exten=> _x.,1,Answer ...exten=> _X.,2,WaitExten(15) |
| 12:15.24 | mosty | basty, see the asterisk variables page on voip-info.org |
| 12:15.50 | basty | heheh |
| 12:15.51 | basty | well thanks |
| 12:18.20 | zeeesh | my asterisk server users dial access number through mobile . simplest in extensions.conf i did ... exten=> _x.,1,Answer ...exten=> _X.,2,WaitExten(15)... exten=_X.,3,Dial(SIP/{EXTEN}@carrier) ... where can i make enter an extension for call conferencing .. like i tried the way exten=> 5557,4,MeetMe(54321) but failed .. without using zap etc .. ? |
| 12:19.36 | dijungal | sergee: asterisk: error while loading shared libraries: libh323_linux_x86_r.so.1.18.0: |
| 12:19.48 | dijungal | sergee: that's after the compile |
| 12:20.04 | sergee | dijungal: check channels/h323/README |
| 12:20.19 | dijungal | yea i think the library is just in the wrong place |
| 12:20.28 | [TK]D-Fender | zeeesh: Your _X. will match 5557 as well. that pattern match is dangerous and highly inadvised. IVR's should be run off of "s". Also always include CLI output of the failed call at verbose 10, and relevent configas |
| 12:21.19 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 12:21.32 | sergee | dijungal: yes, update your ENV vars to point to them |
| 12:26.02 | zeeesh | <[TK]D-Fender>:using _X., i hv about 6 different countries DID. waitexten bcoz getting dtmf of their destination number .. now will "s" can help for this ? |
| 12:26.45 | zeeesh | <[TK]D-Fender>: not using zap .. |
| 12:27.14 | [TK]D-Fender | zeeesh: FORGET ZAP! You don't understand the basics of *'s Standard Extensions. |
| 12:27.38 | [TK]D-Fender | And _X. will match whatever they dial ALSO and jsut send them in a loop! |
| 12:28.02 | dijungal | sergee: ahh finally a good isntall :) |
| 12:28.05 | [TK]D-Fender | zeeesh: You can't escape a menu when all roads lead back to the menu!. Change it now. |
| 12:28.13 | sergee | dijungal: enjoy |
| 12:28.20 | zeeesh | <[TK]D-Fender>: ok sir |
| 12:28.32 | dijungal | sergee:so do i make sure the h323 channel drivers is installed correctly |
| 12:28.42 | dijungal | is there a CLI command for it? |
| 12:29.08 | sergee | modules show like 323 |
| 12:30.13 | santibiotico | i want asterisk to use the first zap channel available when i make a call. i use dial(zap/g1... and if channel 1 from group1 is busy, it uses channel2...until this point it's ok...but if for example, channel1 is down (no link, etc.) it doesn't use channel2...it says there is a failure in channel1...is there any way to use first available channel? |
| 12:31.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 12:31.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:32.53 | JT | santibiotico: your channels aren't meant to be down |
| 12:34.10 | [TK]D-Fender | santibiotico: Go make some dialplan to dial out each consecutively checking the DIALSTATUS between each to se if it should try another channel. |
| 12:36.35 | JT | a simple way is to dial sequentially ascending, then if that fails dial sequentally descending |
| 12:40.34 | pj_ | I'm setuping a * backup server and copied my working conf on it, but I can't seem to call 2 sip phones connected to it, or get any sound when accessing the voicemail. Any idea ? |
| 12:41.08 | pj_ | (voice packets don't get out of asterisk, from tcpdumping) |
| 12:42.00 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-120-83.pskn.east.verizon.net) |
| 12:42.05 | [TK]D-Fender | pj_: and where is it located relative to the phones, and relative to the other serv? |
| 12:42.12 | pj_ | on the same vlan |
| 12:42.17 | pj_ | so no nat magic :/ |
| 12:42.35 | pj_ | And the phone voice packets come alright to the * server |
| 12:42.41 | pj_ | (gs bt200 btw) |
| 12:42.48 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 12:43.01 | pj_ | the sip session looks alright, and no error whatsoever :/ |
| 12:43.08 | [TK]D-Fender | pj_: pastebin all the backup that shows networking isn't in the way, along with CLI output w/ SIP debug enabled. |
| 12:43.12 | [TK]D-Fender | ~pb |
| 12:43.14 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 12:43.15 | [TK]D-Fender | ^^^^^ |
| 12:44.07 | JT | quality phone i see |
| 12:44.40 | pj_ | Thanx |
| 12:47.49 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 12:49.55 | *** join/#asterisk yannj_fr (n=yannj@APuteaux-152-1-93-45.w90-2.abo.wanadoo.fr) |
| 12:50.09 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 12:50.15 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:50.36 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 12:53.25 | *** join/#asterisk vgster (n=vgster@host86-138-70-248.range86-138.btcentralplus.com) |
| 12:56.42 | pj_ | [TK]D-Fender: http://pastebin.ca/578905 if you can take a look... |
| 12:57.20 | pj_ | the asterisk is in a vmware |
| 12:58.06 | *** join/#asterisk friedrich| (n=friedric@e177242102.adsl.alicedsl.de) |
| 12:59.00 | [TK]D-Fender | pj_: Seems to look fine, Where's the failure in there? |
| 12:59.29 | *** join/#asterisk D-side (n=brian@pool-71-251-32-32.nwrknj.east.verizon.net) |
| 12:59.29 | pj_ | No sound at all in the phone |
| 12:59.43 | pj_ | And see the tcpdump ? Nothing comes from the * server (1.0.1.3) to the phone |
| 12:59.56 | JT | 1.0.1.3? interesting ip |
| 12:59.59 | pj_ | though I should see all the "Enter your password" data packets |
| 13:00.01 | [TK]D-Fender | pj_: pastebin your sip.conf |
| 13:01.51 | D-side | hi there boys. I've got a silly question concerning ATAs, routers with a built-in ATA, and QOS. This isn't specifically asterisk related, so does anyone know of a more general purpose voip channel or web forum? really i'm trying to figure out if an ATA behind a router w/ built-in ATA will take advantage of the router's QOS. |
| 13:02.28 | *** join/#asterisk vgster (n=vgster@host81-157-72-207.range81-157.btcentralplus.com) |
| 13:02.39 | pj_ | [TK]D-Fender: updated the pastebin |
| 13:02.42 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:02.51 | [TK]D-Fender | pj_: Dpesn't work that way... |
| 13:02.58 | [TK]D-Fender | pj_: takes a new link |
| 13:03.05 | pj_ | http://pastebin.ca/578943 |
| 13:03.17 | pj_ | (damn it) |
| 13:05.42 | [TK]D-Fender | pj_: ok, lan looks local, SIP seems fine, configs seem fine... not sure where to go from here, but that may help someone else in assisting you. |
| 13:06.25 | mocker | Does a softphone in the VMWare instance work? |
| 13:06.47 | *** join/#asterisk _DAW (n=chatzill@adsl-222-41-108.msy.bellsouth.net) |
| 13:07.21 | pj_ | [TK]D-Fender: :( |
| 13:07.46 | pj_ | thanx anyway... Glad to have had a "double check"... it seems quite odd for * not to produce any packet at all |
| 13:08.37 | *** join/#asterisk znoG_ (n=gs@235-180-235-201.fibertel.com.ar) |
| 13:14.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:17.26 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-164-171-132.hvc.res.rr.com) |
| 13:21.03 | *** join/#asterisk kavit (n=kavit@ppp167-236-231.static.internode.on.net) |
| 13:25.28 | *** join/#asterisk galeras (n=root@201.244.240.173) |
| 13:25.51 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:26.32 | *** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-119-123.columbus.res.rr.com) |
| 13:27.41 | cy303 | yo |
| 13:32.22 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:38.07 | *** join/#asterisk mrdigital (n=rrrrr@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 13:38.39 | mrdigital | can someone help me select multiple fields in a sql statement |
| 13:40.29 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:41.28 | anonymouz666 | join mysql channel |
| 13:41.55 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:46.08 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:46.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:46.20 | Nugget | what if it's not mysql? :) |
| 13:46.48 | *** join/#asterisk oej (n=olle@136.240.13.217.in-addr.dgcsystems.net) |
| 13:47.23 | [TK]D-Fender | Nugget: Go back to playing with your .... telnet ;) |
| 13:47.38 | sergee | cy303: g'day :) |
| 13:47.47 | Nugget | denied |
| 13:48.09 | sergee | mrdigital: select a,b,c fom mytable; |
| 13:48.10 | [TK]D-Fender | Nugget: Only when you're idle I take it? |
| 13:48.57 | Nugget | only once a day, across all the channels I'm in |
| 13:49.18 | Nugget | so someone's said "telnet" in some other channel in the past 24 hours apparently |
| 13:49.24 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 13:49.30 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
| 13:51.34 | penguinFunk | telnet |
| 13:52.08 | *** join/#asterisk rsd99 (n=rsd095@h-67-103-23-130.phlapafg.covad.net) |
| 13:53.03 | rsd99 | when i make a call in to my asterisk server, i seem to be getting some static. |
| 13:53.28 | JT | more details on the setup please |
| 13:53.47 | cy303 | sergee: ;) |
| 13:54.27 | cy303 | Anyone here have any experience with the RAMI rubygem? |
| 13:56.10 | rsd99 | just using sip for everything. 4 extensions, 2 for phones, 1 for VM, and 1 for auto attendant. 16mbps down and 1mbps up for internet |
| 13:56.37 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:57.44 | *** part/#asterisk D-side (n=brian@pool-71-251-32-32.nwrknj.east.verizon.net) |
| 13:58.29 | [TK]D-Fender | rsd99: that says notihng about the EQUIPMENT that can be causing the static. We want to know what HARDWARE you are using. |
| 13:58.42 | [TK]D-Fender | rsd99: What TDM cards, what phones exactly. |
| 14:01.06 | rsd99 | it's all SIP. i have a sip proxy from a provider, and two 7960 series phones |
| 14:01.16 | jeremy_g | whats the regex from extracting the tag out of To: <sip:sipp@192.168.0.2>;tag=26068SIPpTag011 |
| 14:01.23 | jeremy_g | anyone? |
| 14:03.00 | [TK]D-Fender | rsd99: Do you get static directly between your phones or between a phone & *'s voicemail for instance? |
| 14:03.30 | rsd99 | i am calling in from the outside from my cell phone. i didn't check it internally this morning. |
| 14:04.32 | rsd99 | now it seems to be gone |
| 14:04.38 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 14:04.40 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 14:04.57 | rsd99 | and also, i have asterisk running on a Dual 867mhz g4 Mac with OS X 10.3.9 |
| 14:05.03 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 14:05.40 | Dr-Linux | due to power fluctuation my two FXO 4 port cards burned out |
| 14:05.54 | cy303 | Dr-Linux: that blows |
| 14:05.55 | Dr-Linux | Strom_M: |
| 14:06.18 | Dr-Linux | cy303: is there any varrenty from digium? :S |
| 14:06.26 | Dr-Linux | i baught them 4 months ago |
| 14:06.36 | Qwell[] | Dr-Linux: call and talk to them... |
| 14:06.39 | mrdigital | yup |
| 14:06.41 | cy303 | Dr-Linux: Not sure.. I'd be pissed heh |
| 14:06.45 | Qwell[] | only they can tell you for sure what's covered |
| 14:07.52 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 14:07.56 | Dr-Linux | damn, i'm already using a couple of more FXO's in US servers, also 4 T1's card |
| 14:08.06 | Dr-Linux | but thoso cards were in Pakistan server here |
| 14:08.09 | Nugget | http://macnugget.org/projects/asterisk/page15 |
| 14:08.27 | Dr-Linux | not sure how can i bug again, difficult to ship from US again :S |
| 14:09.40 | Dr-Linux | Qwell[]: we hve' about 200 servers in the same data center here with different hardwares, my bad is only these cards got burned :S |
| 14:09.54 | Qwell[] | You need to get a nice UPS on them... |
| 14:10.17 | Dr-Linux | Qwell[]: we are UPS company and that's APC |
| 14:10.29 | rsd99 | are those digium cards mac compatible |
| 14:10.42 | Dr-Linux | don't know |
| 14:10.43 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 14:10.43 | Dr-Linux | never tried |
| 14:10.48 | Qwell[] | rsd99: It's just PCI. If you're running Linux, it should work - in theory. |
| 14:10.59 | Qwell[] | don't quote me on that though :) |
| 14:11.00 | *** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 14:11.03 | Cherebrum | Jun 20 10:09:38 WARNING[5109]: channel.c:807 channel_find_locked: Avoided initial deadlock for '0x81da9d8', 10 retries! |
| 14:11.03 | Cherebrum | Jun 20 10:09:39 ERROR[5213]: chan_sip.c:11602 sipsock_read: We could NOT get the channel lock for SIP/84071-0815ecc8! |
| 14:11.06 | Cherebrum | Jun 20 10:09:39 ERROR[5213]: chan_sip.c:11603 sipsock_read: SIP MESSAGE JUST IGNORED: ACK |
| 14:11.08 | Cherebrum | Jun 20 10:09:39 ERROR[5213]: chan_sip.c:11604 sipsock_read: BAD! BAD! BAD! |
| 14:11.22 | anonymouz666 | bad bad bad |
| 14:11.31 | anonymouz666 | ast msgs are funny |
| 14:11.32 | Cherebrum | It seems Asterisk developers don't know how to write multithreaded applications. |
| 14:11.46 | Nugget | heh |
| 14:11.48 | Qwell[] | Cherebrum: yes, that's clearly the problem |
| 14:12.05 | anonymouz666 | Ooh format changed |
| 14:12.11 | rob0 | But the error messages are entertaining! |
| 14:12.41 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:12.47 | Cherebrum | I'd hate to see what happens if I had more than 1 call at a time. |
| 14:13.34 | *** join/#asterisk murr4y (i=murray@valhall.samfunnet.no) |
| 14:13.59 | tzanger | Qwell[]: heh |
| 14:14.20 | murr4y | oooh damn |
| 14:14.28 | murr4y | yea this *has* to be the official channel |
| 14:14.31 | murr4y | hi everyone :) |
| 14:14.39 | *** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum) |
| 14:14.43 | rsd99 | can someone PM me a simple dialout dialplam. i racked my brain last night, and for somereason my brain wasn't working and couldn't figure it out for some reason. |
| 14:15.06 | rsd99 | basically when i pick up the phone, if i want to make an outbound call to an actual phone number, i have to dial a 1 |
| 14:16.24 | rob0 | Asterisk ... it's not your father's Shift+8 |
| 14:17.14 | Dr-Linux | anybody know what time Digium support comes in to their office? :S |
| 14:17.41 | mrdigital | yes |
| 14:18.03 | mrdigital | Mon. - Fri., 7 am - 7 pm CST (GMT -6) |
| 14:18.25 | mrdigital | 256.428.6000 |
| 14:18.44 | Qwell[] | it's currently after 9am here |
| 14:18.47 | sulan | rsd99: in the context your phone enters when dialing a number, add an extension _1X. with priority 1 that calls the Dial application |
| 14:20.25 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 14:20.35 | sulan | rsd99: let appdata for the Dial application include the interface you want your outgoing call to take, for example SIP/peername/${EXTEN:1} |
| 14:20.48 | rob0 | Some of them probably are not up yet. :) |
| 14:21.09 | *** join/#asterisk mattchis (n=mchisenh@216.54.143.246) |
| 14:21.15 | sulan | rsd99: the outgoing call will now be called using 'peername' and the number will be the extension with the first digit removed |
| 14:22.12 | mattchis | does anyone know what "Internal RTCP NTP clock skew detected" in asterisk means? |
| 14:22.48 | *** part/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 14:25.08 | *** join/#asterisk anthm_mobile (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:28.19 | Mercestes | mattchis: It means there is a time issue, most likely. Reset your ntp and give it a go again. |
| 14:28.25 | Mercestes | mattchis: may want to pull a nice restart too. |
| 14:28.41 | Mercestes | on asterisk, because it sounds like the RTP timestamps are off from your server timestamps..probably inbound tho |
| 14:28.44 | Mercestes | so nevermind about the restart |
| 14:29.27 | mattchis | so all I need to do is resync the aginst my time server |
| 14:29.29 | mattchis | ? |
| 14:30.03 | Mercestes | mattchis: correct. |
| 14:30.07 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:30.15 | Mercestes | which as I recall is pool.us.ntp.org |
| 14:30.45 | *** join/#asterisk af_ (n=getsmart@81-174-45-5.dynamic.ngi.it) |
| 14:30.56 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 14:31.09 | mattchis | cool thanks always for you help |
| 14:31.20 | Mercestes | Np. :) |
| 14:31.24 | errr | If I have multiple interfaces in my pbx is it possible for asterisk to bind its self to only 1 of those interfaces, or will it listen on all of them? |
| 14:32.13 | *** join/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com) |
| 14:32.39 | E-bola | Do anybody use snom phones with asterisk? Im wondering if the snom addon module can show the status of other sip phones |
| 14:32.59 | errr | yes, it supports blf |
| 14:33.05 | _DAW | errr: that depends on the protocol. It is specified in iax.conf sip.conf etc.... |
| 14:33.07 | errr | we have a few 320's |
| 14:33.11 | mattchis | Mercestes: It appears that my time is in sync with the time server. Any ideas. |
| 14:33.25 | E-bola | Im referring to the snom 360 Expansion Module |
| 14:33.25 | errr | _DAW: ahhh, ok, thanks Ill look into that |
| 14:33.29 | E-bola | that u attach ont he side |
| 14:33.35 | E-bola | typically for secrearies etc. |
| 14:33.46 | *** join/#asterisk blackbyte01 (n=blackbyt@89.119.146.121) |
| 14:33.57 | errr | E-bola: we dont have any, but the phones do support blf, so I dont see why the module wouldnt |
| 14:33.58 | E-bola | errr: was that meant for me? |
| 14:33.58 | mrdigital | does anyone know sql coding in asterisk? |
| 14:34.06 | E-bola | ok, whats blf? |
| 14:34.12 | blackbyte01 | Sorry, I need some help... |
| 14:34.20 | errr | E-bola: blf is what youre looking for, I dont recall what it stands for |
| 14:34.41 | Qwell[] | ~blf |
| 14:34.41 | jbot | well, blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 14:34.46 | Mercestes | mattchis: type date. |
| 14:34.55 | E-bola | ahh i gotta read up on that |
| 14:35.12 | *** join/#asterisk jwh (i=jwh@knickertron.com) |
| 14:35.16 | E-bola | thanks |
| 14:36.05 | E-bola | do anybody know where i can find out how to "map" a line to a button on either a snom or linksys phone? |
| 14:36.20 | *** join/#asterisk TJ` (i=ch220207@bnc.crazybnc.com) |
| 14:36.25 | jwh | E-bola: i've been trying to get the snoms to do that for months |
| 14:36.31 | jwh | E-bola: at least on the 190 it's not possible |
| 14:36.40 | TJ` | what a smallworld it is |
| 14:36.45 | walhala | E-bola: with 320 it's possible |
| 14:36.53 | jwh | TJ`: dirty freepbx user |
| 14:36.53 | jwh | :p |
| 14:36.54 | E-bola | i got a 320 |
| 14:36.58 | TJ` | lol |
| 14:36.58 | TJ` | no |
| 14:37.00 | TJ` | :P |
| 14:37.04 | TJ` | trix |
| 14:37.15 | jwh | same thing, negates the need to know what you're doing :P |
| 14:37.19 | [TK]D-Fender | TJ`: Trixbox = FreePBX |
| 14:37.22 | walhala | E-bola: so which version a of * do you use ? |
| 14:37.25 | *** join/#asterisk af_ (n=getsmart@81-174-45-5.dynamic.ngi.it) |
| 14:37.32 | TJ` | [TK]D-Fender did u know grass was green too? |
| 14:37.48 | E-bola | walhala: 1.4 |
| 14:37.55 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:38.06 | [TK]D-Fender | TJ`: Indeed. and do you know why the sky is blue? Because if it were green we wouldn't know when to stop mowing! |
| 14:38.16 | walhala | E-bola: i'm running 1.2 and work but you have to patch your chan_sip.c |
| 14:38.28 | TJ` | the sky is actually black |
| 14:38.31 | pj_ | Blue is green in japan. |
| 14:38.32 | blackbyte01 | I am developing a php script to manage my voip system based on asterisk... I need commands (or functions) to log every call done by an account... |
| 14:38.47 | blackbyte01 | can someone halp me? |
| 14:38.51 | blackbyte01 | *e |
| 14:38.56 | Qwell[] | ~cdr |
| 14:38.56 | jbot | well, cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 14:39.07 | walhala | blackbyte01: use the cdr in sql |
| 14:39.28 | CelticSoul | Hi, If I use PHP AGI to dial an extension (for example 33333), what will happen when the user at 33333 pick up the phone? |
| 14:39.38 | TJ` | .... |
| 14:39.41 | TJ` | he'll say hello |
| 14:39.43 | blackbyte01 | what's that? |
| 14:39.56 | walhala | blackbyte01: cdr is Call Detail Record |
| 14:40.12 | walhala | you can store it into a database such as MySQL |
| 14:40.26 | CelticSoul | I meant What happen in PHPAGI and Asterisk? |
| 14:40.27 | pj_ | [TK]D-Fender: can the fact that I have no sound card explains why I get no sound in voicemail ? |
| 14:40.30 | blackbyte01 | thanks! |
| 14:40.33 | blackbyte01 | i will! |
| 14:40.47 | walhala | blackbyte01: take a look on voip-info at realtime |
| 14:40.47 | pj_ | (I remember it had some impact "back in the days") |
| 14:40.49 | [TK]D-Fender | pj_: No. |
| 14:40.55 | pj_ | :( |
| 14:41.02 | pj_ | I wish you'd say yes :/ |
| 14:41.12 | pj_ | At least I'd know |
| 14:41.13 | Qwell[] | [TK]D-Fender: it isn't too late |
| 14:41.24 | jkiff | You wish to be lied to? |
| 14:41.27 | jkiff | Interesting. |
| 14:41.49 | Mercestes | pj_: Could it be one way audio? |
| 14:42.02 | pj_ | It could |
| 14:42.15 | pj_ | My box doesn't want to tell me if it hears me or not though |
| 14:42.18 | pj_ | No nat |
| 14:42.24 | pj_ | but no packet leaving asterisk box either |
| 14:42.27 | walhala | some sccp or skinny users here ? |
| 14:42.48 | pj_ | The server is in a vmware box... I have replicated the same exact conf on a physical box and it works fine |
| 14:43.09 | Mercestes | pj_: Maybe it is vmware then. |
| 14:43.26 | *** join/#asterisk xacatecas (n=jkroon@c1-218-13.tbnb.isadsl.co.za) |
| 14:43.31 | pj_ | doesn't explain much... |
| 14:44.27 | walhala | obody use a cisco 79XX with asterisk ? |
| 14:45.37 | xacatecas | hi folk, i'm busy trying to get asterisk up and running on a gentoo installation. I've now managed to finally compile version 1.2.17 (there was an issue with the h323 module it seems) and have it installed. however, i'm seeing a version of 1.4.x as well, I've been trying to figure out what the difference between 1.2.x and 1.4.x is, so far with no succes - anybody mind explaining? |
| 14:45.44 | holiday_42 | pj_: if conf is same, perhaps something else.. errant firewall rules maybe? |
| 14:46.22 | pj_ | no firewall |
| 14:46.25 | pj_ | :/ |
| 14:46.29 | CelticSoul | someone please |
| 14:46.58 | pj_ | and packets do not appear on a local tcpdump on the * server |
| 14:46.59 | holiday_42 | pj_: is the file size of the vm zero? |
| 14:47.17 | pj_ | you mean space left ? |
| 14:47.30 | pj_ | there's two whole gigs free |
| 14:47.59 | holiday_42 | nope... i mean if you look at... uh /var/spool/asterisk/voicemail (or whatever) and look at the voice mail files |
| 14:48.12 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:49.21 | *** join/#asterisk inv_arp[work] (n=junya@c-71-229-122-61.hsd1.fl.comcast.net) |
| 14:52.12 | *** join/#asterisk ^^ARcANgEL^^ (n=arcangel@189.129.88.8) |
| 14:52.32 | ^^ARcANgEL^^ | hello |
| 14:53.01 | Mercestes | hello |
| 14:53.12 | pj_ | Ohh |
| 14:53.19 | jwh | :o |
| 14:53.21 | pj_ | holiday_42: thing is, it doesn't even play "password" |
| 14:53.34 | pj_ | well it does play, except there's no sound coming out |
| 14:53.46 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 14:53.50 | Qwell[] | yeah, that's one-way audio for sure |
| 14:53.55 | ^^ARcANgEL^^ | somebody podria to help me, is that i am obstructed installs asterisk but not to form it |
| 14:54.11 | ^^ARcANgEL^^ | ??? |
| 14:54.14 | Qwell[] | what? |
| 14:54.30 | _DAW | wow |
| 14:54.34 | Mercestes | Did you ./configure? |
| 14:54.36 | Qwell[] | babelfish FTW |
| 14:54.53 | Qwell[] | Mercestes: how the heck did that make any sense to you? |
| 14:55.01 | pj_ | Qwell[]: and any reason why it should not even send the RTP packets ? |
| 14:55.05 | ^^ARcANgEL^^ | that if somebody podria to help with the installation of the servant asterisk me |
| 14:55.12 | Mercestes | Qwell[]: google translate, gibberish to english. |
| 14:55.23 | pj_ | I mean, I can understand one way if it sent packets to the wrong adress, wrong port, or had a NAT or firewall |
| 14:55.31 | Qwell[] | Mercestes: god bless google |
| 14:55.39 | Mercestes | god bless it! |
| 14:55.47 | pj_ | But packets are not even trying to leave the server... LAZY PACKETS !! /me starts whiping the packets |
| 14:56.07 | Mercestes | ^^ARcANgEL^^, what wiki, howto, or instructions are you looking at? |
| 14:56.26 | CelticSoul | Hi, If I use PHP AGI to dial an extension (for example 33333), what will happen to PHPAGI and Asterisk when the user at 33333 pick up the phone? |
| 14:57.00 | [TK]D-Fender | ^^ARcANgEL^^: Pick another language because nobody will understand your english. |
| 14:57.03 | Mercestes | CelticSoul, Hang there until dial completes. |
| 14:57.25 | Mercestes | [TK]D-Fender, LOL. |
| 14:57.27 | Qwell[] | ^^ARcANgEL^^: what is your native language? |
| 14:57.44 | Mercestes | ^^ARcANgEL^^, what is your prime directive? |
| 14:58.06 | [TK]D-Fender | Mercestes: "Take us to your leader!" |
| 14:58.30 | pj_ | All the rtp paketz are belong to us |
| 14:58.44 | holiday_42 | pj_:how do you know that no packets leave? seems like that and "well it does play, except there's no sound coming out" seem contradictory? |
| 14:58.45 | ^^ARcANgEL^^ | spanish but a little english |
| 14:58.55 | Qwell[] | ^^ARcANgEL^^: very little english. Please speak spanish. |
| 14:59.05 | Mercestes | que es podria? |
| 14:59.28 | ^^ARcANgEL^^ | es que como vi que hablaban ingles por eso |
| 15:00.02 | pj_ | holiday_42: it does "-- Playing password" |
| 15:00.19 | pj_ | And no packet leave because I don't see any on any interface with tcpdump |
| 15:00.24 | Qwell[] | pj_: turn on rtp debug, and see where it's trying to send the packets |
| 15:00.31 | pj_ | it's not trying |
| 15:00.35 | pj_ | (tried rtp debug) |
| 15:00.49 | mihinomenest | I should be able to do something like, "dtmfmode=inband,rfc2833" in sip.conf, right? |
| 15:01.04 | Qwell[] | mihinomenest: I don't believe so |
| 15:01.11 | pj_ | it only gets the packets, never try to give them back |
| 15:01.13 | pj_ | evil * |
| 15:01.13 | Mercestes | pj_: =/ so there isn't even an RTP stream attempted? |
| 15:01.18 | mihinomenest | I say then, "grumble." |
| 15:01.21 | pj_ | doesn't seem so |
| 15:01.29 | Qwell[] | mihinomenest: is there a reason you need to be sending both? |
| 15:01.30 | Mercestes | mihinomenest, dtmfmode=auto |
| 15:01.55 | pj_ | if you mind to take a look : http://pastebin.ca/578979 |
| 15:02.22 | *** part/#asterisk ^^ARcANgEL^^ (n=arcangel@189.129.88.8) |
| 15:02.49 | pj_ | oh, wait, I got one Sent RTP packet |
| 15:02.53 | *** join/#asterisk dcm_ (n=dcm@207.59.3.77) |
| 15:03.42 | *** join/#asterisk rgsteele (n=rgsteele@nat-pool.agora-net.com) |
| 15:03.58 | mihinomenest | Qwell[]: well, no. |
| 15:04.09 | Mercestes | how come your voicemail log is like a dozen lines, and my voicemail log is like....2 lines? |
| 15:05.18 | rgsteele | Hey folks. I've got several analog lines from the telco going into a TDM400P in my asterisk box. The problem I've got is horrible echo on the internal side - clients calling in hear no echo. But, internally if our mouths get too close to the phones, or we speak too loudly, we get a really bad echo on our end. I've tried futzing with echocancel, echotraining, rx and tx gain, and still... |
| 15:05.20 | rgsteele | ...can't resolve it. Any suggestions? |
| 15:05.52 | mihinomenest | the problem is, the voip provider that we resell to our customers demands that we use linksys PAP2s set to automatically download their configs via TFTP. |
| 15:05.56 | rgsteele | Asterisk 1.2.13 and zaptel 1.2.11 |
| 15:05.59 | Mercestes | rgsteele, handset or speakerphone? |
| 15:06.06 | rgsteele | Mercestes: Handset. |
| 15:06.12 | Mercestes | hrm. |
| 15:06.15 | Mercestes | what type of phone? |
| 15:06.26 | pj_ | Mercestes: no idea... trixbox 2.2 here |
| 15:06.40 | rgsteele | Several different types. One's a Sipura SPA-841, the others are a variety of Cisco's. |
| 15:07.03 | mihinomenest | well, their default config specifies DTMF via SIP INFO, and apparently, if you call from one of their SIP "circuits" to another, the DTMF is never changed from SIP to whatever the other side needs. |
| 15:07.17 | Mercestes | pj_: oh, that makes it easy then. |
| 15:07.19 | Mercestes | ~trixbox |
| 15:07.20 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 15:07.25 | mihinomenest | so, my PBX is looking for inband and ignoring my customer's key presses. |
| 15:07.38 | pj_ | Well it's not trixbox related at all |
| 15:07.44 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:07.48 | mihinomenest | of course, the provider's Voicemail server is ignoring all DTMF, so it's really their problem. |
| 15:07.49 | [TK]D-Fender | Mercestes: I'd have done that, but he is decidedly NOT a chump even though using Trxbox.... |
| 15:07.50 | Mercestes | doesn't matter. It's a mess to troubleshoot |
| 15:07.57 | jwh | set the correct mode on asterisk and voip phones? |
| 15:08.10 | Mercestes | pj_, honestly tho, I think vmware is doing something silly with yoru network interface. |
| 15:08.14 | jwh | like, i use info |
| 15:08.15 | Cyber-Dogg | does anyone have any good references on how to setup asterisk with MSSQL? |
| 15:08.18 | jwh | but my providers use inband etc |
| 15:08.25 | Cyber-Dogg | I've seen some on PostgreSQL and MySQL |
| 15:08.25 | Mercestes | pj_, I will *bet* it has something to do with natting between your virtual interface and your *real* interface. |
| 15:08.28 | [TK]D-Fender | pj_: I suspect the same (VMWare) |
| 15:08.30 | jwh | just setting the right mode in sip.conf sorts out any interactivity problems |
| 15:08.36 | Cyber-Dogg | should ODBC work the same with MSSQL? |
| 15:08.38 | pj_ | Mercestes: eating the RTP packets before they come out from the linux server and only one way ? |
| 15:08.40 | holiday_42 | pj_: *shrugs* vm works here on vmware, but moh very choppy |
| 15:08.58 | Mercestes | pj_: Or in, really, as I'm not 100% sure that even works |
| 15:09.02 | pj_ | I have an ESX server, I want to give it a shot |
| 15:09.10 | Mercestes | go for it. |
| 15:09.19 | mihinomenest | jwh: if I set "rfc2833" in my sip.conf, no one works, not even incoming calls from PSTN works. |
| 15:09.21 | pj_ | And there is no natting |
| 15:09.25 | mihinomenest | I have to have it set to inband. |
| 15:09.42 | Mercestes | rgsteele, Well, thsi is the standard stuff but, give fxotune a try, and check yoru interrupts |
| 15:09.44 | jwh | mihinomenest: inbound you will need to match your provider |
| 15:09.44 | holiday_42 | meh, it was discussed on the list, virtualization seems to mess up the timing pretty |
| 15:09.45 | *** join/#asterisk svenna_ (n=svenna@p548D2E1A.dip0.t-ipconnect.de) |
| 15:09.45 | mihinomenest | and, when I set it to inband, voip customers from the same provider can't navigate my IVR. |
| 15:09.51 | jwh | outbound can be set differently |
| 15:10.00 | pj_ | holiday_42: timing as in clock time ? |
| 15:10.11 | mihinomenest | outbound doesn't matter. |
| 15:10.22 | jwh | ok, well what do you providers use? |
| 15:10.24 | holiday_42 | pj_: that too |
| 15:10.29 | jwh | you need to have the same setting as they do |
| 15:10.30 | Mercestes | mihinomenest, try dtmfmode-auto |
| 15:10.36 | pj_ | and you're basically saying that the sound should be choppy |
| 15:10.39 | pj_ | it's not, I got none |
| 15:10.44 | jwh | however, if you pass IVR onto something else, then that may cause issues |
| 15:10.45 | pj_ | I wish I had choppy |
| 15:10.53 | holiday_42 | :) |
| 15:10.55 | pj_ | then I could say it's because of vmware |
| 15:11.08 | pj_ | But the packets not being emitted by asterisk is another problem |
| 15:11.17 | holiday_42 | agreed |
| 15:11.18 | pj_ | which has nothing to do with trixbox, or vmware |
| 15:11.32 | mihinomenest | jwh: and for inbound, I do. all of my stations are set to rfc2833 because it seems like that works best. |
| 15:11.45 | pj_ | or maybe, but then give me a "reasonable cause" beside "it's black magic, don't go there" |
| 15:12.08 | jwh | mihinomenest: hm |
| 15:12.38 | holiday_42 | pj_: are you using asterisknow pre built vmware image? |
| 15:13.06 | pj_ | nope |
| 15:13.29 | pj_ | I had to install from scratch because the prebuilt image is not tailored for ESX servers |
| 15:13.36 | mihinomenest | jwh: apparently, we've got the 2nd highest guy at the voip provider working on it. |
| 15:13.36 | holiday_42 | oic |
| 15:13.49 | jwh | mihinomenest: hehe |
| 15:14.16 | rgsteele | Mercestes: Ah, I found a solution that seems to work - I set the txgain to -4.5 |
| 15:14.35 | rgsteele | Mercestes: I noticed only when I raised my voice or spoke really closely to the handset that I got awful feedback. |
| 15:14.48 | rgsteele | So, I turned that down, and I could still hear myself well on the cell phone I called in from. |
| 15:14.53 | holiday_42 | pj_: can you easily migrate the virtual machine to vmware server or workstation? |
| 15:15.28 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:15.37 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-70-18-14-219.nycmny.east.verizon.net) |
| 15:15.51 | *** join/#asterisk Splat (n=splat@home.heehawhills.com) |
| 15:16.50 | Mercestes | rgsteele: Hrm, as long as it works. :) |
| 15:17.04 | pj_ | it's "doable" yes |
| 15:17.17 | rgsteele | Mercestes: Yeah. To hell with all that "in theory" stuff ;) |
| 15:17.28 | Mercestes | rofl |
| 15:17.30 | pj_ | Oh wait, I got _one_ RTP packet :/ |
| 15:18.02 | Mercestes | pj_: then quit complaining! :P |
| 15:18.29 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 15:18.36 | pj_ | :) |
| 15:19.13 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
| 15:20.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:20.40 | falz | morning. Is tehre any hacky way to background an agi() call? I'm converting Read() to an agi script that uses swift from bourne. (I can't use Swift() as I can't upgrade to 1.4.x yet) |
| 15:21.00 | falz | so, any way to read text like Background() or WaitExten() does, but while an AGI is running that's playing voice |
| 15:21.23 | ManxPower | falz: you cannot background an AGI. |
| 15:21.40 | *** join/#asterisk gisasi (n=chatzill@ip-240-130.sn2.eutelia.it) |
| 15:21.53 | ManxPower | Your agi can put it'self into the background, just like any other daemon, but it would usually be bad idea. |
| 15:21.59 | falz | is there any command that will background itsself and wait for input PRIOR to the agi, or would the agi simply always take the foreground? |
| 15:22.10 | falz | yea, that sounds like a lot of orhpaned processes |
| 15:22.17 | falz | and just messier than I'd like |
| 15:22.18 | ManxPower | falz: ALL things in the dialplan take place in the forground. |
| 15:22.39 | ManxPower | even Background() runs in the corefround |
| 15:22.43 | ManxPower | foreground, that is. |
| 15:22.44 | falz | Background() is the type of functionality I'm looking for obviously |
| 15:22.57 | ManxPower | falz: so why don't you use it? |
| 15:22.58 | falz | but without playing back a pre-recorded .gsm |
| 15:23.08 | JerJer | eeek - If anyone is flying united today, stay home! |
| 15:23.11 | ManxPower | then use Read or Waitexten |
| 15:23.22 | falz | the stuff Im playing back isn't prerecorded. |
| 15:23.30 | JerJer | Fox just alerted that United has canceled all flights |
| 15:23.34 | falz | it's sounding like I'll have to prerecord things. |
| 15:23.49 | falz | or upgrade to 1.4 if Swift() has any background type of options |
| 15:23.52 | ManxPower | falz: Make up your mind. Do you want to play audio or not? |
| 15:24.04 | falz | of course I do. but I'd like the audio to be interrupted by user input |
| 15:24.18 | ManxPower | best of luck with that. |
| 15:24.33 | ManxPower | The way I solved the issue many years ago is still, in my opinion, the best way. |
| 15:24.46 | falz | and that is _______? |
| 15:24.53 | ManxPower | Unless your text-to-speech audio file rendering takes more than 2 seconds. |
| 15:25.16 | ManxPower | falz: generate the audio file, use background, delete the audio file. |
| 15:25.33 | Qwell[] | JerJer: what, why? |
| 15:25.34 | falz | do you have some delayed backgrounded job that delets it? |
| 15:26.10 | Mercestes | Falz: Just use System() at that point. |
| 15:26.11 | falz | or are you using like agi_uniqueid, passing it to a new variable, and after the Background() calling a new script to rm? |
| 15:26.14 | JerJer | Qwell[]: major 'system' problem - more detail soon... they just did a fox news alert |
| 15:26.21 | Qwell[] | huh |
| 15:26.23 | JerJer | i'm presuming computer problems |
| 15:26.28 | Qwell[] | good thing my wife is flying delta :P |
| 15:26.32 | [TK]D-Fender | Qwell[]: terrists! |
| 15:26.44 | Mercestes | Well, today is 6-20. |
| 15:26.57 | Qwell[] | Mercestes: that makes perfect sense |
| 15:27.00 | anonymouz666 | JerJer! |
| 15:27.00 | Qwell[] | except that it doesn't |
| 15:27.01 | Mercestes | Good day to fly a plane into something since it has such a strong, numerical meaning |
| 15:27.14 | ManxPower | falz: you can put the file in a temp directory, then have a cron job that deltes the files. Yo can also have exten => h do the deleteions |
| 15:27.16 | Mercestes | qwell[]: Exactly! |
| 15:27.30 | Mercestes | Or you can delete it right after you background it |
| 15:27.43 | Mercestes | anything involving rm -dvfr should suffice. |
| 15:27.45 | falz | I'll experiment with that fun stuff. |
| 15:27.48 | Qwell[] | I made a script that used festival... I did some slick md5 hashing stuff on it |
| 15:27.57 | Qwell[] | so if the same text was ever repeated, it wouldn't re-generate it |
| 15:28.02 | falz | know offhand if the built in Swift() stuff can be backgrounded, or same issues there? |
| 15:28.22 | ManxPower | falz: no apps do that |
| 15:28.26 | falz | ok. |
| 15:28.27 | [TK]D-Fender | Qwell[]: Oh sorry... I should have said Democratic Atheist Lesbian ACLU spokespeople en-route to an Anti-war rally! ;) |
| 15:28.30 | Mercestes | ManxPower, don't do that..:( It makes my files hang out and refuse to be deleted. |
| 15:28.50 | ManxPower | Mercestes: exactly, you short sighted pbx admin |
| 15:28.55 | falz | well, no matter what there will have to be some cron job anyway, I wouldnt trust anything deleting. find /tmp/dir -name "*.wav" -mtime +1 |
| 15:28.57 | Mercestes | :( |
| 15:29.54 | Mercestes | I'm feeling violated again...am I in the right channel? This is ASTERISK isn't it? *looks around* oh..good. |
| 15:30.01 | ManxPower | Mercestes: Good. Maybe next time you'll think about the implications of doing something wrong. |
| 15:30.29 | falz | ManxPower: when you're creating your .wav files, are you just using agi_uniqueid to name them? That var is new to me as of this morning, but it seems unique enough :) |
| 15:30.39 | ManxPower | Mercestes: Do you also add "r" to all your Dial lines and use exten => _. all over the place too? |
| 15:31.01 | ManxPower | falz: It was 4 years ago. I don't remember. |
| 15:31.09 | falz | oh. so not using it anymore? |
| 15:31.25 | Qwell[] | falz: when I was doing my festival script, I just named them the md5 of the text |
| 15:31.38 | falz | hmm yea, an md5 shouldn't take long |
| 15:31.44 | Qwell[] | it doesn't |
| 15:32.05 | ManxPower | falz: You entered in your Zip Code (USA ONLY) and the app would connect to weather.com, screen scrape the required text, sent it to Cepstral (older verison), then played the file, then deleted the file. |
| 15:32.09 | Qwell[] | even with long text, it took less than half a second to search through existing files and if needed, generate the file |
| 15:33.11 | falz | hmm I'll look at some of the old festival stuff online and see if there's some prewritten stuff I can.. "borrow" |
| 15:33.41 | falz | seems like a chicken/egg thing (or I'm dumb) in knowing the md5sum prior to looking for it |
| 15:33.55 | ManxPower | What was the previous name "swift"? |
| 15:33.55 | Qwell[] | falz: You always md5 the text |
| 15:34.00 | falz | ahhh |
| 15:34.38 | falz | hence, the dumb disclaimer. |
| 15:34.40 | ManxPower | if I know that I can prolly fine my ancient script somewhere. |
| 15:34.46 | falz | that's a good starting point. |
| 15:35.27 | *** join/#asterisk galeras (n=root@201.244.240.173) |
| 15:35.57 | Mercestes | ManxPower, only when it's your mom calling. :P |
| 15:37.55 | [TK]D-Fender | ManxPower: Retreat!!! He used the "Ur Mom" defense, you have no chance to survive! Make your time! |
| 15:38.12 | Mercestes | bwahahaha |
| 15:38.47 | Mercestes | yay! i'm blonde! |
| 15:38.51 | ManxPower | [TK]D-Fender: he drops 5 hit points for using "Ur" |
| 15:40.20 | [TK]D-Fender | whee! |
| 15:45.23 | Mercestes | whee. |
| 15:52.55 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 15:55.07 | *** join/#asterisk FreezeS (n=bla@82.208.157.125) |
| 15:55.11 | FreezeS | hey guys |
| 15:55.21 | FreezeS | I've got some core dumps from asterisk when it crashed |
| 15:55.26 | FreezeS | how do I analize them ? |
| 15:56.26 | pj_ | Ok, update to the captivating story of my lazy asterisk.... it seems to block on playing sound... the single RTP packet I get must be from vm-password, but then if I enter the password asterisk stay blocked on vm-youhave and doesn't go further |
| 15:56.54 | pj_ | Any idea why * would block on playing a sound file ? (permission, file size, checked) |
| 15:58.17 | [TK]D-Fender | pj_: any chance of a VLAN issue? |
| 15:59.05 | pj_ | Well I would have seen them leaving the server at least... but now I'm really sure about it |
| 15:59.19 | pj_ | because it freezes at "-- Playing 'vm-youhave' (language 'en')" |
| 16:00.18 | pj_ | where it should go on and go "-- Playing 'digits/3'" instead |
| 16:01.12 | pj_ | so it looks more like asterisk block reading the sound file |
| 16:02.30 | *** join/#asterisk Waverly360 (n=Waverly3@231.sub-70-222-77.myvzw.com) |
| 16:03.15 | Qwell[] | Nugget: ping |
| 16:03.31 | Waverly360 | Hey guys, I'm doing some research on "tie lines" and what I've found just states that they are just PRI's connecting one PBX to another..is that correct? |
| 16:03.59 | Mercestes | pj_: Permissions issue? |
| 16:04.37 | pj_ | nope, no change, and chmoded 775 just to make sure |
| 16:04.38 | [TK]D-Fender | Waverly360: that is 1 kind. |
| 16:04.43 | pj_ | good user, good group |
| 16:04.44 | pj_ | :( |
| 16:05.10 | [TK]D-Fender | Waverly360: You could also be dealing with Frame relay gear, DS0 E&M boxes, TONS of stuff |
| 16:05.14 | *** join/#asterisk Yozhik (n=Yozhik@72.171.70.169) |
| 16:05.30 | jwh | hm |
| 16:05.32 | Yozhik | Hello |
| 16:05.38 | jwh | any major carriers/operators in here? |
| 16:05.40 | Yozhik | I am new using this chat |
| 16:05.45 | jwh | (who use asterisk in production) |
| 16:06.06 | Waverly360 | [TK]D-Fender: Yeah, I'm trying to figure out what I have. We're looking to connect an asterisk box to another PBX via a tie line. I don't know exactly what they're using though. I have the model number of the card in the PBX though. |
| 16:06.17 | Yozhik | joined in to ask questions regarding using asterisk on MPLS |
| 16:06.25 | jwh | Yozhik: shoot |
| 16:06.29 | [TK]D-Fender | Waverly360: How about boring old analog? |
| 16:06.34 | Yozhik | anybody kow anything about it? |
| 16:06.51 | jwh | Yozhik: what do you want to know? |
| 16:06.53 | Yozhik | Well, I have already setup asterisk servers |
| 16:06.57 | Yozhik | (mostly AAH) |
| 16:07.05 | Yozhik | an dnow I want to set it up in a company |
| 16:07.05 | Waverly360 | [TK]D-Fender: the tie lines are already in place. I've tried analog before with another pbx and we had a lot of problems with disconnect supervision |
| 16:07.07 | Nugget | Qwell[]: pong |
| 16:07.19 | Yozhik | and they are about to implement MPLS to connect their remote offices |
| 16:07.25 | Qwell[] | Nugget: flightaware needs a way to call people when a flight lands/takes off :p |
| 16:07.28 | Yozhik | my question is: is there a difference? |
| 16:07.28 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 16:07.38 | Nugget | yeah, I've thought about that, but it's low on the list. |
| 16:07.38 | jwh | Yozhik: shouldn't matter |
| 16:07.42 | [TK]D-Fender | Waverly360: well a "tie line" is not a MEANS, its a PURPOSE |
| 16:07.44 | Yozhik | not at all? |
| 16:07.47 | jwh | Yozhik: as long as the providers network isn't congested |
| 16:07.55 | Yozhik | perfect J |
| 16:08.08 | Yozhik | thanks |
| 16:08.13 | Waverly360 | [TK]D-Fender: Oh..so I need to figure out what they're using for tie lines |
| 16:08.15 | [TK]D-Fender | Waverly360: Just like people who think they bought an RRSP ;) |
| 16:08.34 | *** part/#asterisk Yozhik (n=Yozhik@72.171.70.169) |
| 16:08.56 | Waverly360 | [TK]D-Fender: hah hah |
| 16:08.56 | [TK]D-Fender | Waverly360: I caught a perch on my fishing "line"... perhaps you could use that! ;) |
| 16:08.56 | *** join/#asterisk Yozhik (n=Yozhik@72.171.70.169) |
| 16:09.17 | Waverly360 | [TK]D-Fender:Ok ok, so my knowledge of terminology sucks..I get it :P |
| 16:10.07 | Yozhik | One more question |
| 16:10.19 | festr__ | hello, any idea, why in 1.4.4 does not work Pickup (show channels: SIP/festrntb-b68257f 117@macro-dial_ext_1 Ring Dial(SIP/festrntb|60|jtT) |
| 16:10.29 | festr__ | Pickup(117) should work is it? |
| 16:10.30 | Qwell[] | Nugget: and the search thing is totally not showing me this flight :p |
| 16:10.44 | Yozhik | Is there a big difference in the programming and configuring of asterisk with business edition? |
| 16:10.50 | file | festr__: specify the context. |
| 16:10.58 | festr__ | file: i've tried that |
| 16:11.00 | festr__ | file: no success |
| 16:11.17 | festr__ | file: if you mean Pickup(117@macro-dial_ext_1) |
| 16:11.19 | jwh | Yozhik: not sure, only used the normal version, but i'd assume its the same with more features |
| 16:11.25 | file | that's not the context |
| 16:11.26 | Waverly360 | [TK]D-Fender: Well, from what I can gather, it looks like their using a pri out of the back of a cisco 2600 into their current pbx to a rdtu1 card |
| 16:11.32 | Yozhik | I am interested in implementing in the company but feel a little insecure to do it myself and so far I read at Digiums |
| 16:11.33 | file | where did you call Macro from? |
| 16:11.39 | FreezeS | my asterisk crashes a lot here: 0xb6dac307 in set_pointer () from /usr/lib/asterisk/modules/format_mp3.so |
| 16:11.44 | Yozhik | business edition is quiet easy |
| 16:11.59 | Waverly360 | [TK]D-Fender: I'm guessing that I can plug that into a pri card on an asterisk box, and configure it to work the same. |
| 16:12.00 | FreezeS | and I get this interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 |
| 16:12.04 | festr__ | file: so thats the context? i'll try thank you! |
| 16:12.05 | neverblue | good softphone? |
| 16:12.09 | neverblue | ekiga? |
| 16:12.11 | festr__ | file: but in 1.2 it works |
| 16:12.12 | [TK]D-Fender | Waverly360: Entirely likely |
| 16:12.16 | FreezeS | what program should I use to encode proper mp3 files for asterisk ? |
| 16:12.33 | Qwell[] | Nugget: yeah, no info for DAL1185 today :p |
| 16:12.34 | [TK]D-Fender | FreezeS: LAME |
| 16:12.47 | FreezeS | that's what I've been using :( |
| 16:12.49 | Waverly360 | [TK]D-Fender: guess I won't know for sure unless I test it out... |
| 16:12.51 | denon | I agree, MP3s are lame .. |
| 16:12.53 | denon | use ulaw |
| 16:12.53 | FreezeS | what is the proper format ? |
| 16:12.53 | denon | :) |
| 16:13.03 | FreezeS | hmm, actually, very good idea |
| 16:13.08 | Yozhik | JWH: have you set up an asterisk server with AAH? |
| 16:13.16 | Waverly360 | [TK]D-Fender: Ok, I feel a bit better about this now...maybe one day you won't make fun of me when I ask these n00b questions :) |
| 16:13.17 | jwh | Yozhik: no, I just use vanilla asterisk |
| 16:13.18 | FreezeS | what windows program can I use for that ? |
| 16:13.19 | [TK]D-Fender | denon: Amazing how I say so much with so few words ;) |
| 16:13.28 | [TK]D-Fender | FreezeS: **LAME** |
| 16:13.29 | Yozhik | forgive my ignorence |
| 16:13.36 | Yozhik | but how is it? |
| 16:13.38 | denon | hehe |
| 16:13.41 | Yozhik | is there a link to download it? |
| 16:13.42 | FreezeS | [TK]D-Fender: to convert into ulaw |
| 16:13.55 | jwh | Yozhik: I meant, standard asterisk distribution from www.asterisk.org |
| 16:14.10 | Yozhik | I see |
| 16:14.33 | Yozhik | I will dowload it. Is there any handbook or manual which might guide me? |
| 16:14.45 | Yozhik | to set uo the vanilla edition? |
| 16:14.52 | jwh | voip-info.org is a fairly good resource |
| 16:15.02 | jwh | but the config examples that come with it are good |
| 16:15.08 | [TK]D-Fender | Yozhik: ... |
| 16:15.09 | [TK]D-Fender | ~book |
| 16:15.10 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:15.14 | Yozhik | Yes |
| 16:15.19 | Yozhik | Got it |
| 16:15.24 | Yozhik | to be honest |
| 16:15.26 | [TK]D-Fender | Yozhik: Get reading then |
| 16:15.34 | jwh | hm |
| 16:15.36 | Yozhik | my deficiency my be that I have never studied linux |
| 16:15.43 | jwh | is that the same one that was published a few yars ago? |
| 16:15.44 | Yozhik | an dhave been trying to learn on my own |
| 16:15.45 | jwh | years* |
| 16:15.57 | [TK]D-Fender | Yozhik: So NOW you're deciding to be honest? ;) |
| 16:16.02 | Yozhik | The book is from year sago |
| 16:16.13 | [TK]D-Fender | Yozhik: The book is still fine. |
| 16:16.14 | `Sean | Guys [TK]D-Fender can i have this work like this |
| 16:16.15 | `Sean | exten => s,1,Answer() |
| 16:16.15 | `Sean | exten => s,n,Playback(rh-british) |
| 16:16.15 | `Sean | exten => s,n,Dial(Zap/1,45) |
| 16:16.21 | [TK]D-Fender | Yozhik: I suck at linux and do just fine. |
| 16:16.22 | `Sean | instead oof Dail(Zap/1) |
| 16:16.25 | `Sean | can i pout a phone number? |
| 16:16.29 | `Sean | to forward it to |
| 16:16.32 | `Sean | like my cell number? |
| 16:16.40 | Yozhik | interesting, I feel intimidated to try the original version because |
| 16:16.45 | Yozhik | I did not study linux |
| 16:16.56 | [TK]D-Fender | `Sean: Assuming Zap/1 is an FXO portm sure |
| 16:17.01 | Yozhik | and I have done fine with AAH |
| 16:17.21 | Yozhik | which forced me to program some stuff inlinux with commands |
| 16:17.23 | [TK]D-Fender | Yozhik: Well don't. You need to learn a FEW basics |
| 16:17.41 | Yozhik | Any suggestiong for new bie? |
| 16:17.52 | Yozhik | website, book, something? |
| 16:17.55 | Mercestes | Yozhik, vanilla asterisk |
| 16:18.15 | festr__ | file: the problem is, that i've context for every extension. when calling from exten to exten it will call from that context macro(dial exten..) so i cannot predict context :( |
| 16:18.25 | Yozhik | so fa I understand vanilla asterisk is the standard version for download. |
| 16:18.29 | Yozhik | is that correct? |
| 16:18.44 | Mercestes | festr__, That would be your problem then. I have no clue what the question was, but that's definately the problem. |
| 16:18.54 | jwh | vanilla just means plain, as in, the original asterisk |
| 16:19.18 | festr__ | Mercestes: :) |
| 16:19.32 | [TK]D-Fender | Yozhik: Sounds like what you're looking for. |
| 16:19.32 | `Sean | [TK]D-Fender anyway to forward without having FXO? |
| 16:19.39 | festr__ | so i've to backport working app_directedpickup from 1.2 to 1.4 then |
| 16:19.48 | `Sean | like is it possible i can forward it out to my celly via asterlink |
| 16:19.56 | [TK]D-Fender | `Sean: Ask yourself this "how is * going to send the call to me?" |
| 16:20.09 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 16:20.49 | Yozhik | any manual or procedure reference for vanilla version? |
| 16:20.50 | *** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ut.comcast.net) |
| 16:22.26 | Yozhik | Can I download it from asterisk.org? version Asterisk 1.4.5 ? |
| 16:22.30 | [TK]D-Fender | Yozhik: the BOOK, and for download & instal : www.asterisk.org |
| 16:23.00 | Yozhik | ok, perfect |
| 16:23.04 | Yozhik | got to work |
| 16:23.09 | joe | [TK]D-Fender: what's the largest call center you know of using * ? ie number of agents/calls? |
| 16:23.11 | Yozhik | I appreciate your help guys |
| 16:23.14 | joe | [TK]D-Fender: oh and hi btw :) |
| 16:23.15 | Yozhik | thank you |
| 16:23.22 | Yozhik | good luck |
| 16:23.32 | *** part/#asterisk Yozhik (n=Yozhik@72.171.70.169) |
| 16:23.36 | [TK]D-Fender | joe : not sure really |
| 16:25.53 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:25.53 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 16:28.06 | Mercestes | joe: Remind me waht acd and icd is again.. |
| 16:29.12 | joe | Mercestes: http://www.voip-info.org/wiki-ICD http://www.voip-info.org/wiki/view/ACD |
| 16:30.18 | Mercestes | joe: I didn't use ACD/ICD, I used queues and ring strategys |
| 16:30.26 | Mercestes | but, I beat 100 |
| 16:31.15 | [TK]D-Fender | Mercestes: By ACD I'm sure he means app_queue , ICD is 3rd party |
| 16:31.40 | joe | [TK]D-Fender: thanks :) |
| 16:31.53 | jwh | anyone here who uses asterisk in a production/carrier environment, and knows what sort of simaultaneous calls you can get from asterisk, whether that was clustered etc? |
| 16:32.29 | Qwell[] | Somewhere between 1 and an infinite number of calls, depending on hardware, bandwidth, and many other factors |
| 16:33.05 | *** join/#asterisk joebob777as7 (n=corn13re@67-42-57-190.eugn.qwest.net) |
| 16:34.36 | joebob777as7 | how do i record a call? someone told me *1 but when i press it in the person on the other end hears me and nothing seems to happen... can someone point me in the right direction? |
| 16:35.12 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 16:35.20 | jwh | joebob777as7: Monitor() |
| 16:35.43 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 16:35.52 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 16:35.56 | joebob777as7 | i don't want to record all calls only on demand... |
| 16:36.23 | jwh | mmm |
| 16:36.53 | [TK]D-Fender | joebob777as7: Go lookup "features.conf" on the WIKI |
| 16:37.05 | joebob777as7 | ok thanks |
| 16:37.10 | [TK]D-Fender | joebob777as7: and "show application dial" |
| 16:39.01 | joebob777as7 | [TK]D-Fender, where is the wiki? |
| 16:39.32 | [TK]D-Fender | ~wikis |
| 16:39.33 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 16:39.34 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:39.40 | Dandre | hello, |
| 16:41.03 | Dandre | when using the manager interface (AGI) what is the difference between update and append in updateconfig command? Update seem |
| 16:41.23 | Dandre | seems to append if the variable does not exists |
| 16:45.09 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 16:46.39 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 16:47.28 | *** join/#asterisk Alan_Hicks (n=alan@208.62.162.112) |
| 16:47.41 | *** join/#asterisk atumanov (n=atumanov@192.219.104.10) |
| 16:47.45 | Alan_Hicks | Howdy folks. This is slightly off-topic, but I figured this would be the best place to ask my question. |
| 16:48.50 | *** join/#asterisk zpertee (n=root@oh-69-34-21-229.sta.embarqhsd.net) |
| 16:48.54 | atumanov | I heard there's a moddable router which self-sufficiently could be used to run as an asterisk gateway? does anybody here know what that is? |
| 16:48.57 | Alan_Hicks | I have a customer that wants to update her phone system, and I've considered an Asterisk solution. Currently she has 24 PSTN lines, and would be better served with a PRI. Some other companies are quoting her on proprietary PBX systems, and they know they can make theirs work with a PRI (which would save my client money monthly). |
| 16:49.44 | zpertee | does anyone have or know of any example expect scripts that are used with asterisk |
| 16:50.08 | Alan_Hicks | I can purchase the equipment needed to test PSTN lines, and perhaps even a PRI, but I have no actual PRI lines coming into our business, or any other business that I know of. Is there anyway to setup a test box and test a PRI without commiting to an expensive lease from the phone company? |
| 16:50.33 | atumanov | is it linksys wrt54G? |
| 16:50.36 | rsd99 | i know they make isdn simulators would that help |
| 16:51.13 | Alan_Hicks | rsd99: I'm not certain. :^) I'm realy a newb to asterisk (but I have read the book!) and I know very little about digital voice lines. |
| 16:51.33 | *** join/#asterisk Slingky (n=Slingky@modemcable199.182-200-24.mc.videotron.ca) |
| 16:51.55 | [TK]D-Fender | Alan_Hicks: Whats to test? It just works... |
| 16:52.07 | Slingky | could someone tell me if linksys spa-2102 is best adapter for connecting analog phones and fax to asterisk ? |
| 16:52.34 | Alan_Hicks | [TK]D-Fender: I don't want to sell my client a solution that I haven't personally implimented and can garauntee will work. |
| 16:52.34 | [TK]D-Fender | Slingky: You should keep analog fax machines on a line of their own thats as far away from * as possible. |
| 16:52.55 | falz | hmm does Background() only playback .gsm's (not .wav)? |
| 16:53.07 | [TK]D-Fender | Alan_Hicks: We've done it, and continue to do so. You're chicken&egging yourself. |
| 16:53.11 | Alan_Hicks | In other words, I want to know something inside-out before I give it to a client. My reputation isn't worth chancing on anything. |
| 16:53.25 | [TK]D-Fender | Alan_Hicks: Lack of experience with * and telco gear is preventing you from DOING so. |
| 16:53.44 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:53.46 | [TK]D-Fender | Alan_Hicks: then sub-contract parts of it |
| 16:53.49 | Alan_Hicks | [TK]D-Fender: I wouldn't be as skittish if it were my business. |
| 16:53.59 | Alan_Hicks | That's an idea... |
| 16:54.05 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:54.33 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 16:54.52 | pj_ | Hear hear, I have found the root of the evil in my * box, and would like your advice on how to fix it... It seems that unloading the zaptel modules makes the sound come back ! |
| 16:55.18 | Alan_Hicks | I just don't want to get in the situation where I may be struggling to make something work, particularly with this client. The previous company she used sold her this horrible "solution" which they worked on for a year and never made work. |
| 16:55.33 | Alan_Hicks | Naturally, she's skittish of anything "new" and untested. |
| 16:55.55 | tzafrir_laptop | pj_, what is your timing source? ztdummy or a card? |
| 16:56.11 | pj_ | Hmmm... that could definitely be it |
| 16:56.27 | pj_ | Right now a "card", however it's a redfone box, and line is not plugged in yet |
| 16:56.36 | pj_ | so it's a ztd_eth card |
| 16:57.19 | Alan_Hicks | Can anyone link me to some particularly good documentation on PRIs? |
| 16:57.31 | tzafrir_laptop | pj_, what does zttest show? is the timing source working ok? |
| 16:57.43 | Alan_Hicks | The O'Reilly book didn't have a lot to say on that subject. |
| 16:58.05 | bkruse | it will :] |
| 16:59.40 | pj_ | ( tzafrir_laptop checking ...) |
| 16:59.47 | *** join/#asterisk marcan (i=1337@117.Red-88-5-77.staticIP.rima-tde.net) |
| 16:59.57 | pj_ | Anyway it sounds good.. it means I probably will have my problem solved when I plug in the line |
| 17:00.04 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 17:01.21 | galeras | Please be cool and give me your points to choose the right phone: Aastra, Grandstream, Linksys, Polycom, ... |
| 17:01.28 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 17:01.34 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 17:01.36 | pj_ | it seems to block and freeze on "measuring accuracy..." |
| 17:01.51 | Nuitari | ~phones |
| 17:01.52 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 17:02.10 | [TK]D-Fender | galeras: Polycom > All |
| 17:02.29 | [TK]D-Fender | ~gs |
| 17:02.29 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 17:02.31 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:02.58 | pj_ | we bought 50 of them and 1/4 were defective |
| 17:03.07 | pj_ | I mean, _more_ defective than the other ones |
| 17:04.01 | joebob777as7 | [TK]D-Fender, so do i need to create an extension to make Monitor() work i'm confused |
| 17:04.16 | *** join/#asterisk dennisharrison (n=dennisha@68-114-124-171.dhcp.slid.la.charter.com) |
| 17:04.22 | dennisharrison | man, I hate asking for help |
| 17:04.26 | dennisharrison | but I am pulling my hair out ;p |
| 17:04.42 | EmleyMoor | I appear to have a strange problem - whenever a certain number is dialed over my POTS line by Asterisk, I seem to be put through to a fax machine... is there any good way I can test what the FXO port is actually dialing? |
| 17:04.43 | dennisharrison | have an older box that lost power last night |
| 17:05.10 | galeras | EmlseMoor: set verbose 10 |
| 17:05.13 | dennisharrison | when it came back up it spews |
| 17:05.14 | dennisharrison | Jun 20 12:03:16 WARNING[12663] loader.c: Loading module chan_zap.so failed! |
| 17:05.21 | dennisharrison | so ... |
| 17:05.44 | Mercestes | dennisharrison, check your permissions. |
| 17:05.48 | galeras | Btw, Thanks, seems Polycom is the right path |
| 17:06.09 | dennisharrison | Mercestes, :) I hope you are right! |
| 17:06.13 | [TK]D-Fender | joebob777as7: Go read it all again. you're looking for Dynamic features, and the "wW" optiosn in app_dial |
| 17:06.23 | Mercestes | I'll bet /dev/zap is owned by root:root |
| 17:06.42 | Mercestes | or root:dialout more likely |
| 17:07.05 | dennisharrison | yeah modprobe tells me it can't locate chan_zap |
| 17:07.08 | dennisharrison | lemme check |
| 17:07.09 | tzafrir_laptop | adduser asterisk dialout |
| 17:07.15 | Mercestes | you don't modprobe chan_zap |
| 17:07.22 | tzafrir_laptop | # adds asterisk to the group dialout on debian |
| 17:07.22 | Mercestes | you modprobe zaptel and whatever driver yoru card uses |
| 17:07.26 | dennisharrison | ok |
| 17:07.28 | dennisharrison | zaptel then |
| 17:07.29 | dennisharrison | yes |
| 17:07.37 | Mercestes | Did zaptel load? |
| 17:07.41 | dennisharrison | yes |
| 17:07.45 | Mercestes | k. |
| 17:07.48 | tzafrir_laptop | your module's driver pull zaptel |
| 17:07.51 | Mercestes | now load up your zaptel stuff. |
| 17:08.03 | dennisharrison | ok it is loaded |
| 17:08.08 | Mercestes | now try to load asterisk |
| 17:08.19 | tzafrir_laptop | ztcfg ? |
| 17:08.23 | Mercestes | dunno if it's /etc/init.d/zaptel start or wahtever it is... |
| 17:08.33 | Mercestes | ztcfg == zaptel stuff btw. |
| 17:08.36 | Mercestes | >.> I hope |
| 17:08.44 | dennisharrison | ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) |
| 17:09.21 | dennisharrison | bad card? |
| 17:09.54 | Mercestes | Google the error |
| 17:10.27 | joebob777as7 | reading the wiki is great fun! stupid information database that is poorly outlined! |
| 17:10.32 | *** part/#asterisk joebob777as7 (n=corn13re@67-42-57-190.eugn.qwest.net) |
| 17:11.16 | dennisharrison | apparently this is because I just compiled 1.4 |
| 17:11.21 | dennisharrison | and I should be using 1.2 ? |
| 17:11.22 | Mercestes | ...wow, chatting somethign derogatory then ditching the channel... |
| 17:11.27 | rsd99 | anyone know a good site for ivr prompts? |
| 17:11.33 | Mercestes | that's kinda like kicking a retarded kid in the nutz and then running for it. |
| 17:11.43 | dennisharrison | Mercestes, lol, he did say he had fun though |
| 17:11.49 | dennisharrison | ... he has that going for him ;p |
| 17:11.59 | Mercestes | it is fun! I was advocating it. :D |
| 17:12.11 | dennisharrison | haha |
| 17:12.38 | Mercestes | http://www.syednetworks.com/asterisk-zaptelconf-configuration-with-x100p-fxo-and-tdm04b-cards/feed/ |
| 17:13.09 | Mercestes | http://www.google.com/search?hl=en&q=ZT_CHANCONFIG+Inappropriate+ioctl+for+device |
| 17:13.22 | Mercestes | I have to head to lunch soon but those results aren't looking real promising as a common error |
| 17:13.56 | dennisharrison | yeah |
| 17:13.58 | dennisharrison | doesn't seem so |
| 17:14.04 | dennisharrison | I am thinking the card is borked |
| 17:14.09 | dennisharrison | cheapo cards anyhow |
| 17:14.11 | dennisharrison | I hate fxo |
| 17:14.16 | dennisharrison | so backwards |
| 17:14.20 | [TK]D-Fender | dennisharrison: Did you upgrade your Zaptel at the same time? |
| 17:14.27 | [TK]D-Fender | ^^^^^ |
| 17:14.29 | dennisharrison | hey [TK]D-Fender ! ;p |
| 17:14.36 | dennisharrison | yes I just updated zaptel to 1.4 |
| 17:14.44 | Mercestes | oh... |
| 17:14.48 | Mercestes | then it is a common error... |
| 17:14.49 | Nugget | Qwell[]: DAL1185 has filed their flight plan. |
| 17:14.54 | dennisharrison | eh? |
| 17:15.02 | Mercestes | I thought by "old box" you meant...it just lost power one day then came back online with this new error |
| 17:15.04 | dennisharrison | 1.4.3 |
| 17:15.06 | Nugget | http://flightaware.com/live/flight/DAL1185 |
| 17:15.15 | [TK]D-Fender | dennisharrison: Any chance you updated your kernel as well? That'd nuke the Zaptel .ko's |
| 17:15.20 | dennisharrison | yes I did |
| 17:15.21 | Mercestes | yea, my google search is riddled with zap 1.4 erors on that. |
| 17:15.23 | dennisharrison | I updated the kernel |
| 17:15.26 | dennisharrison | well ok |
| 17:15.30 | dennisharrison | the kernel was updated via yum |
| 17:15.32 | dennisharrison | by someone |
| 17:15.33 | [TK]D-Fender | dennisharrison: recompile & install Zaptel |
| 17:15.34 | dennisharrison | not sure who yet |
| 17:15.36 | dennisharrison | I did |
| 17:15.41 | dennisharrison | 1.4.3 |
| 17:15.42 | Mercestes | do a make clean |
| 17:15.44 | dennisharrison | yep |
| 17:15.47 | Mercestes | then a make && make install |
| 17:15.47 | dennisharrison | did that first |
| 17:15.54 | [TK]D-Fender | dennisharrison: re-modprobe zaptel & the drive for your card(s) |
| 17:15.59 | dennisharrison | even had to reset the spinlock typo for centos |
| 17:16.14 | dennisharrison | [TK]D-Fender, gotcha |
| 17:16.38 | rsd99 | anyone know a good site for ivr prompts? |
| 17:16.42 | [TK]D-Fender | dennisharrison: Then pastebin "cat /proc/interrupts", "ztcfg -vvvv" |
| 17:17.06 | [TK]D-Fender | rsd99: http://www.theivrvoice.com/ |
| 17:17.16 | Jingles | way I understand it, you'll want to be sitting in front of the asterisk box when you 'cat /proc/interrupts' |
| 17:17.23 | Jingles | and not be ssh'd in. |
| 17:17.28 | Dr-Linux | today all issues :( |
| 17:17.33 | [TK]D-Fender | Jingles: No difference |
| 17:17.40 | Dr-Linux | now cisco saying >> Protocol Application Invalid |
| 17:17.41 | [TK]D-Fender | Dr-Linux: www.drphil.com |
| 17:17.45 | EmleyMoor | galeras: And that will help me how? Does it give more information than what asterisk told it to dial? |
| 17:17.48 | *** join/#asterisk prashant_jois (n=prashant@mail.consolidated.ab.ca) |
| 17:17.57 | Mercestes | Jingles: Close. You want to be sitting at the asterisk box for "tcpdump -i eth0" :) |
| 17:18.32 | Mercestes | Dr-Linux, It sounds like the version in your config file does not point to the P0S file you have availabe in TFTP. or vice verse. |
| 17:18.45 | Mercestes | Dr-Linux, Or you are trying to rollback/forward to a version the system isn't liking |
| 17:19.04 | dennisharrison | [TK]D-Fender, http://pastebin.ca/579399 |
| 17:19.07 | prashant_jois | I recently upgraded to asterisk 1.4.5 and now I'm seeing the message "Internal RTCP NTP clock skew detected" come up every so often, what does this mean? |
| 17:19.08 | Mercestes | Dr-Linux, regardless, it's your P0S file... |
| 17:19.16 | EmleyMoor | Hmmm... it turns out it's MY fa machine that is for some reason answering |
| 17:19.18 | EmleyMoor | fax |
| 17:19.33 | Dr-Linux | Mercestes: well, i don't wanna do anything, this phone was just working fine from last 3 months, and now they reported me this issue |
| 17:19.39 | Dr-Linux | and i can't do anything |
| 17:19.41 | [TK]D-Fender | dennisharrison: show me the same w/ the modprobe calls, and add "dmesg" to it. |
| 17:19.43 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 17:19.55 | Dr-Linux | not even i can see what TFTP server it is requesting for |
| 17:20.04 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 17:20.05 | Dr-Linux | Mercestes: but i can ping this phone |
| 17:20.11 | Dr-Linux | from local LAN |
| 17:20.18 | rsd99 | i am just looking for some free ones. this is a project i am doing at home for my own experience |
| 17:20.19 | dennisharrison | my manual modprobes or the barfs or both? |
| 17:20.19 | [TK]D-Fender | EmleyMoor: Let me guess, got an FXS port of a TDM400P with the fax attached, right? |
| 17:20.39 | [TK]D-Fender | rsd99: Go record your own then. |
| 17:20.46 | Mercestes | Dr-Linux, That error doesn't mean the phone is offline, quite the contrary, it means, "I am online, I got my config server, but the firmware I am told I am supposed to have is different than the one that is available." |
| 17:20.54 | [TK]D-Fender | dennisharrison: BOTH :) |
| 17:21.07 | EmleyMoor | No, wrong. Fax is in parallel with FXO |
| 17:21.10 | [TK]D-Fender | dennisharrison: more > less ! |
| 17:21.20 | dennisharrison | lol |
| 17:21.21 | [TK]D-Fender | EmleyMoor: That blows. |
| 17:21.40 | Dr-Linux | Mercestes: my all cisco phone is working since long with same version 7.4 |
| 17:21.54 | Dr-Linux | Mercestes: btw, what's your suggestion |
| 17:21.55 | Dr-Linux | ? |
| 17:22.00 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 17:22.00 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:22.26 | Mercestes | Dr-Linux, Check the .cnf for that phone and make sure it ponits to the correct firmware |
| 17:22.34 | dennisharrison | modprobe didn't error this time |
| 17:22.42 | dennisharrison | http://pastebin.ca/579412 |
| 17:22.55 | Dr-Linux | Mercestes: how can i check? since it's not requesting to TFTP server |
| 17:23.14 | Mercestes | It is not? how are they configured? |
| 17:23.21 | Mercestes | Cisco pretty much demands a TFTP server. |
| 17:24.33 | Dr-Linux | Mercestes: yeah, but it's not requesting on my machine, but how i can i check on what IP this phone is requesting for TFTP? |
| 17:24.57 | [TK]D-Fender | dennisharrison: YUCK. |
| 17:25.08 | dennisharrison | yeah |
| 17:25.10 | [TK]D-Fender | dennisharrison: kill your zaptel source folder, and redo from scratch. |
| 17:25.17 | dennisharrison | [TK]D-Fender, ok |
| 17:25.24 | dennisharrison | should I go back to 1.2? |
| 17:25.27 | dennisharrison | where it was working? |
| 17:25.30 | dennisharrison | or should that not matter? |
| 17:25.31 | rsd99 | dr linux, under settings, select network configuration |
| 17:25.34 | Mercestes | Dr-Linux, Oh, somewhere around settings, and then network settings and you browse on down to TFTP something. TFTProuter and "manual TFTProuter" or something like that. |
| 17:26.04 | Dr-Linux | Mercestes: i know all that, but in this situation, i doesn't let me to press any button |
| 17:26.15 | dennisharrison | this trixbox box has been a bane to me |
| 17:26.25 | dennisharrison | nothing else gives me problems |
| 17:26.40 | Mercestes | dennisharrison, That would be my expectation of trixbox |
| 17:27.04 | dennisharrison | Mercestes, yeah ... customer got hyped about all the widgetness, and it was one of my first fxo setups using asterisk |
| 17:27.13 | dennisharrison | used to use avaya exclusively |
| 17:27.39 | dennisharrison | now I am using just straight asterisk on debian, but would rather not redo this box |
| 17:27.53 | Mercestes | follow fender's suggestion |
| 17:28.01 | Mercestes | but...I now undrestand why you have a cryptic error that doesn't show up often. |
| 17:28.10 | dennisharrison | lol |
| 17:28.13 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 17:28.46 | dennisharrison | http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.18.tar.gz sound good? |
| 17:28.55 | dennisharrison | or should I grab trunk or some other tag/branch? |
| 17:30.11 | [TK]D-Fender | dennisharrison: wait... you upgraded to 1.4 on trixbox?! |
| 17:30.22 | dennisharrison | yeah |
| 17:30.26 | dennisharrison | I haven't touched trixbox in .... |
| 17:30.30 | dennisharrison | almost a year |
| 17:30.39 | [TK]D-Fender | dennisharrison: FreePBX doesn't support 1.4, and god only knows what else will blow up... |
| 17:30.39 | dennisharrison | someone told me they used 1.4 now |
| 17:30.44 | dennisharrison | yeah |
| 17:30.45 | dennisharrison | nice :) |
| 17:30.51 | dennisharrison | going to 1.2 now |
| 17:30.52 | *** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca) |
| 17:30.54 | [TK]D-Fender | dennisharrison: Get your ass off that POS! |
| 17:30.58 | dennisharrison | I know :( |
| 17:31.02 | dennisharrison | I know I should just redo it |
| 17:31.19 | dennisharrison | just ... god I am busy man |
| 17:31.19 | [TK]D-Fender | dennisharrison: To know and not to do is not to know! |
| 17:31.23 | dennisharrison | hey have you seen openmoko.org |
| 17:32.02 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 17:32.02 | dennisharrison | we have been writing python bindings for all the functions and a few other apps for it |
| 17:32.06 | dennisharrison | should be releasing soon |
| 17:33.14 | dennisharrison | ok |
| 17:33.30 | dennisharrison | just compiled |
| 17:33.36 | dennisharrison | the only error I saw was |
| 17:33.36 | dennisharrison | /usr/src/zaptel/ztd-eth.c:189: warning: initialization from incompatible pointer type |
| 17:34.02 | dennisharrison | this is 1.2.18 |
| 17:35.28 | Mercestes | zap 1.2.18 or asterisk 1.2.18? |
| 17:35.41 | dennisharrison | zap |
| 17:36.35 | Mercestes | I wasn't aware there was a zap 1.2.18...hrm. |
| 17:37.03 | dennisharrison | lol |
| 17:37.09 | dennisharrison | well now the address for the error has changed |
| 17:37.11 | dennisharrison | from 25 to 6 |
| 17:39.48 | [TK]D-Fender | dennisharrison: Don't think of this as "tragic system failure', but rather "personal growth motivator and upgrade opportunity" ! :) |
| 17:40.14 | dennisharrison | [TK]D-Fender, lol ... dude, this blows my day ;p |
| 17:40.27 | dennisharrison | oh well eh? |
| 17:40.34 | [TK]D-Fender | dennisharrison: and you didn't even have to hit red-light district! |
| 17:40.49 | dennisharrison | [TK]D-Fender, although .... I might be looking for it tonight ;p |
| 17:42.17 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:42.18 | codey | is it possible to run 2 IAX2 sessions on one port? |
| 17:42.32 | codey | my asterisk acts like its not possible.. |
| 17:43.32 | bkruse | LOL |
| 17:43.34 | Corydon76-work | Sessions, yes. Servers, no. |
| 17:43.56 | bkruse | oh you mean asterisk and another iax service? or 2 asterisks |
| 17:44.01 | codey | so i can't trunk to my isdn-pbx and to my gsm-pbx? |
| 17:44.01 | bkruse | no :[ |
| 17:44.19 | codey | i've got pbx0, pbx1 and pbx2 |
| 17:44.24 | codey | pbx0 is the isdn-"relay" |
| 17:44.31 | codey | pbx1 connects to pbx0 via iax2 |
| 17:44.34 | codey | and to pbx2 via iax2 |
| 17:44.36 | codey | on the same port |
| 17:44.38 | bkruse | yes |
| 17:44.41 | bkruse | thats fine |
| 17:44.48 | codey | seems like its not |
| 17:44.50 | bkruse | you can make more than 1 iax call correct? |
| 17:44.52 | Corydon76-work | Well, you can, if that port is simply an asterisk service which "proxies" for the other two services |
| 17:45.04 | bkruse | So your saying you can call one, or call the other, not both? |
| 17:45.09 | bkruse | I find that hard to believe :[ |
| 17:45.19 | codey | i can call out over the isdn-pbx |
| 17:45.23 | codey | but i cant call the gsm-pbx |
| 17:45.43 | codey | so i would need a fourth box, that proxies the connections of pbx0 to pbx1 and pbx2? |
| 17:45.43 | bkruse | then its a problem between that box and gsm, not because your using two at the same time |
| 17:45.53 | Corydon76-work | We're talking 3 different machines on the same network? |
| 17:46.05 | codey | one box is on a dynamic ip |
| 17:46.11 | codey | the other 2 are on the same net |
| 17:46.26 | Corydon76-work | Then the box on the dynamic IP needs to register to the others |
| 17:46.32 | EmleyMoor | I've fixed the problem - certain numbers were waking up the fax machine |
| 17:47.14 | codey | the dynamic one rejects me with: No authority found |
| 17:47.25 | Corydon76-work | That's a different problem, then |
| 17:47.33 | Corydon76-work | That's authentication |
| 17:47.35 | EmleyMoor | I wish I could detect dring but that won't be possible here with 1.2 |
| 17:48.03 | bkruse | yep |
| 17:48.11 | codey | Corydon76-work: okay.. i'm trying Dial(IAX2/pbx2/1223) |
| 17:48.13 | bkruse | not because it cannot do 2 on the same port :P |
| 17:48.14 | codey | *123 even |
| 17:48.36 | [TK]D-Fender | EmleyMoor: You could get an of-the-shelf DRING line selector and put it in front. |
| 17:48.41 | Corydon76-work | codey: probably your secret isn't set up correctly |
| 17:48.54 | EmleyMoor | [TK]D-Fender: I could |
| 17:48.56 | Corydon76-work | or it doesn't match |
| 17:49.04 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:49.23 | EmleyMoor | What I do for the moment is use a fax machine that only responds to one of them and delay asterisk answering until after the fax machine has had a chance |
| 17:49.46 | codey | Corydon76-work: it is. i've copy&pasted it from the iax.conf on pbx2 |
| 17:50.02 | Corydon76-work | ~pb |
| 17:50.03 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 17:50.17 | Corydon76-work | Show me |
| 17:51.10 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 17:51.28 | magic_hat | anyone got experience installing/using * with Xubuntu? |
| 17:51.37 | rob0 | I have a piddly little system here: X101P to PSTN, Sipura SPA-2000 to analog phone. I'm hearing echo when calling out to PSTN. Any ideas what to check first? No fancy settings so far, defaults. |
| 17:52.01 | rob0 | (That is: using the Sipura to bridge to Zap.) |
| 17:52.39 | Corydon76-work | magic_hat: we strongly recommend that you do NOT run X on your Asterisk server |
| 17:53.05 | magic_hat | corydon76: why's that? |
| 17:53.17 | Corydon76-work | Because X eats interrupts |
| 17:53.29 | *** join/#asterisk jpeeler (n=thepeel@5-124.generic.clemson.edu) |
| 17:53.35 | magic_hat | ah. |
| 17:53.37 | magic_hat | crap! |
| 17:53.45 | bkruse | use X if you want to set it up |
| 17:53.50 | bkruse | but def not during usage or production :[ |
| 17:53.55 | bkruse | it doesnt play nice |
| 17:53.57 | Corydon76-work | It's fine for development machines, just not for production |
| 17:53.59 | *** join/#asterisk mihinomenest (i=pTBB@cerebus.clandestineresearch.com) |
| 17:54.05 | rob0 | Hmmm, maybe that's my problem too. |
| 17:54.17 | magic_hat | yeah, i'm looking for a production setup |
| 17:54.19 | Corydon76-work | Even on my development machine, I occasionally get jittery audio because I run X |
| 17:54.34 | bkruse | figures |
| 17:54.37 | rob0 | Because it seems okay when calling in on SIP to the Sipura-connected phone. |
| 17:55.15 | magic_hat | what's easy to setup, runs well with *? CentOS? |
| 17:55.17 | Corydon76-work | magic_hat: if you're accustomed to Xubuntu, I'd suggest using Ubuntu with the GUI |
| 17:55.31 | Corydon76-work | i.e. install Ubuntu as a "LAMP Server" |
| 17:55.50 | Corydon76-work | err, withOUT the gui |
| 17:56.07 | rob0 | Any GNU/Linux should be fine. The choice should be made according to what you as an admin need. |
| 17:56.31 | magic_hat | Corydon: I'm not accustomed to Xubuntu... just saw an opportunity to take an xubuntu box off someone's hands. |
| 17:56.41 | codey | Corydon76-work: http://slexy.org/paste/3172 |
| 17:57.01 | awk | well xubuntu has nothing to do with the console |
| 17:57.10 | awk | it doesn't use the same way to update packages, you have a stupid ui pakage manager so knowing apt isn't needed. |
| 17:57.13 | magic_hat | Basically what I need is something easy to set up that runs * well. I'm using it solely as as an asterisk server. |
| 17:57.44 | awk | magic_hat well i have a couple debian boxes a couple ubuntu boxes some fedora boxes some gentoo boxes and they all run lovely |
| 17:57.48 | awk | I believe there is alot of documentation for ubuntu though |
| 17:57.51 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 17:58.01 | awk | aswell as fedora |
| 17:58.05 | bkruse | ha, its like the biggest room in this server |
| 17:58.09 | Corydon76-work | codey: uh, which one is the dynamic server? |
| 17:58.22 | codey | pbx2 |
| 17:58.29 | the_5th_wheel | is there some way to use a modem as an incoming line as one can do with isdn? |
| 17:58.41 | Corydon76-work | codey: why register from pbx1 to pbx0, then? |
| 17:58.43 | codey | i actually had to set all hosts to host=dynamic because i wasn't able to register to pbx0 anymore |
| 17:58.48 | awk | the_5th_wheel no... |
| 17:58.57 | awk | an isdn modem with a hfc chipset yes |
| 17:59.02 | codey | Corydon76-work: pbx1 is the box that does all my sip-stuff |
| 17:59.11 | codey | Corydon76-work: routes calls to either gsm or isdn |
| 17:59.24 | Corydon76-work | codey: you should have 2 register statements, both on pbx2... pbx0 and pbx1 should know each other by static IP, not by registration |
| 17:59.26 | codey | Corydon76-work: so i actually need to hook up to pbx0 to do the isdn stuff. |
| 17:59.43 | codey | Corydon76-work: they did - til i added the pbx2 user |
| 17:59.51 | awk | the_5th_wheel but i wouldn't recomend using isdn modems, ive had some bad eperience with a couple boxes. some worked allright, but iyou get what you pay for. |
| 17:59.55 | Corydon76-work | codey: not according to what you have here |
| 18:02.01 | the_5th_wheel | what would the cheapest way be to interface a analouge line to a server then? just for experimenting and stuff? |
| 18:02.04 | codey | okay, just changed it ... it wasn't able to get called a few mins ago |
| 18:02.07 | codey | so i had to change it. |
| 18:02.15 | codey | how pbx0 is static on pbx1 |
| 18:02.17 | codey | *now |
| 18:02.52 | codey | but i'm still not able to call pbx2 |
| 18:03.25 | codey | omfg |
| 18:03.27 | codey | found the problem |
| 18:03.29 | Corydon76-work | codey: get rid of your friend definitions... spell each one out as user and peer |
| 18:03.40 | codey | the iax name wasn't the same on pbx2 |
| 18:03.43 | *** join/#asterisk austin_j (n=chatzill@austin-j.its.dist.maricopa.edu) |
| 18:03.49 | awk | the_5th_wheel you get analouge cards, but i wouldn't sugest them either, ive had problems with them too. |
| 18:04.02 | awk | the_5th_wheel so your experiemnce wouldn't be the best. |
| 18:04.09 | austin_j | Anyone use asteribank from xorcom? |
| 18:04.35 | EmleyMoor | This is going to sound really stupid - I can't get MusicOnHold to work |
| 18:05.55 | the_5th_wheel | awk, well, what would i then use? i cant get a digital line(as theanalouge line comes at next to nothing)(this is for a not for profit org) |
| 18:06.40 | awk | the_5th_wheel well you can get analouge cards just make sure you buy a decent card, do your homework.. you need something with good echo cancelation.. |
| 18:07.21 | awk | the_5th_wheel the last thing you want, is calls dropping, some calls not getting answered. etc etc |
| 18:07.44 | awk | i've had numerous problems, but once i started spending money all the problems went away :) |
| 18:08.00 | the_5th_wheel | awk: the problem is i cant spend much money |
| 18:08.14 | awk | the_5th_wheel what is your budget? |
| 18:11.17 | the_5th_wheel | everything i want to do is just to make my life easier. my budget is a couple of under rands(which translates into maybe a 100 US$) |
| 18:11.29 | *** join/#asterisk raining (n=raining@135.197.233.220.exetel.com.au) |
| 18:11.50 | codey | same problem again |
| 18:11.52 | codey | what the fuck |
| 18:12.02 | [TK]D-Fender | the_5th_wheel: how many lines? |
| 18:12.09 | raining | greetings all |
| 18:12.37 | the_5th_wheel | [TK]D-Fender: two. one one a premicell one pots |
| 18:12.51 | EmleyMoor | How do I see what formats are actually supported for native MoH? |
| 18:13.03 | awk | the_5th_wheel iesh, i'm not the right person to ask then. try ask somebody about cheap analouge cards |
| 18:13.16 | [TK]D-Fender | the_5th_wheel: .... |
| 18:13.23 | [TK]D-Fender | ~ygwypf |
| 18:13.24 | jbot | i heard ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 18:13.28 | raining | how can i set up my asterisk to accept direct calls to say sip:123@mydomain.com? |
| 18:13.49 | dennisharrison | [TK]D-Fender, alrighty man, thanks a bunch again :) I am headed out to go futz with it |
| 18:15.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:16.04 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:18.56 | raining | hello? |
| 18:22.31 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 18:23.06 | uwe | hello, any idea what can possibly make the music on hold play only once ? |
| 18:23.12 | *** join/#asterisk rogerius (n=mama@201.29.18.64) |
| 18:23.15 | rogerius | hello |
| 18:24.38 | raining | hi |
| 18:24.51 | raining | how do you alllow unauthenticated incoming calls via SIP? |
| 18:25.04 | rogerius | how do i configure asterisk to pass a call to openser? when i type the extension of a sip number, i want to call this sip registered @ openser, in the same machine |
| 18:25.08 | raining | basically i want anyone to be able to call sip:500@mydomain.com |
| 18:25.17 | rogerius | calling from pstn |
| 18:25.48 | raining | set up your dialplan? |
| 18:26.15 | awk | rogerius wouldn't you do it the other way around? |
| 18:26.45 | rogerius | awk what do you mean? |
| 18:27.22 | awk | well why would you wat asterisk to do the registration and not the ser? |
| 18:28.01 | awk | brb |
| 18:28.51 | EmleyMoor | I got native MoH working but it sounded awful! |
| 18:29.11 | codey | <PROTECTED> |
| 18:29.12 | codey | whee |
| 18:29.13 | codey | new error. |
| 18:29.38 | codey | and that extension is there for sure. |
| 18:29.38 | VJFROMGT | I am getting a SIP/2.0 401 Unauthorized but user/pass are correct |
| 18:29.53 | *** join/#asterisk fbffff (n=fbffff@dsl092-129-089.chi1.dsl.speakeasy.net) |
| 18:30.02 | VJFROMGT | extension is there for sure |
| 18:30.15 | VJFROMGT | if i config it with xlite it works,, with pap2 i get 401 |
| 18:30.16 | awk | back |
| 18:30.45 | awk | codey: and is the context correct? |
| 18:30.47 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 18:30.52 | VJFROMGT | pap2 reads Registration State:Can't connect to login server |
| 18:31.09 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:33.11 | holiday_42 | VJFROMGT: I have pap2v2, had to turn up the verbosity in the asterisk CLI... I found with my pap2 the secret was munged... had to change it to some other value, save settings, change it back, save again. |
| 18:33.47 | VJFROMGT | what sort of value works for u? |
| 18:33.54 | holiday_42 | VJFROMGT: oh, sound like it's not getting to the * box at all in your cae. |
| 18:34.13 | VJFROMGT | but * is giving a 401 as soon as i turn on pap |
| 18:35.05 | codey | awk: yes |
| 18:37.43 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 18:38.07 | [TK]D-Fender | codey: "No such context/extension" <- this is NOT lying. your configs are wrong keep reading until you find it or your eyes bleed. |
| 18:38.57 | holiday_42 | VJFROMGT, i'll check mine.. sec. |
| 18:39.06 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 18:39.16 | codey | tsurko: it IS there |
| 18:39.17 | codey | pbx0:/etc/asterisk# grep $(cat iax.conf | grep context | sed -e 's,^context=,,') extensions.conf |
| 18:39.20 | codey | [fsincoming] |
| 18:39.38 | *** join/#asterisk GaryH (n=GaryH@wallace.garysoft.co.uk) |
| 18:39.42 | codey | xten => _XX,1,Answer() |
| 18:39.42 | codey | exten => _XX,2,Dial(SIP/${EXTEN},60) |
| 18:39.42 | codey | exten => _XX,3,Congestion |
| 18:39.45 | codey | +e |
| 18:40.09 | VJFROMGT | if i want to see pasword, what should i set verbose to? |
| 18:40.32 | holiday_42 | I dunno... i just used 9 nines |
| 18:40.40 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 18:41.29 | codey | [TK]D-Fender: i don't even see a call coming in on the other pbx |
| 18:42.30 | [TK]D-Fender | codey: pastebin the CLI output of a call that fails with SIP debug & verbose 10. And include the full configs related to that server (sip & extensions) |
| 18:42.44 | *** join/#asterisk marcan (i=1337@117.Red-88-5-77.staticIP.rima-tde.net) |
| 18:43.02 | codey | [TK]D-Fender: it's not sip |
| 18:43.10 | [TK]D-Fender | codey: IAX, whatever |
| 18:43.22 | [TK]D-Fender | codey: Everything related to the call attempt |
| 18:43.28 | [TK]D-Fender | \~pb |
| 18:43.30 | [TK]D-Fender | ~pb |
| 18:43.31 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 18:43.32 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 18:45.14 | Alan_Hicks | Would anyone recommend the O'Reilly book "T1: A Survival Guide" to some one interested in learning more (a lot more) about T1s, or is there another book you prefer for that? |
| 18:45.33 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 18:45.39 | Jingles | most anything from ORA is worth owning as reference material. |
| 18:46.46 | Alan_Hicks | Yeah, I love their books, but was wondering if there might be something better out there that I don't know about. |
| 18:47.56 | [TK]D-Fender | Alan_Hicks: may be more of a DATA book than a VOICE book.... |
| 18:48.29 | holiday_42 | VJFROMGT, i leave the SIP stuff alone, just change line 1 (and line 2 if you want) just specify username/password/proxy server & registrar address |
| 18:48.44 | Alan_Hicks | The more I read online, the more I realize how much I don't know, and the more my curiosity grows. |
| 18:48.55 | *** join/#asterisk perf3kt (i=perf3kt@iupui-vpn-32-59.noc.iupui.edu) |
| 18:50.40 | [TK]D-Fender | Alan_Hicks: A lot of this stuff you only really learn when you start getting your hands dirty, and *'s side of T1 is a lot simpler than the DATA side |
| 18:59.34 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 19:02.31 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 19:03.21 | *** join/#asterisk shtoom (n=godson@59.93.115.56) |
| 19:04.07 | ramindia_ | any one recomend me tools to diagnosis voice break problems in network ? |
| 19:11.54 | *** join/#asterisk enmaca (n=enmaca@189.157.117.149) |
| 19:13.14 | *** join/#asterisk mrdigital (n=mrdigita@207-172-229-100.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 19:13.43 | mrdigital | whats the exten => command to dial ring group 600 |
| 19:14.00 | *** join/#asterisk linuxsouth (n=chatzill@72.242.225.99) |
| 19:15.46 | *** part/#asterisk Alan_Hicks (n=alan@208.62.162.112) |
| 19:17.29 | *** join/#asterisk sysreq (n=sysreq@210.145-ppp.3menatwork.com) |
| 19:17.44 | [TK]D-Fender | mrdigital: "show application dial" and there is no inherent thing called "group 600". |
| 19:18.14 | ramindia_ | [TK]D-Fender: any suggestion on diagnosis voice breaks |
| 19:19.23 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 19:19.58 | [TK]D-Fender | ramindia_: check your bandwith , latency & jitter |
| 19:21.14 | *** part/#asterisk linuxsouth (n=chatzill@72.242.225.99) |
| 19:22.09 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 19:22.09 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:23.12 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:23.36 | *** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com) |
| 19:30.37 | ramindia_ | [TK]D-Fender: i dont see any problem in terms of bandwidth ,latency |
| 19:30.45 | ramindia_ | how can i check the jitter any tools |
| 19:32.32 | *** join/#asterisk linsouth (n=allen@72.242.225.99) |
| 19:32.44 | *** part/#asterisk shtoom (n=godson@59.93.115.56) |
| 19:33.55 | [TK]D-Fender | ramindia_: Don't know. |
| 19:34.30 | _DAW | ramindia_: ethereal does some basic stuff. Get me a tcpdump and I'll run it through wineyeq for you. |
| 19:37.07 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 19:42.21 | errr | when using the directory from an ivr, where does it look to see if it can find someone to match whats being searched for? |
| 19:45.00 | russellb | Does anyone else get *really* annoyed with the coiled cables attached to phone handsets? |
| 19:45.09 | rob0 | :) |
| 19:45.10 | [TK]D-Fender | errr: "show application directory" |
| 19:45.10 | russellb | errr: voicemail.comf |
| 19:45.27 | errr | great guys thanks |
| 19:46.32 | errr | is there anyway I can make it look on some other server? |
| 19:46.45 | errr | we have a central voicemail server |
| 19:47.02 | rob0 | I used to work in a multi-floor office building, and we'd take the handset cords to the stairwell to dangle / untangle them. |
| 19:47.03 | russellb | you can store voicemail config in a database that is available across servers ... |
| 19:47.04 | [TK]D-Fender | errr: Sure, mount its volume into the voicemail folder. |
| 19:47.15 | russellb | rob0: nice :) |
| 19:48.15 | [TK]D-Fender | rob0: Shows your users were shcmucks who'd allow long term mangling to occur. I had the same phoone with my Norstar 8x24 for at least 7 years and it was IMPECCABLE |
| 19:49.21 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 19:49.46 | rob0 | That, and low-bid (state gov't) equipment :) |
| 19:52.30 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:56.42 | errr | hey I have some patton sn4112's has anyone ever set any of these up? |
| 20:05.15 | *** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.mn.comcast.net) |
| 20:08.33 | *** part/#asterisk dcm_ (n=dcm@207.59.3.77) |
| 20:09.05 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 20:13.57 | *** join/#asterisk rsd99 (n=rsd095@h-67-103-23-130.phlapafg.covad.net) |
| 20:14.55 | *** join/#asterisk sponger (n=sponger@66-193-153-10.static.twtelecom.net) |
| 20:15.56 | brea | What will happen if all of my identifiers don't get cleared? |
| 20:16.04 | brea | Will it take up memory? |
| 20:18.54 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:18.58 | philippel | ran into an interesting challenge I'm wondering if I'm missing anything a way to excape it, the following line of dialplan (which I auto-generate): |
| 20:19.01 | philippel | exten => 220,n,Set(__ALERT_INFO=${IF($["x${ALERT_INFO}"="x"]?http://127.0.0.1/Bellcore-dr3:${ALERT_INFO}}) |
| 20:19.43 | philippel | the : in http: is interpreted as the end of the 'true' result |
| 20:20.36 | philippel | I've tried excaping it with \, putting the whole thing in quotes and even setting a variable one statement ahead and using the variable instead of the string in the clause, all with the same results - any thoughts, different way of escaping it? |
| 20:21.24 | philippel | (vs. a different way of writing the dialplan which I can otherwise do but this seems to have interesting consequences for the general use of the IF() function |
| 20:21.54 | sponger | that has always seemed to me one of the sucky things about the asterisk dialplan |
| 20:22.07 | Mercestes | anyone installed asterisk on Debian? |
| 20:22.18 | Mercestes | s/debian/ubuntu/ |
| 20:22.32 | sponger | philippel check out adhearsion (am i allowed to say that in here?_ |
| 20:22.45 | sponger | its lets you write the dialplan in ruby |
| 20:22.57 | philippel | sponger I'm not interested in doing that |
| 20:23.22 | rsd99 | i just got asterisk up and running last night. and wow......it'll take me a while to learn |
| 20:23.27 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 20:24.00 | philippel | I'm interested to know if there is a way in the IF clause to handle a : in the results escaped some how or if I have to change all of my dialplan generation for such cases |
| 20:24.08 | Mercestes | [TK]D-Fender, what do you install * on? |
| 20:24.32 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 20:24.52 | [TK]D-Fender | Mercestes: Generally on Inten X86 based computers |
| 20:24.57 | [TK]D-Fender | Intel* |
| 20:25.19 | Mercestes | [TK]D-Fender, Sorry, what distro? |
| 20:25.19 | [TK]D-Fender | Mercestes: Usually CentOS, if not, Slackware |
| 20:25.56 | Mercestes | oh... |
| 20:26.51 | sponger | has anyone in here got ztdummy running on a VE using openvz? |
| 20:27.47 | tzafrir_laptop | Mercestes, I got it installed on Ubuntu (not much more than that) |
| 20:28.30 | Mercestes | tzafrir_laptop, Yea, I get the compile errors on it, I founda howto that might (should) work. |
| 20:28.43 | tzafrir_laptop | What compile errors? |
| 20:29.20 | tzafrir_laptop | packages names generally rather similar to debian |
| 20:29.52 | pifiu | hey fender |
| 20:29.53 | pifiu | wasup |
| 20:30.51 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 20:30.54 | diclophis-work | hello all |
| 20:31.05 | diclophis-work | how would i get all manager events/actions logged in the console? |
| 20:31.20 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 20:31.21 | [TK]D-Fender | diclophis-work: Start coding |
| 20:31.22 | diclophis-work | i thought there was a manager.conf setting for it, but i cant seem to find it |
| 20:31.29 | [TK]D-Fender | pifiu: y0 |
| 20:32.02 | kombi | wold someone know the command line argument to start wireshark and listen on eth0? The manpage is somewhat hefty there.. |
| 20:32.29 | kombi | of the top of his/her head I mean.. |
| 20:32.45 | diclophis-work | [TK]D-Fender: are you saying that the manager interface doesnt have any facilities to log to the console? |
| 20:32.47 | *** join/#asterisk robin_z (n=robin@rapid2.gotadsl.co.uk) |
| 20:32.50 | kombi | s/wold/would/ |
| 20:32.56 | robin_z | meep? |
| 20:32.56 | [TK]D-Fender | diclophis-work: Nothing I'm aware of. |
| 20:33.02 | diclophis-work | arg |
| 20:34.02 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-366565.home.otenet.gr) |
| 20:34.23 | diclophis-work | could have sworn i saw a setting for it |
| 20:34.26 | kombi | or is there a command line packet sniffer somewhere? |
| 20:34.32 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:34.51 | diclophis-work | kombi: isnt that what wireshark is? |
| 20:34.52 | [TK]D-Fender | kombi: tetherial |
| 20:34.59 | diclophis-work | the only other thing i know of is etheral |
| 20:35.03 | [TK]D-Fender | kombi: I think its called tshark now |
| 20:36.02 | kombi | thanks! |
| 20:37.26 | *** part/#asterisk falz (n=falz@proxy.supranet.net) |
| 20:37.31 | mocker | Are toll free numbers pointed directly to a PRI or are they pointed to a DID at the LEC? |
| 20:37.46 | mocker | I always though it was done at the LEc. |
| 20:37.50 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 20:38.21 | Jingles | tethereal is the command line sniffer, wireshark is the gui version. |
| 20:38.25 | Jingles | good times. I use it often. |
| 20:38.44 | anonymouz666 | good times bad times you know I had my share |
| 20:40.35 | *** part/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 20:40.44 | *** join/#asterisk joshaidan (n=brianj@thunderbay-voip-4.vianet.ca) |
| 20:41.59 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 20:43.03 | joshaidan | Does asterisk give you the option to erase your temporary greeting? |
| 20:43.20 | Qwell[] | joshaidan: same way you record it |
| 20:43.42 | [hC] | you know, early versions did not give you a choice |
| 20:43.45 | [hC] | you had to remove it from the fs |
| 20:44.11 | joshaidan | ah... thanks :) |
| 20:44.14 | [hC] | chances are if you are running anything above 1.2.5(?) you just go in to the same place you did to record it |
| 20:44.17 | [hC] | and theres an option to delete it |
| 20:47.27 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 20:47.46 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 20:48.19 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 20:52.34 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 20:54.41 | mrdigital | hey voicemeup? |
| 20:57.28 | perf3kt | I get a 407 proxy authenication required in my debug when trygin to recieve a call in |
| 20:57.38 | *** join/#asterisk yangvnc (i=yang@84-16-238-88.internetserviceteam.com) |
| 20:57.38 | *** part/#asterisk rsd99 (n=rsd095@h-67-103-23-130.phlapafg.covad.net) |
| 20:57.39 | kombi | people don't hear me when they call, everything else is fine. Where must I investigate? |
| 20:57.48 | yangvnc | Hello, I am looking for VOIP provider which can give FREE (european) numbers accessible over PSTN, can anyone suggest me anything? |
| 21:00.57 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-366565.home.otenet.gr) |
| 21:01.07 | nasls_lsa | kl-anill ?!?!? |
| 21:01.10 | *** part/#asterisk nasls_lsa (n=chatzill@athedsl-366565.home.otenet.gr) |
| 21:01.11 | kombi | Jingles: would you know a nice way to invoke it, debugging by one way audio? |
| 21:01.11 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 21:02.10 | *** join/#asterisk mitcheloc (n=mitchelo@001-794-703.area1.spcsdns.net) |
| 21:04.26 | kombi | which ports are used for RTP? |
| 21:04.54 | austin_j | Anyone here have experience with McCloud USA for Voice/Data connectivity? |
| 21:05.11 | Jingles | kombi: invoking the command line packet sniffer? |
| 21:05.26 | austin_j | Mercestes: Asterisk works on Ubuntu. My preference is CentOS though. |
| 21:05.28 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 21:06.11 | Mercestes | austin_j, Your awesome |
| 21:06.21 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 21:07.00 | kombi | Jingles: just did like tshark -i eth0 and noticed way more RTP packets with two way audio.. |
| 21:07.18 | kombi | firewall is open for 10000:20000 though.. |
| 21:07.32 | Jingles | right - when you're engaged in a 'phone conversation', there's loads of RTP traffic - carries the voice payload. |
| 21:07.56 | austin_j | Anyone plopped freepbx on top of an existing asterisk installation before? |
| 21:08.37 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 21:08.46 | kombi | Jingles: my inbound calls have no incoming audio and - as I now found out - hardly any RTP packets, I wonder why since iptables is wide open |
| 21:09.05 | tzafrir | apt-get install freepbx ... |
| 21:09.36 | bkruse | apt-get remove self |
| 21:09.39 | austin_j | :) |
| 21:10.08 | kombi | Jingles: I wonder where I should investigate next |
| 21:10.18 | austin_j | I'm just wondering what'll happen when freepbx encounters the hand written sip.conf/etc |
| 21:11.09 | austin_j | I've only ever installed freepbx on a system with nothing on it; that's why I ask |
| 21:11.25 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.183) |
| 21:11.56 | tzafrir | austin_j, with sip.conf the requirements are not that strict. However you're taking your chances regarding when an upgrade of freepbx will run over your sip.conf |
| 21:12.01 | kombi | on a box with its own static ip, absolutely no need for nat right? |
| 21:12.16 | austin_j | tzafrir: I suspected that ;) |
| 21:12.36 | austin_j | kombi: maybe for the phones connecting to it |
| 21:13.35 | kombi | austin_j: I meant in [general] where the sip provider is registered, no nat=yes necessary |
| 21:14.39 | austin_j | kombi: if both ends are not natting then nat=yes is not needed. It doesn't hurt as far as I can tell.. I always set nat=yes. |
| 21:14.40 | dijungal | anyone here with experience in getting Cisco (H.323) to pass calls into Astrerisk 1.4.5 |
| 21:14.42 | dijungal | ? |
| 21:15.28 | austin_j | kombi: nat=yes tells asterisk to put extra stuff in the headers |
| 21:17.13 | mrdigital | why am i getting # not in service? |
| 21:17.17 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 21:18.19 | kombi | Jingles: in tshark, how do you filter udp? |
| 21:18.20 | austin_j | mrdigital: misconfigured? |
| 21:18.48 | perf3kt | tz: you knwo of the 407 authenication error, that would prevent incoming calls |
| 21:18.53 | holiday_42 | it could be true (for example the sip user not registered) or misconfiured dialplan |
| 21:19.40 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 21:20.15 | tzafrir | perf3kt, there are quite a few potential reasons |
| 21:20.22 | holiday_42 | peek at asterisk cli, with verbose should give a hint |
| 21:20.32 | tzafrir | incorrect password? |
| 21:21.02 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 21:21.05 | tzafrir | you should see users and passwords with 'sip show peers' (sorry, I know you're not a newb) |
| 21:22.51 | Jingles | kombi: not udp (in the filter line) |
| 21:23.34 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 21:23.34 | kombi | thanks! |
| 21:24.34 | kombi | (what do you mean by "in the filter line" though?, can you give an example? |
| 21:25.02 | perf3kt | holiday: were you talking to me |
| 21:25.21 | kombi | tshark -i eth0 -f udp looks good |
| 21:25.23 | Jingles | there's a spot in the top toolbar area of wireshark. |
| 21:25.31 | Jingles | oh - not using the gui? |
| 21:25.38 | kombi | oh, no gui here (never had one ever..;) |
| 21:25.46 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 21:25.56 | kombi | at least not on a nix |
| 21:26.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:26.29 | Jingles | well, you can do just a flat capture (tethereal -i eth1 -w capturefile) |
| 21:26.42 | Jingles | then open the capturefile in wireshark on a machine with a gui. |
| 21:27.04 | Jingles | I can't stand fiddling with packet caps without a gui tool. |
| 21:27.26 | kombi | they nicely run by you there..;) |
| 21:27.54 | kombi | I can see loads of RTP zooming by on the ok connection and way less on the one with one way audio |
| 21:28.06 | kombi | just what does that tell me? |
| 21:28.15 | Jingles | ok. so, usually that means there's a firewall blocking your rtp traffic. |
| 21:28.36 | bkruse | or being filtered, and slowly. |
| 21:28.50 | kombi | $IPTABLES -A FORWARD $UDP -d $IP_afm/32 --destination-port 10000:20000 -j ACCEPT though.. |
| 21:29.20 | kombi | slowly.. hmm |
| 21:30.54 | kombi | nay, couldn't, the working line shows hundreds of packets coming in through it, it's the inbound call that does not let audio travel outside.. |
| 21:31.08 | kombi | weired.. |
| 21:33.42 | bkruse | i could see with maybe hundreds of calls, but not 1. |
| 21:35.47 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 21:36.20 | *** part/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
| 21:36.26 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 21:37.30 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 21:37.32 | kombi | kbruse: I wonder where all those RTP packets go.. |
| 21:40.02 | *** join/#asterisk rantsh (n=chatzill@201.210.16.238) |
| 21:40.19 | rantsh | Hi guys |
| 21:40.43 | kombi | hi rantsh |
| 21:41.03 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 21:41.04 | rantsh | I came back just to tell you you guys rule! :) I got to record calls trouble-free with * 1.4.5 |
| 21:42.31 | rantsh | and I believe I understand * better than ever before (remember, I'm a complete n00b hehehe) |
| 21:43.37 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 21:45.19 | Qwell[] | rantsh: that's always a good thing |
| 21:46.08 | rantsh | yup |
| 21:46.14 | *** part/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 21:46.34 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:54.44 | rantsh | anyway, on my new assigment now (actually it was my first assigment but my boss was mercifull and let me study with call recording) |
| 21:55.32 | rantsh | I need to setup codec translations, I've been searching all over voip-info.org, but haven't got any useful information on where to start looking |
| 21:56.10 | rantsh | anyone knows where I can start at least searching or reading on this subject? |
| 21:56.28 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 21:57.07 | *** join/#asterisk dijungal (n=chatzill@209.59.110.5) |
| 21:57.40 | dijungal | anyone with Cisco <-(H.323)-> Asterisk experience? |
| 21:58.16 | dijungal | i'm trying to send voip calls from a cisco router to asterisk using ther H.323 protocol |
| 21:58.25 | [TK]D-Fender | rantsh, * translates automatically if both sides of a bridged call can't agree on a common codec. |
| 21:58.35 | dijungal | i see the router actually sending packets but asterisk does not pickup |
| 21:59.21 | [TK]D-Fender | dijungal, Have you set up an H.323 entry for it to use & a dialplan context to land in for processing? |
| 22:00.05 | dijungal | i've setup the h.323 entry in h323.conf (i'm using chan_h323) |
| 22:00.17 | dijungal | i have not setup any dial plans as yet |
| 22:00.41 | dijungal | i am expecting asterisk to answer the call and then drop it.. cause it does not know how to route it |
| 22:01.10 | dijungal | but it does not even acknowledge the call |
| 22:01.35 | [TK]D-Fender | dijungal, if * doesn't have an appropriate dialplan to match, you won't see ANYTHING unless you're in chennel debug mode. |
| 22:01.50 | [TK]D-Fender | dijungal, Even picking up and hanging up is all DIALPLAN. |
| 22:01.58 | dijungal | how do i go into channel debugging mode ? |
| 22:02.29 | [TK]D-Fender | dijungal, First just set the context in your H323 entry and make a catch-all exten for it to land on. |
| 22:02.37 | dijungal | ok... i will create a simple dialplan and see if it works.. Answer, wait(2), then hangup :) |
| 22:02.48 | [TK]D-Fender | dijungal, that should tell you if your auth is correct without too much work. |
| 22:02.56 | dijungal | ok |
| 22:02.57 | dijungal | will do |
| 22:03.15 | [TK]D-Fender | dijungal, for this test, I'd suggest "_X." but only for the test period. |
| 22:03.46 | dijungal | irie |
| 22:04.22 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:07.39 | rantsh | I know it translates codecs automatifcally,BUT for some reason (I don't really know or understand well enough) My company needs to translate calls from g729 to g711 on certain calls |
| 22:10.40 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:14.05 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) |
| 22:15.38 | [TK]D-Fender | rantsh, it will only happen as needed |
| 22:16.19 | rantsh | so there's no way I can do this translation at will? |
| 22:19.30 | [TK]D-Fender | rantsh, you'd have to have each side disagree on codec. |
| 22:20.36 | [TK]D-Fender | rantsh, if you say that one side can only speak G729 and the other G711 then all calls from their respective sides will be in that codec. if they end up being bridged together , * will automatically translate them |
| 22:21.53 | rantsh | (clueless) |
| 22:22.59 | *** join/#asterisk Fulk (n=jon@cpc3-stap7-0-0-cust848.nott.cable.ntl.com) |
| 22:26.30 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 22:26.32 | mxmasster | hi all |
| 22:27.11 | mxmasster | on a g.711 codec average bandwidth used is ~80 kbps |
| 22:27.23 | mxmasster | so if both call legs are on your system that's ~160 kbps |
| 22:27.25 | lesouvage | Does anybody has good experience with integrating Gtalk in an Asterisk box, does it worth a try |
| 22:27.50 | mxmasster | so 1000 calls would be 160 Mbit of sustained data traffic? |
| 22:28.09 | mxmasster | is this an accurate number? |
| 22:30.25 | [TK]D-Fender | mxmasster, there are only 3 kinds of people..... those that get mat, and those that don't. |
| 22:30.28 | [TK]D-Fender | math* |
| 22:32.21 | Mercestes | [TK]D-Fender, You spend most of your life in IRC....and you can't type..... |
| 22:32.24 | Mercestes | *tsk tsk |
| 22:32.39 | kombi | jeez, i've tried everything now, just cant find why there is one way audio on inbound calls.. |
| 22:33.41 | kombi | is it certain that you can't cause this behaviour just my messed up configs? |
| 22:34.01 | Jingles | best way to test that theory - put both phones on the same network segment. |
| 22:34.33 | kombi | good one Jingles! |
| 22:34.38 | bkw__ | mxmasster, you can get 1200 calls on a 100mbit @ 40ms on ulaw |
| 22:34.42 | bkw__ | but you don't have much wiggle room |
| 22:34.58 | *** join/#asterisk ToyMan (n=Stuart@cpe-68-175-3-247.hvc.res.rr.com) |
| 22:34.58 | kombi | let me find another soft phone.. |
| 22:35.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 22:38.00 | mxmasster | how does that math work |
| 22:38.02 | mxmasster | ? |
| 22:39.10 | Fulk | ping |
| 22:39.26 | Fulk | I'm using a digium TDM card and I have to configure my * box to reboot daily to stop it locking up |
| 22:39.29 | JT | mxmasster: it works fine |
| 22:39.30 | Fulk | some kind of memory leak |
| 22:39.33 | JT | mxmasster: how does yours? |
| 22:39.38 | mxmasster | g.711 with 40ms and rtcp and 1200 channels shows 90.72 Mbit per second |
| 22:39.43 | Fulk | anyone else experienced that? |
| 22:39.49 | JT | mxmasster: you don't add up both directions |
| 22:40.24 | mxmasster | JT i'm not talking about both directions between UA and Server - i am talking about UA - Server - Gateway |
| 22:40.33 | mxmasster | leg 1 UA - server |
| 22:40.38 | mxmasster | leg 2 Server - Gateway |
| 22:40.40 | JT | <PROTECTED> |
| 22:40.49 | *** join/#asterisk ber111 (i=brad@neu.cow.org) |
| 22:40.52 | mxmasster | right that was my question |
| 22:41.02 | JT | and you were adding up both directions |
| 22:41.05 | JT | you don't do that... |
| 22:41.06 | tzafrir_laptop | Fulk, memory leak? why won't you sample it? |
| 22:41.16 | ber111 | if you execute a script via DeadAGI, how do you check if hte hcannel is hung up in that script so you can clean it up |
| 22:41.40 | mxmasster | JT you just agreed with me, and then you just told me that it is not the case |
| 22:41.48 | Fulk | tzafrir, munin shows the memory usage grow and grow throughout the week |
| 22:41.50 | tzafrir_laptop | add an hourly cron of ps aux or whatever |
| 22:42.08 | JT | mxmasster: i disagreed both times, you do NOT add both directions in bandwidth calculations |
| 22:42.13 | JT | they're seperate calculations |
| 22:42.18 | tzafrir_laptop | Fulk, usage of what? |
| 22:42.26 | rantsh | btw, going back to call recording, I just noticed that it doesn't work on uLaw calls, any clue on why, it works perfectly with the rest of my codecs |
| 22:42.28 | Fulk | lemme check |
| 22:42.30 | tzafrir_laptop | the total system memory? swap space? |
| 22:42.49 | mxmasster | JT okay so how do I properly make the calculation then |
| 22:42.55 | Fulk | been a while, the cron job "solved" the problem |
| 22:42.57 | mxmasster | If I have one user make a g.711 call to a vendor |
| 22:43.02 | mxmasster | through our server |
| 22:43.07 | Fulk | but I thought I'd take a punt in here - can't stand things not being "perfect" |
| 22:43.14 | JT | that's approximately 85kbit/s |
| 22:43.20 | mxmasster | what is my total bandwidth usage |
| 22:43.38 | JT | 85kbit/s each way, maybe a fraction under |
| 22:43.48 | JT | using sip |
| 22:44.23 | *** join/#asterisk Strom_M (n=strom@60-241-200-26.static.tpgi.com.au) |
| 22:47.43 | Fulk | hmm, appears to be cache - that would preclude an application leak |
| 22:48.04 | Fulk | perhaps the kernel doesn't play well with the PowerEdge server |
| 23:02.11 | *** join/#asterisk CrashHD (n=timf@70.96.98.65) |
| 23:02.26 | CrashHD | hello |
| 23:03.09 | CrashHD | how can I capture the sip response code once returned? |
| 23:03.10 | jwh | Hi |
| 23:03.24 | *** join/#asterisk SwK_ (n=SwK@user-69-73-37-99.knology.net) |
| 23:03.25 | CrashHD | I have a server sending a 603 when I've hit my call limit |
| 23:03.44 | CrashHD | which should then roll to the next carrier |
| 23:04.00 | CrashHD | but because I have busy handling setup s-BUSY sends busy() |
| 23:04.42 | CrashHD | sip_header() looked good but I saw in the sip debug that the 603 code wasn't followin gany of the fields but rather was at the top of the message |
| 23:05.00 | CrashHD | any thoughts? |
| 23:05.53 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 23:07.14 | jwh | hmm, not sure tbh |
| 23:07.32 | Jingles | don't you use 'IsChanAvail' to roll over? |
| 23:08.08 | CrashHD | I'm dialing multiple providers |
| 23:08.12 | CrashHD | not end devices |
| 23:08.18 | _DAW | a |
| 23:08.26 | CrashHD | well trying at least |
| 23:09.00 | *** join/#asterisk perf3ktion (n=perf3kt@adsl-68-73-150-167.dsl.ipltin.ameritech.net) |
| 23:10.10 | *** join/#asterisk xpander4 (n=gaston@adsl-074-169-108-059.sip.bct.bellsouth.net) |
| 23:12.14 | *** join/#asterisk bcnl (n=mike@S010600131078957c.vc.shawcable.net) |
| 23:12.14 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 23:12.38 | Cybertoy | hi .. I'm trying to register a new device but get "Device does not match ACL" error... anyone know what that means? |
| 23:12.46 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 23:12.51 | Cybertoy | I see that errror on the asterisk CLI ... |
| 23:15.08 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 23:15.32 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
| 23:16.41 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:16.49 | CoffeeIV | I have a voicetronix openpci4 card, and I am trying to use it with asterisk 1.2. Voicetronix has drivers version 2.*, 3.*, and 4.0 -- is there a particular version of the voicetronix drivers that works with asterisk 1.2, or should they all work ? |
| 23:18.29 | bcnl | has anyone here used ParkAndAnnounce |
| 23:18.31 | bcnl | much? |
| 23:18.52 | bcnl | I would like to dial multiple extensions to notify, it seems to crap out at the & |
| 23:21.28 | CrashHD | so nobody knows how I can parse the sip return code? |
| 23:21.35 | CrashHD | and use it in my dial plan? |
| 23:22.01 | snuffy22 | there isn't a way i know of after dial to get the exact code no |
| 23:22.22 | snuffy22 | generally if its a 500 error.. its 'congestion' returned in ${DIALSTATUS} |
| 23:22.51 | CrashHD | well 603 is being returned |
| 23:23.03 | CrashHD | but I don't want it to think busy signal |
| 23:23.08 | *** join/#asterisk forsaken_ (i=forsaken@201.64.24.247) |
| 23:23.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:24.09 | snuffy22 | well only thing i can think of is ask in asterisk-dev or on the bugs.digium page about it see if you get shut down so to speak |
| 23:25.14 | JerJer | moo |
| 23:25.26 | _VoiceMeUp_COM | jerjer |
| 23:25.29 | _VoiceMeUp_COM | can i ask ou something ? |
| 23:25.34 | JerJer | can you? |
| 23:26.11 | _VoiceMeUp_COM | $tU |
| 23:26.23 | _VoiceMeUp_COM | in ser.. i need to regex if starts with "6666" then strip first 4 |
| 23:26.28 | _VoiceMeUp_COM | you good on that ? |
| 23:27.59 | snuffy22 | mm.. i'd thought that wouldn't be too hard but i've got noidea on ser regex |
| 23:38.51 | dijungal | TKD-Fender: ok so it's still not working after i've add the dialplan |
| 23:39.01 | dijungal | asterisk just not picking up the call from the cisco |
| 23:39.51 | [TK]D-Fender | dijungal, try "h323 debug" and see if you cn see the call coming in. also pastebin your h323 & extensions configs |
| 23:39.52 | [TK]D-Fender | ~pb |
| 23:39.52 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 23:52.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:52.57 | dijungal | TKD-Fender: http://pastebin.ca/580122 |
| 23:58.19 | [TK]D-Fender | extension.conf ---------------- ? |
| 23:58.42 | dijungal | oooh yeh that's the extensions.conf config |
| 23:58.45 | [TK]D-Fender | just a typo on PASTING I hope, and I'm missing the CLI output & debug... |
| 23:58.59 | dijungal | i got nothing on the CLI |
| 23:59.17 | dijungal | i'm telling u, asterisk does not budge |
| 23:59.27 | [TK]D-Fender | dijungal, go enable debug on H323. |
| 23:59.34 | dijungal | did that |
| 23:59.36 | dijungal | same thing |
| 23:59.55 | [TK]D-Fender | maybe your Cisco in improperly configured then |
| 23:59.56 | dijungal | i did "h.232 debug" |