00:00.06 | mog_home | yes |
00:00.08 | mog_home | <PROTECTED> |
00:00.24 | fugitivo | DrJES: did you setup zapata.conf correctly? |
00:00.58 | DrJES | I believe so, as I have a channel => X for the new card channels, but still no go. |
00:01.15 | Rawplayer | dudes: exten => 101,2,Dial(SIP/Kevin1) |
00:01.17 | DrJES | Channel => 1,2,3,4 all work |
00:01.17 | Rawplayer | i have that |
00:01.19 | fugitivo | what error do you get? |
00:01.28 | DrJES | Sec... |
00:02.07 | DrJES | Nov 25 20:01:53 ERROR[6888]: chan_zap.c:10250 setup_zap: Unable to reconfigure channel '7' |
00:02.07 | DrJES | Nov 25 20:01:53 WARNING[6888]: chan_zap.c:11010 reload: Reload of chan_zap.so is unsuccessful! |
00:02.28 | fugitivo | pastebin your zaptel.conf and zapata.conf |
00:03.23 | *** join/#asterisk grimse (n=grimse@p5481CCBF.dip.t-dialin.net) |
00:03.44 | DrJES | Will that not flood the channel? |
00:03.52 | fugitivo | ~pastebin |
00:03.54 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
00:04.15 | DrJES | Oh, you just want to access it? Sec... |
00:05.13 | DrJES | wget http://ednet.ns.ca/~macleajb/z.txt |
00:05.27 | dudes | Rawplayer - 101,2 |
00:05.44 | dudes | pastebin your extension |
00:06.27 | fugitivo | DrJES: you only have 4 channels there |
00:06.53 | DrJES | Sorry... Sec, I had them commented out.... |
00:07.47 | DrJES | Reget it please :) |
00:07.56 | DrJES | Added sample to channels. |
00:10.13 | DrJES | Of course with that config it fails on channel 5 : |
00:10.17 | DrJES | Nov 25 20:08:08 ERROR[6925]: chan_zap.c:10250 setup_zap: Unable to reconfigure channel '5' |
00:10.17 | DrJES | Nov 25 20:08:08 WARNING[6925]: chan_zap.c:11010 reload: Reload of chan_zap.so is unsuccessful! |
00:10.36 | Rawplayer | dudes: http://www.nomorepasting.com/paste.php?pasteID=53336 |
00:11.42 | dudes | does 100 work |
00:11.49 | h3x | why dosent lilo just hack ircd to add pastebins to irc itself. |
00:11.53 | fugitivo | DrJES: your zaptel.conf |
00:12.08 | DrJES | Sec.... |
00:12.29 | h3x | or add a bot for it |
00:13.07 | Qwell[] | h3x: /j #h3xpaste |
00:13.07 | DrJES | wget http://ednet.ns.ca/~macleajb/zt.txt |
00:13.11 | DrJES | Comments removed |
00:13.23 | h3x | bah. |
00:13.24 | Qwell[] | or |
00:13.39 | Qwell[] | haha, oops |
00:13.43 | Qwell[] | /query h3x - ctrl-v |
00:14.13 | h3x | the only people that complain about channel flooding are assholes with single window clients |
00:14.23 | Rawplayer | dudes: you mean in sip.conf? |
00:14.29 | Qwell[] | I irc with 640x480 |
00:14.33 | Qwell[] | one paste kills me |
00:15.04 | dudes | Rawplayer - when you dial extension 100 does it work |
00:15.19 | fugitivo | DrJES: if your modules are fxo, you should use fxsks=5-6 |
00:16.17 | Rawplayer | dudes: no |
00:16.39 | Rawplayer | i get 404 not found |
00:16.40 | dudes | try sip debug and see what happens when you dial it |
00:16.49 | dudes | then it's not setup in sip.conf right |
00:17.05 | Rawplayer | shall i paste that to? |
00:17.36 | dudes | sure |
00:18.12 | Rawplayer | http://www.nomorepasting.com/paste.php?pasteID=53337 |
00:18.42 | Qwell[] | Thats a freakishly long url... |
00:18.53 | Qwell[] | almost better to just paste crap into the channel :p |
00:19.06 | Rawplayer | :| |
00:19.11 | Rawplayer | i'am gonne shower |
00:19.14 | Rawplayer | bb in 10 minutes |
00:19.15 | dudes | context=sip |
00:19.24 | dudes | in extensions it's in [numbers] |
00:19.30 | Rawplayer | hmm |
00:19.34 | dudes | so change that and then sip reload |
00:19.36 | Rawplayer | it should be 100? |
00:21.19 | dudes | context=numbers |
00:21.26 | DrJES | Both cards are the same |
00:21.56 | fugitivo | DrJES: change that in your zaptel.conf |
00:22.41 | DrJES | What about 7 and 8 ? |
00:23.03 | *** join/#asterisk ginvent (n=joseph@adsl-63-199-244-155.dsl.sndg02.pacbell.net) |
00:23.04 | DrJES | ztcfg shows: |
00:23.14 | DrJES | Channel 01: FXO Kewlstart (Default) (Slaves: 01) |
00:23.18 | DrJES | Channel 02: FXO Kewlstart (Default) (Slaves: 02) |
00:23.22 | DrJES | Channel 03: FXS Kewlstart (Default) (Slaves: 03) |
00:23.24 | Qwell[] | ~pb |
00:23.25 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
00:23.26 | DrJES | Channel 04: FXS Kewlstart (Default) (Slaves: 04) |
00:23.26 | DrJES | Channel 05: FXO Kewlstart (Default) (Slaves: 05) |
00:23.27 | DrJES | Channel 06: FXO Kewlstart (Default) (Slaves: 06) |
00:23.27 | DrJES | Channel 07: FXS Kewlstart (Default) (Slaves: 07) |
00:23.27 | DrJES | Channel 08: FXS Kewlstart (Default) (Slaves: 08) |
00:23.31 | ginvent | Need a bit of help. I just installed asterisk, and I can't get it running... what is the command to run directly (not as server)? |
00:23.41 | DrJES | So aren't 5-6 FKOs? |
00:23.42 | Qwell[] | ginvent: -c will run it in console mode |
00:24.06 | ginvent | What does this mean: ....asterisk: relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load |
00:24.06 | ginvent | Ouch ... error while writing audio data: : Broken pipe |
00:24.13 | test34 | How can I know if my phone line uses reverse polarity for hangups? (because when somebody gets the voicemail, it doesn't always detect the hangup..) |
00:24.28 | *** join/#asterisk L|NUX (n=linux@202.141.252.82) |
00:24.46 | ginvent | How can I fix the pipe? |
00:24.53 | m160858 | excuse me, i wrote this line register => 3108023056@sip.broadvoice.com:XXXXXXXXX:3108023056@sip.broadvoice.com/201 |
00:24.55 | Dr-Linux | i have 2 FXO cards (4 port each), so in the zaptel.conf it will be "fxsks=1-8" or "fxsks=8" or what ? |
00:24.58 | newl | test34: you call your carrier and ask if your service is provisioned with reversal on idle. |
00:25.04 | test34 | ginvent, try: asterisk -cvvvvvv |
00:25.07 | Dr-Linux | i have no FXS cards tho |
00:25.15 | m160858 | but, i not receive the call and this ext |
00:25.27 | ginvent | Test34, I get that same response |
00:25.38 | ginvent | 'Error while writing audio data... broken pipe. |
00:25.41 | m160858 | which should be my problem |
00:25.43 | newl | ginvent: that sounds like an mpg123/mpg321 issue. |
00:25.47 | ginvent | hmmm. |
00:25.51 | test34 | ok thanks newl |
00:25.53 | ginvent | I had it running before. |
00:26.07 | ginvent | I was using asterisk 1.0 no prob. |
00:27.05 | newl | test34: most PSTN services don't have it (at least here in .au and in the US). It's mainly targeted at pay phones and PABX systems. |
00:27.36 | DrJES | http://pastebin.ca/31263 I think has the failed module load |
00:27.45 | m160858 | :| |
00:27.47 | test34 | newl, ok I'm in the US.. is there anything else I should look into ? |
00:28.20 | m160858 | who speak spanish?? |
00:28.35 | newl | test34: not that I can think of. Though that's not to say that someone else here (or perhaps info on the wiki) may suggest something else. :) |
00:28.58 | ginvent | What is a relocation error? relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load |
00:29.01 | m160858 | please, i need help |
00:29.42 | test34 | http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html -> they have a couple of other suggestions there |
00:29.55 | test34 | like getting rid of my cheap x100p |
00:30.15 | m160858 | hey? i'm here xD |
00:30.41 | m160858 | i have problems with this register => 3108023056@sip.broadvoice.com:XXXXXXXXX:3108023056@sip.broadvoice.com/201 |
00:31.05 | m160858 | the calls doesn't send to the ext 201 |
00:31.57 | m160858 | somebody understand this problem? |
00:32.04 | m160858 | i have 5 accounts |
00:32.04 | newl | m160858: is extension 201 in the context assigned to that sip providers entry? If not, it may be falling into the default context and rejecting the call because it doesn't exist. |
00:32.12 | m160858 | yes |
00:32.24 | m160858 | of course, exists |
00:33.29 | newl | ginvent: that error would probably indicate a mismatch in the asterisk daemon version and the module version. Rebuild and reinstall fresh. |
00:34.46 | m160858 | i have 5 account, with 2 ext each 1 .. and 1 context por each account |
00:35.11 | m160858 | if, i call the ext 201 .. the call send to ext 208 |
00:36.48 | *** join/#asterisk muHaarib (n=chatzill@66.237.2.170.ptr.us.xo.net) |
00:37.08 | muHaarib | can anyone help a dumb guy with a quesiton on an incoming context? |
00:37.08 | test34 | newl, what if I enable hanguponpolarityswitch=yes and my phone company doesn't use it? it will only not detect the hangup or it will cause some more problems ? |
00:37.31 | test34 | muHaarib, dont ask to ask |
00:37.38 | muHaarib | Thanks |
00:37.58 | muHaarib | I use gafachi iax2 connection with the outside world (PSTN) |
00:38.01 | *** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300017.sympatico.ca) |
00:38.05 | m160858 | someguy can help me? |
00:38.16 | muHaarib | I get phone calls routed out perfectly from my phone |
00:38.32 | muHaarib | gafachi has provided me with an IAX2 connection |
00:38.37 | muHaarib | I set it up in iax.con |
00:38.41 | muHaarib | sorry iax.conf |
00:38.53 | muHaarib | and I set up two contexts in extensions.conf |
00:39.01 | muHaarib | [gafachi-outgoing] |
00:39.05 | ginvent | I keep getting the same error... I just rebuilt fresh... but I keep getting: asterisk: relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load |
00:39.05 | ginvent | root@gottlieb3:/usr/src/asterisk-addons-1.2.0 # Ouch ... error while writing audio data: : Broken pipe |
00:39.05 | ginvent | Ouch ... error while writing audio data: : Broken pipe |
00:39.12 | muHaarib | and [gafachi-incoming] |
00:40.17 | muHaarib | I can't for the life of me get [gafachi-incoming] to work ... now I know that its dumb user ... but if somebody could give me an example of an incoming context for an IAX2 connection |
00:40.32 | muHaarib | maybe I could be think better. |
00:40.41 | asteriskmonkey | anyone good with pri's and asterisk? |
00:41.00 | asteriskmonkey | my pri plays busy signals on numbers that are not in service |
00:41.17 | asteriskmonkey | sorry pri dosnt asterisk does... |
00:42.13 | test34 | ginvent, you can try to check google's results: http://www.google.com/search?q=error+while+writing+audio+data%3A+%3A+Broken+pipe+asterisk&start=0&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-US:official |
00:42.24 | DrJES | Well, recompiled zaptel. Reloaded modules, stop and started asterisk, now all channels show up... Go figure. But I'm :) happy it's now working. |
00:43.31 | asteriskmonkey | kram: drunkkilla: you around? |
00:43.46 | ginvent | Test, I didn't get any results. |
00:43.52 | Qwell[] | drunkkilla...haha |
00:43.59 | Qwell[] | drumkilla: ^ |
00:44.14 | asteriskmonkey | lol |
00:44.25 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
00:44.41 | asteriskmonkey | damn bbz every time i come in here your blasting that |
00:44.43 | benjk | can we please get this bbz guy banned from this channel |
00:44.59 | asteriskmonkey | dont want no advertisments in here please |
00:45.00 | test34 | ./ignore bbz |
00:45.01 | benjk | somebody with ops privileges PLEASE |
00:45.09 | asteriskmonkey | ./ignore bbz |
00:45.18 | benjk | ./ignore bbz |
00:45.24 | Qwell[] | ... |
00:45.26 | asteriskmonkey | without the . |
00:45.28 | benjk | how do I do that? |
00:45.31 | benjk | ah ok |
00:45.44 | benjk | nice one, thx |
00:46.29 | asteriskmonkey | anyhow Qwell, you know much about pri and hangup msgs? |
00:46.45 | asteriskmonkey | i get the proper one set but i get a fast bust instead or number out of service |
00:47.06 | Qwell[] | asteriskmonkey: no |
00:47.15 | asteriskmonkey | darn it |
00:48.27 | *** part/#asterisk Rav1974 (n=r@pool-68-161-69-3.ny325.east.verizon.net) |
00:51.41 | asteriskmonkey | i need a digium guy anyone of em here? |
00:51.51 | *** join/#asterisk Cresl1n (n=matt@72.146.33.157) |
00:52.28 | asteriskmonkey | twisted: you around |
00:52.38 | asteriskmonkey | drumkilla: you around? |
00:52.55 | benjk | hey guys, a friend of mine is trying to come to the channel but he can't seem to log on to freenode.net |
00:53.01 | benjk | he's getting [19:50] Info: Lookup error: Unknown host!! |
00:53.07 | benjk | any ideas? |
00:53.09 | Qwell[] | benjk: #irc.freenode.net |
00:53.36 | benjk | ok will ask there |
00:53.39 | benjk | thx |
00:54.27 | benjk | are you sure that's an existing channel? |
00:54.31 | kippi | how can i uninstall asterisk? |
00:54.38 | Qwell[] | benjk: no |
00:54.41 | Qwell[] | #freenode maybe |
00:54.46 | ginvent | I think my $150 frys special with ubuntu and asterisk has finally bitten me good. asterisk 1.0 no prob... asterisk 1.2... no workee |
00:54.47 | benjk | ah ok |
00:55.21 | mog_home | man i wish i had gotten a 150 dollar special... |
00:55.24 | ginvent | Ok... time to try a Domo arigato... Mr. REboot-o |
00:55.35 | Qwell[] | ginvent: GQ? |
00:55.39 | ginvent | mog, they have them everyonce and a while. |
00:55.44 | ginvent | yeah gq. |
00:55.44 | Qwell[] | Shittiest brand ever :p |
00:55.55 | Qwell[] | "Great Quality"...yeah, my ass |
00:55.55 | ginvent | Hey... they have been working for me for over 1 year. |
00:55.56 | ginvent | :D |
00:55.58 | mog_home | yeah but i dont have a frys out here |
00:56.03 | m160858 | hello? somebody can help me? |
00:56.07 | ginvent | mog... ah... that is a prob. |
00:56.09 | mog_home | i just need a shitty emachine |
00:56.19 | Qwell[] | mog_home: wait for a dell deal |
00:56.22 | mog_home | to test stuff on |
00:56.23 | ginvent | For an asterisk server they work great... except when running ubuntu like I am. :D |
00:56.25 | mog_home | yeah |
00:56.27 | Qwell[] | mog_home: They sometimes have stuff for ~$250 |
00:56.27 | *** join/#asterisk ceph__ (n=amit@adsl-146-57-227.mia.bellsouth.net) |
00:56.31 | Qwell[] | WITH a monitor... |
00:56.31 | ginvent | deal deal for $150? that will be the day. |
00:56.36 | mog_home | i dont need monitor |
00:56.36 | benjk | that really SUXXXX |
00:56.39 | ginvent | lol |
00:56.41 | ginvent | brb |
00:56.44 | mog_home | just need headless box for testing |
00:56.46 | benjk | you cannot write to #freenode |
00:56.50 | benjk | only listen in |
00:56.52 | m160858 | there exists some irc on spanish ? |
00:57.10 | benjk | so how do people get help when they can't get on? |
00:57.19 | Qwell[] | benjk: They don't |
00:57.21 | mog_home | whats he using to connect ben |
00:57.23 | Qwell[] | problem solved. :P |
00:57.25 | mog_home | its so easy |
00:57.26 | *** part/#asterisk m160858 (n=jsaenz@200.89.12.46) |
00:57.31 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
00:57.33 | benjk | the same client I am using |
00:57.44 | benjk | and I told him to do exactly what I do |
00:57.57 | m160858 | excuse, some guy speak spanish? |
00:58.04 | benjk | it seems that freenode tries to do some reverse lookup of some thing |
00:58.08 | benjk | [19:50] Info: Lookup error: Unknown host!! |
00:58.17 | ceph__ | Hello..can anyone recommend a good online store for IP phones that they've used with * |
00:58.27 | m160858 | or somebody knows a channel on spanish? |
00:58.44 | benjk | m160858: wait a few minutes |
00:58.55 | mog_home | asterisk-es maybe? |
00:59.08 | benjk | my friend is from Colombia |
00:59.17 | benjk | he is tryuing to come to the channel |
00:59.24 | benjk | you can talk to him in Spanish |
00:59.53 | mog_home | maybe russian |
00:59.53 | asteriskmonkey | ceph_: massivetel.com they sell voip stuff uber cheap... no onlne shop though :P i buy my digium iaxys from there |
01:00.55 | ceph__ | *monkey..thanks...have you ever tried voipsupply.com? |
01:01.27 | asteriskmonkey | no from what i head there decent prices but returns on bad product are a headache |
01:01.40 | asteriskmonkey | massivetel sells iaxys cheaper though :) |
01:02.04 | asteriskmonkey | ah 1 big difference massivetel=canada voipsupply=us |
01:02.56 | ceph__ | good to know. |
01:03.03 | test34 | is iax2 the best protocol to use with asterisk ? |
01:03.19 | Qwell[] | test34: there is no "best" protocol to use |
01:03.53 | benjk | IAX2 is the best protocol.period |
01:04.13 | mog_home | lol |
01:04.15 | Qwell[] | each have their benefits - even h323 |
01:04.19 | mog_home | no that is h323 |
01:04.24 | mog_home | and mgcp |
01:04.26 | mog_home | they own you |
01:04.31 | *** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com) |
01:04.31 | benjk | yeah the Ford Etzel also had some good |
01:04.34 | Qwell[] | sccp <3 |
01:04.55 | mog_home | blech |
01:04.56 | mog_home | qwell you had to go and take it to far |
01:05.08 | Rawplayer | dudes: still here? |
01:05.11 | Qwell[] | What, got a problem with sccp? :( |
01:05.18 | Qwell[] | sccp is hot |
01:05.19 | RoyK | ~lart himself |
01:05.19 | benjk | freenode really sucks ballz |
01:05.19 | test34 | ok I guess I will go read some more before I choose then.. so far the only advantage Ive read is that iax2 uses less ports.. which is not a big advantage for me |
01:05.22 | mog_home | well as a fat man i could never like skinny |
01:05.45 | mog_home | you use the chan_sccp from sourceforge |
01:05.47 | Qwell[] | mv chan_sccp chan_fccp |
01:05.52 | mog_home | lol |
01:05.57 | Qwell[] | oh, wait |
01:05.58 | mog_home | fccp kicks ass |
01:06.00 | Qwell[] | shit, that was chan_sip |
01:06.02 | Qwell[] | :D |
01:06.14 | asteriskmonkey | iax2 is supernice crips sound with ulaw :) |
01:06.19 | Rawplayer | [numbers] |
01:06.20 | Rawplayer | exten => 100,1,Wait,1 |
01:06.20 | Rawplayer | exten => 100,2,Dial(SIP/Kevin1) |
01:06.20 | Rawplayer | exten => 100,3,Hangup |
01:06.22 | Qwell[] | mog_home: Is the sf one the same as the berlios.de one? |
01:06.28 | mog_home | yeah i think so |
01:06.29 | Rawplayer | the phonenumber of kevin is 100 right? |
01:06.35 | mog_home | yes |
01:06.41 | RoyK | fccp. what is that? fucking chicks coming personally? |
01:06.44 | mog_home | with a pointless one second wait |
01:06.57 | mog_home | no royk |
01:06.59 | Qwell[] | No, it isn't |
01:07.06 | Qwell[] | mog_home: different one |
01:07.14 | mog_home | oh really |
01:07.21 | mog_home | where does berli one come from |
01:07.27 | Qwell[] | berlios.de :P |
01:07.28 | RoyK | mog_home: too bad |
01:07.32 | Qwell[] | chan-sccp.berlios.de I think |
01:07.35 | mog_home | who wrote it? |
01:07.58 | Qwell[] | Sergio something maintains it |
01:08.03 | Qwell[] | it was skinny, then something else |
01:08.14 | Qwell[] | maybe it was derived from the sf one |
01:08.30 | mog_home | mmhm |
01:08.34 | mog_home | and its straight gpl? |
01:08.44 | Qwell[] | <PROTECTED> |
01:08.46 | Qwell[] | <PROTECTED> |
01:08.47 | Qwell[] | <PROTECTED> |
01:08.50 | Qwell[] | got me...think so |
01:08.53 | mog_home | what features does it support? |
01:08.58 | Qwell[] | like all of sccp, heh |
01:09.00 | mog_home | yeah its based of the sf one |
01:09.04 | mog_home | really? |
01:09.06 | mog_home | wow |
01:09.07 | Qwell[] | I'd say it's pretty complete. hints and stuff |
01:09.11 | Qwell[] | realtime now :D |
01:09.14 | mog_home | didnt know it was that far along |
01:09.18 | Qwell[] | or, soon, if he accepts my patch |
01:09.29 | mog_home | i know the chan_skinny doesnt do everything |
01:09.44 | Qwell[] | he's done a great job with it, imo |
01:09.52 | mog_home | yeah def. |
01:09.56 | Qwell[] | the code is a bit ugly, doesn't follow the * guidelines, but... |
01:10.26 | Qwell[] | the config is a bit ugly (it's like the zap configs, where the part that defines what it is is at the end) |
01:10.28 | *** join/#asterisk ginvent (n=joseph@adsl-63-199-244-155.dsl.sndg02.pacbell.net) |
01:10.34 | benjk | not that the code which does follow the guidelines would be any prettier |
01:10.48 | test34 | lots of voip providers offer you pyramid schemes as opportunities ? |
01:10.50 | ginvent | I am still getting this error any help? asterisk: relocation error: /usr/lib/asterisk/modules/res_odbc.so: undefined symbol: ast_load |
01:10.56 | ginvent | I rebuilt |
01:11.00 | ginvent | I recompiled clean. |
01:11.06 | ginvent | not sure what it could be. |
01:11.07 | Qwell[] | in fact, today he's celebrating 6 months of chan_sccp2 |
01:11.16 | ginvent | No response on google. |
01:11.32 | ginvent | I don't want to go back to 1.0... |
01:11.57 | Qwell[] | ginvent: Did you remove the modules before you upgraded? |
01:12.00 | mog_home | chan skinny can act like both sides or just gateway like mgcp? |
01:12.07 | ginvent | Qwell? |
01:12.10 | Qwell[] | mog_home: I'm not sure, actually |
01:12.15 | ginvent | how would I do that? |
01:12.22 | mog_home | can you talk to cisco gateways |
01:12.25 | mog_home | or just phones |
01:12.29 | Qwell[] | ginvent: rm /usr/lib/asterisk/modules/* |
01:12.38 | Qwell[] | ginvent: then make, make install |
01:12.41 | Qwell[] | mog_home: not sure... |
01:12.43 | mog_home | make clean |
01:12.44 | mog_home | make install |
01:12.47 | Qwell[] | right |
01:12.51 | Qwell[] | what he said |
01:13.02 | nitram | Qwell[]: hinting already works in chan_sccp, i have that setup with a 7960 and a 7920 here |
01:13.16 | Qwell[] | nitram: yes, thats what I said |
01:13.20 | Qwell[] | it supports hints and such |
01:13.24 | nitram | mog_home: it's just for phones afaik |
01:13.46 | Qwell[] | and hints with 7960+7914 = <3 |
01:13.48 | mog_home | thats what i thought |
01:13.53 | ginvent | OK, trying it now... I rm /usr/lib/asterisk/modules/* then make clean... make... and make install |
01:14.06 | Qwell[] | ginvent: make install will compile, you can skip the make |
01:14.11 | ginvent | oops... hehe |
01:14.13 | ginvent | too late. |
01:14.13 | ginvent | :D |
01:14.17 | Qwell[] | doesn't matter |
01:14.20 | Qwell[] | make install won't recompile |
01:14.29 | ginvent | k |
01:14.31 | Qwell[] | just one less step |
01:14.43 | ginvent | Well, it can't hurt to try to re-build... again. |
01:14.56 | ginvent | although on this QC pc... it's a tad slow... hehe |
01:15.01 | Qwell[] | qc? |
01:15.10 | Qwell[] | gq? |
01:15.10 | ginvent | GQ I mean. |
01:15.13 | ginvent | lol |
01:15.13 | Qwell[] | right |
01:15.20 | ginvent | I am all doped up on turkey. |
01:15.29 | Qwell[] | Is it even a celeron? |
01:15.36 | Qwell[] | probably uses a transmeta chip or something, heh |
01:15.39 | ginvent | Sempron... nothing but the best. :D |
01:15.46 | Qwell[] | oh, wow |
01:15.48 | ginvent | with SIS |
01:15.52 | mog_home | sis rules! |
01:15.57 | ginvent | it's a sempron 2200 |
01:15.59 | Qwell[] | sempron > duron > celeron |
01:16.05 | mog_home | amen brother |
01:16.13 | ginvent | I just turned up the overclock a touch. hang on to your shorts! |
01:16.24 | Qwell[] | overclocked gq sempron...oh god |
01:16.28 | ginvent | lol |
01:16.38 | Qwell[] | I can just see that opening a portal to hell |
01:16.44 | Qwell[] | satan spawn stepping out... |
01:16.54 | ginvent | I am pretty sure the power supply can not handle it. |
01:16.57 | nitram | too bad, chan_bluetooth didn't get as far as chan_sccp ;) |
01:17.01 | ginvent | I have heard horror stories about it. |
01:17.10 | Qwell[] | heh, probably what, 250 watts? |
01:17.21 | *** join/#asterisk jaristizabal (n=jaristiz@69.79.133.185) |
01:17.24 | ginvent | I think they used the supplies from an old amiga 500 |
01:17.26 | ginvent | :D |
01:17.30 | benjk | finally |
01:17.35 | benjk | hi javier |
01:17.45 | jaristizabal | hi |
01:17.52 | ginvent | ok, I am on the make install |
01:18.06 | mog_home | hehe |
01:18.07 | ginvent | done... dare I asterisk -cvvvv |
01:18.09 | jaristizabal | great! |
01:18.27 | benjk | javier, there is somebody who needs some help in Spanish |
01:18.36 | ginvent | Holy crap that did it! |
01:18.36 | benjk | m160858, are you still here? |
01:18.42 | ginvent | you guys are DA BOMB!!!! |
01:18.44 | jaristizabal | i speak spanish |
01:18.49 | ginvent | BOMB!!!!!!!! |
01:18.51 | benjk | m160858 ??? |
01:19.00 | jaristizabal | who need help in spanish?? |
01:19.04 | benjk | m160858 |
01:19.07 | ginvent | Qwell, I should send you a GQ pc just for your tech support. |
01:19.15 | ginvent | Then you too can have your own gateway to hades. |
01:19.20 | kippi | great! |
01:19.20 | benjk | but it looks like he is gone now |
01:19.31 | kippi | I have just done a clean install of asteris |
01:19.34 | mog_home | man think of how much you could sell that for ginvent |
01:19.41 | mog_home | to like a group of gothes |
01:19.43 | mog_home | or some thing |
01:19.54 | benjk | gothes? |
01:19.57 | ginvent | lol mog... |
01:20.00 | kippi | just tried to start it and got this error |
01:20.07 | mog_home | yeah i added the e |
01:20.16 | mog_home | for emphasis |
01:20.33 | m160858 | hey? |
01:20.34 | kippi | http://pastebin.com/438312 |
01:20.43 | m160858 | benjk? |
01:21.19 | mog_home | m1 |
01:22.35 | jaristizabal | hola |
01:22.43 | jaristizabal | si |
01:22.48 | jaristizabal | quien te dijo? |
01:23.54 | benjk | javier, you should chat to m160858 in a private chat |
01:25.54 | Qwell[] | ginvent: yeah, do that |
01:26.48 | Qwell[] | ginvent: There is always paypal too. My own gateway to hell |
01:27.17 | mog_home | hes gone qwell |
01:27.24 | Qwell[] | yeah, my client sucks |
01:27.32 | Qwell[] | it's still tab completing it :p |
01:27.40 | mog_home | indeed |
01:27.48 | Qwell[] | hell, I couldn't /part a channel earlier |
01:27.52 | Qwell[] | tip: don't use chatzilla |
01:28.12 | Qwell[] | I tried to do a driveby poking of file, but my client didn't leave the channel afterwards |
01:29.34 | benjk | Qwell, it seems that most irc clients have serious issues |
01:29.45 | Qwell[] | benjk: none are this bad |
01:30.08 | benjk | maybe if you are a complete geek |
01:30.28 | benjk | if you just want to use it and not get in the way, then they all have serious flaws |
01:35.58 | fugitivo | bitchx runs whithout problems |
01:36.13 | mog_home | i have the best one |
01:36.14 | mog_home | telnet |
01:36.17 | niZon | anyone know if verizon phones can be used on other providers, or are they locked? |
01:36.25 | Qwell[] | niZon: unlock it |
01:36.33 | fugitivo | mog_home: ugly :) |
01:36.44 | niZon | I'd love to know how :\ |
01:36.49 | mog_home | its so easy no wonder its number 1 |
01:36.50 | niZon | without spending more money |
01:37.21 | *** join/#asterisk znoG (n=gs@OL101-122.fibertel.com.ar) |
01:38.02 | Qwell[] | niZon: google away |
01:38.21 | mog_home | why google when someone on ebay will do it for 10 bucks |
01:38.39 | Qwell[] | ship it to them, and they'll also clone your phone :p |
01:38.47 | Qwell[] | free calls for them :D |
01:39.05 | mog_home | lol im all for sharing qwell |
01:39.16 | mog_home | thankfully i dont have a phone |
01:40.25 | file[laptop] | Verizon is a CDMA provider anyway... what other provider would you use it on? |
01:40.31 | trixter | benjk: you alive? |
01:40.40 | benjk | yes I think I am |
01:40.53 | trixter | there are now 580,067 entries in my list, so adding a few wouldnt work :P |
01:41.07 | benjk | lemme see |
01:41.31 | trixter | pulling from my old list to make my current list better, like the netherlands I had 189 in my old one but this one had less info and it went to like 5 or 6 entries |
01:41.33 | benjk | you need to add 8821 |
01:41.53 | trixter | that will be tough my old list was only 5500 |
01:41.54 | benjk | another 8821 and you get a very lucky number :-) |
01:42.32 | benjk | just add a bunch of e164.org blocks |
01:42.58 | benjk | they give every member a block of 100 number in a non-existing country code |
01:43.06 | benjk | 80 something |
01:43.20 | benjk | 882 or something like that |
01:43.24 | trixter | ha |
01:43.29 | Dr-Linux | i have created an IVR in [ivr] context, i want if the caller dial any extension for [default] context during greetings, he should redirect to desire extension |
01:43.42 | *** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
01:43.45 | trixter | um what? |
01:43.47 | Dr-Linux | my all user's extensions are in [default] context |
01:44.04 | Dr-Linux | only [ivr] is there |
01:44.08 | Dr-Linux | any clue? |
01:44.12 | benjk | Dr-Linux, that's bad practise |
01:44.34 | trixter | in default exten => _X.,1,goto(ivr,1) |
01:44.36 | trixter | is that what you want? |
01:44.41 | trixter | I really dont understand the question |
01:44.55 | benjk | yes but you shouldn't have your users in the default context |
01:45.13 | Dr-Linux | http://pastebin.com/438292 |
01:45.14 | benjk | thats like giving world provileges to your system files |
01:45.25 | trixter | that depends |
01:45.28 | Dr-Linux | this is my IVR, and it works okey |
01:45.42 | benjk | I disagree |
01:46.00 | trixter | I think that you can prevent abuse from default if you want to |
01:46.01 | Dr-Linux | but i have more then 200 extensions, |
01:46.06 | trixter | whether or not people do is a different story |
01:46.21 | benjk | because even if you start off with a setup where you say "my users have no right to do anything" you may then later add some services which cost you money and you forget to lock it down |
01:46.37 | trixter | benjk: I got a plantronics m2500 bluetooth headset today for $10 ... one of those loss leaders to get you in the story |
01:46.43 | trixter | I am somewhat happy with my purchase : |
01:46.43 | Dr-Linux | :S |
01:46.55 | benjk | trix: nice |
01:47.02 | trixter | :) |
01:47.12 | benjk | did you already take delivery or did you just order it? |
01:47.25 | trixter | cash purchase |
01:47.28 | trixter | got the last one too |
01:47.33 | benjk | hehe |
01:47.35 | trixter | radio shacks sale today |
01:47.38 | joat | clearance? |
01:47.39 | trixter | limit 10 per customer, yes 10 |
01:47.50 | trixter | no its black friday, so almost every store has a big sale today |
01:47.50 | benjk | wow |
01:47.57 | benjk | ah ok |
01:48.06 | benjk | but its saturday |
01:48.11 | benjk | :-) |
01:48.16 | trixter | $59.99 - $30 sale == $29.99 with a $20 mail in rebate |
01:48.23 | trixter | it was friday :P |
01:48.26 | trixter | it still is here |
01:48.29 | joat | hmm... wonder if the local shack will have it |
01:48.36 | benjk | yeah I figured that |
01:48.37 | trixter | I expect to see a ton of these on ebay soon |
01:48.38 | trixter | $40 or so |
01:48.48 | trixter | cause its gonna be a lot of profit for anyone that got any amount |
01:48.50 | benjk | I remember this thread a few years ago on Digium's mailing list |
01:49.01 | benjk | Digium announced some new product |
01:49.07 | benjk | don't know what it was |
01:49.07 | trixter | ds3000? |
01:49.08 | asteriskmonkey | ok that is simple to do |
01:49.09 | trixter | :P |
01:49.28 | trixter | that and their 24 port T1/E1 card I dont think ever actually made it into production |
01:49.31 | benjk | anyway, Mark said something like "we should have these new <whatever> out this spring" |
01:49.54 | Qwell[] | 24 port T1/E1? |
01:49.56 | benjk | and some guy from Australia asked "is the northern or southern hemisphere spring" |
01:50.03 | trixter | that is their new not yet available product htey are selling |
01:50.04 | Qwell[] | surely you mean the TDM2400P? |
01:50.09 | trixter | the ds3000 (DS3 card) was the other one |
01:50.17 | benjk | No that was 3 years ago or so |
01:50.19 | Qwell[] | That's 24 analog ports |
01:50.28 | Qwell[] | not 24 T1/E1's :p |
01:50.38 | trixter | yeah actually that is what I meant 24 analog.. and afaik that doesnt ship for a couple more weeks |
01:50.48 | mog_home | mmm 24t1/e1 board... |
01:50.56 | trixter | the ds3000 is 28 T1 |
01:50.56 | trixter | :P |
01:50.57 | benjk | how much is the 24 port card? |
01:51.00 | mog_home | its already shipping trixter |
01:51.13 | benjk | mog: how much is it? |
01:51.18 | trixter | ahh, heard that it wasnt yet available from someone @digium last week or something |
01:51.21 | trixter | guess they were wrong |
01:51.36 | mog_home | well our resellers and distributors already should have some |
01:51.44 | mog_home | and i think we are selling it on site now? |
01:51.47 | mog_home | i could be wrong |
01:51.52 | trixter | not everyone there actually is in sales, or whatever, according to cnet's interview with mark they make $10M/year so odds are they have real departments |
01:52.02 | trixter | on which site? |
01:52.05 | mog_home | very true trixter |
01:52.09 | mog_home | our site |
01:52.14 | trixter | who is 'our' |
01:52.14 | mog_home | id have to go check |
01:52.15 | trixter | ? |
01:52.19 | mog_home | i try to stay out of sales |
01:52.23 | mog_home | i work at digium trixter |
01:52.26 | trixter | ahh |
01:52.59 | trixter | well then what about the ds3000? |
01:53.02 | benjk | mog, just rougly, is it in the 500 USD ballpark, 1000 USD ballpark? |
01:53.06 | trixter | that has been promised for a while ... :P |
01:53.21 | Rawplayer | arent there any decent samples of voip? |
01:53.29 | mog_home | depends on the config |
01:53.29 | Rawplayer | with sip and extensions.conf |
01:53.31 | Qwell[] | mog_home: Do you know if it's possible to buy a 410, then get the echo can later? |
01:53.32 | trixter | its module based iirc 4 ports on a module, so I would push towards the higher end of things |
01:53.39 | mog_home | yes qwell |
01:53.45 | mog_home | older boards need software upgrade |
01:53.48 | mog_home | which we do for free |
01:53.51 | Qwell[] | ahh, cool |
01:53.55 | mog_home | but any board bought for a while now should work |
01:54.05 | mog_home | its a little cheaper to get em bundled |
01:54.05 | Qwell[] | does it cost more to get a 410 then +1 than a 411 does? |
01:54.08 | Qwell[] | k |
01:54.10 | mog_home | but not that much more |
01:54.20 | mog_home | i think it is 1000 and 800 if you get it with board |
01:54.22 | mog_home | but i dont know |
01:54.50 | trixter | I say we go chase the sales guys with torches and pitchforks |
01:54.51 | Qwell[] | haha |
01:54.54 | Qwell[] | http://www.voipsupply.com/product_info.php?products_id=772 |
01:55.02 | Qwell[] | They marked the price DOWN from $84 to $89 |
01:55.17 | mog_home | did it go up 5 |
01:55.18 | mog_home | lol |
01:55.24 | mog_home | yay bugs |
01:55.35 | Qwell[] | THAT is why I'll NEVER buy from voipsupply |
01:55.41 | benjk | If Mark get the Japanese caller ID working for analog zaptel, we may actually order a few of those 24 port analog boards |
01:55.50 | Qwell[] | I saw a TDM400p with 4 FXS for like $150 once |
01:55.55 | mog_home | ooh thats hot benjk |
01:55.58 | Qwell[] | I called them on it, so they "fixed it", to like $500 |
01:56.00 | mog_home | thats like a dinner for me |
01:56.03 | Qwell[] | then I called them on it again |
01:56.09 | mog_home | yikes |
01:56.13 | benjk | BTW, mog, thanks for pulling Mark into this yesterday, that was very helpful |
01:56.20 | mog_home | i do what i can |
01:56.27 | mog_home | and i dont like people talking smack ^_^ |
01:56.31 | benjk | I have been trying to get him to do that for 2 yeas |
01:56.33 | benjk | ;-) |
01:56.44 | asteriskmonkey | wooo fixed the pri odd issue :) |
01:56.50 | Qwell[] | somebody got cid working for jp... |
01:56.53 | Qwell[] | who was that? |
01:56.59 | mog_home | dont yet |
01:57.02 | mog_home | mark is working on it |
01:57.07 | Qwell[] | I think I recall somebody saying they got it working already |
01:57.14 | mog_home | yeah benjk ^_^ |
01:57.20 | mog_home | it works with voicetronix hw |
01:57.21 | Qwell[] | no, somebody else... |
01:57.26 | Qwell[] | was like...3 weeks ago |
01:57.48 | mog_home | i missed it thats for sure |
01:57.55 | Qwell[] | maybe it was indications |
01:58.37 | benjk | mog, the voicetronix hardware is overpriced (1800 USD for a four port card here in JP) and it doesn't work so well, if it works at all |
01:59.11 | mog_home | eep |
01:59.36 | file[laptop] | meeo |
02:00.22 | Qwell[] | download the latest firmware!? |
02:00.38 | benjk | mog, Voicetronix have signed a deal with a Japanese company to sell their boards in Japan |
02:00.51 | benjk | the Japanese company specialises in Bayonne |
02:00.54 | benjk | not Asterisk |
02:01.09 | benjk | so all the work for Japan stuff is mostly going into Bayonne |
02:01.33 | mog_home | fun stuff |
02:01.36 | benjk | and the Japanese company sells those boards with a 300 % profit margin |
02:01.45 | benjk | that's how Japanese companies think |
02:01.51 | mog_home | man i need to make a hard phone |
02:01.58 | mog_home | and charge 100% markup |
02:02.03 | mog_home | and call it misco |
02:02.04 | Qwell[] | mog_home: 800% |
02:02.08 | benjk | sell five boards a month with 300 or 500 or 1000 or 2000 percent markup |
02:02.10 | mog_home | nah im not gready |
02:02.10 | Qwell[] | call it... |
02:02.12 | Qwell[] | Poortel |
02:02.13 | Qwell[] | :D |
02:02.14 | mog_home | lol |
02:02.18 | mog_home | genius |
02:02.31 | Qwell[] | I've got a bunch of names that would violate trademark |
02:02.32 | benjk | instead of selling 100 200 500 cards with 20 25 or 30 percent markup |
02:02.34 | mog_home | i think i will have "random" pricing |
02:02.36 | Qwell[] | like Qwell Communications |
02:03.02 | mog_home | some one calls in |
02:03.04 | mog_home | its 1 grand |
02:03.07 | mog_home | some one else |
02:03.08 | Qwell[] | rofl |
02:03.10 | mog_home | 50 cents |
02:03.21 | benjk | to be fair though, those guys have gone through the trouble of taking out Japanese type approval for the Voicetronix board |
02:03.22 | Qwell[] | Don't make it that extreme...people will just call back |
02:03.39 | mog_home | heh thats why i have to get ani information on my t1 |
02:03.43 | mog_home | catch people doing that |
02:03.45 | Qwell[] | heh |
02:03.46 | mog_home | price doubles |
02:04.09 | mog_home | and i have to make my hw hard to buy like cisco or polycom |
02:04.14 | Qwell[] | ugh |
02:04.22 | Qwell[] | Make them sign an ICA |
02:04.23 | mog_home | to own my phone you must have a phd in telephony and go to the misco training course |
02:04.31 | mog_home | and to even talk to me |
02:04.31 | Qwell[] | and then have NO mention AT ALL on your site, of HOW to complete the ICE |
02:04.33 | Qwell[] | ICA |
02:04.37 | mog_home | ill need 3 ndas |
02:04.41 | mog_home | yes |
02:04.50 | mog_home | man id be a billionaire |
02:04.50 | Qwell[] | That's what cisco does, heh |
02:04.51 | *** join/#asterisk ThePeopleGA (n=kemtram@rev-204.120.18.37.genesiswireless.us) |
02:04.52 | mog_home | over night |
02:04.57 | mog_home | without even selling anything |
02:05.05 | mog_home | then i would have to flea the country |
02:05.09 | mog_home | go to portugual |
02:05.12 | mog_home | and go by jose |
02:05.14 | Qwell[] | because you never actually made any hardware? |
02:05.19 | trixter | why portugal? |
02:05.28 | mog_home | who would think i would go to portugal |
02:05.28 | Qwell[] | no extradition treaty? |
02:05.37 | mog_home | no one would see it comming |
02:05.41 | Qwell[] | we would |
02:05.44 | Qwell[] | because now we know |
02:05.49 | Qwell[] | foiled |
02:05.54 | mog_home | heh thats why i am really going to khazakastan |
02:05.54 | trixter | poprtugal is friends with the US, they are part of nato, odds are they have an extradition treaty to here at least :) |
02:05.56 | mog_home | oops |
02:06.14 | Qwell[] | You can come to Qwellsakistan |
02:06.21 | Qwell[] | We don't have an extradition treaty |
02:06.31 | Qwell[] | no fcc either |
02:06.35 | mog_home | heh qwellsakistan sounds like a fun place to go |
02:06.38 | Qwell[] | it is |
02:06.40 | trixter | but would you really want to goto a place where its short name is 'Qwell sak' ? |
02:06.42 | mog_home | where you from qwell? |
02:06.49 | Qwell[] | Qwellsakistan... |
02:07.13 | mog_home | which is in the vicinity of |
02:07.18 | alephcom | How are the taxes there? |
02:07.27 | trixter | 200% |
02:07.27 | Qwell[] | alephcom: 120% income tax |
02:07.32 | trixter | ha |
02:07.46 | Qwell[] | but, we have free healthcare |
02:07.52 | alephcom | Lol, I think he's confused. He says Qwellsakistan and means Canada. :-) |
02:07.54 | Qwell[] | and we're communist |
02:08.10 | mog_home | 120% glad ill be unemployed |
02:08.15 | Qwell[] | mog_home: exactly |
02:08.16 | trixter | ok french canada :P |
02:08.26 | mog_home | welfare? |
02:08.37 | Qwell[] | mog_home: 6000 a week |
02:08.55 | alephcom | Hmmm, I think I'll be emigrating. |
02:08.59 | mog_home | freaking sweet |
02:09.00 | Qwell[] | which is also taxed |
02:09.06 | mog_home | ill have to tell mark i got a better offer |
02:09.08 | alephcom | oops. |
02:09.10 | trixter | but 6000 qwellsak dollars is like $0.0000001 you couldnt even get 1 minute of voip time with it |
02:09.17 | Qwell[] | trixter: free voip |
02:09.31 | trixter | is that free voip as in taxed? |
02:09.32 | alephcom | only if you're unemployed though. |
02:09.39 | Qwell[] | trixter: no, truly free |
02:09.46 | trixter | who provides that? |
02:10.01 | Qwell[] | we contracted with vonage |
02:10.08 | mog_home | ewwww |
02:10.09 | mog_home | vonage |
02:10.10 | alephcom | rofl |
02:10.15 | Qwell[] | past tense |
02:10.18 | Qwell[] | they cut us off |
02:10.19 | trixter | heh.. should contract with www.trxtel.com :) |
02:10.22 | Qwell[] | So now we go through fwd |
02:14.00 | asteriskmonkey | contract with massivetel damn it :) |
02:16.17 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
02:17.42 | asteriskmonkey | damn free calls to all those countries |
02:17.44 | asteriskmonkey | that looks sick |
02:17.46 | asteriskmonkey | is it a joke |
02:18.54 | *** join/#asterisk diego_br (n=diegovie@200-213-122-121-mns.cpe.vivax.com.br) |
02:19.42 | Vco | of course, a good percentage of those taxes also go towards subsidizing the development of beer that doesn't taste like piss |
02:20.05 | Qwell[] | Vco: No, we're a vodka drinking people |
02:20.13 | Vco | meh.. |
02:20.30 | Vco | unless it's got some gin and an olive involved i dont' really touch vodka |
02:20.51 | trixter | asteriskmonkey: nope, it should be up soon for all |
02:21.08 | trixter | still testing |
02:21.31 | asteriskmonkey | how are you doing it, using enum? |
02:21.58 | trixter | not entirely |
02:22.04 | trixter | not everything is in enum |
02:22.04 | asteriskmonkey | spill |
02:22.11 | asteriskmonkey | ah dundi/enum |
02:22.22 | trixter | mostly through donated services and equipment |
02:22.44 | trixter | right now there is no dundi or enum built in |
02:23.03 | trixter | its all straight to a pstn gateway i do plan on adding enum and dundi though |
02:23.55 | asteriskmonkey | well whats the local pstn its on ? |
02:24.05 | asteriskmonkey | and is it a pri ? ds3 oc-12? |
02:24.13 | trixter | it uses several |
02:24.24 | asteriskmonkey | several what? bits of string? |
02:24.32 | asteriskmonkey | whats it connected with... |
02:24.35 | trixter | yes that must be it |
02:24.37 | mog_home | pris i imagine |
02:24.43 | Dr-Linux | asteriskgeeks: what will be the sequence of the [ivr] and [default] contexts ? |
02:25.00 | asteriskmonkey | default first ivr next |
02:25.28 | asteriskmonkey | well trixter where are you located ill trade you some channels on my local pri |
02:25.35 | Dr-Linux | okey let me try |
02:26.18 | trixter | trx telecommunications, inc has multiple facilities depending on what is being terminated.. trx is a real clec in montana but has servers elsewhere for other operations |
02:26.21 | Dr-Linux | asteriskgeeks: i understand and where will be include => default |
02:26.32 | Dr-Linux | as sequence? |
02:26.46 | asteriskmonkey | in the [ivr] context |
02:27.12 | asteriskmonkey | http://www.voip-info.org |
02:27.17 | Dr-Linux | okey let me try |
02:27.26 | asteriskmonkey | trixter you own that clec? |
02:28.36 | asteriskmonkey | darn it i run a hackjob of a clec :) would be nice to find some other partners to swap ld with |
02:28.56 | Qwell[] | asteriskmonkey: I give you free minute to Qwellsakistan |
02:29.16 | asteriskmonkey | lol non one has internet there |
02:29.23 | Qwell[] | everyone has internet |
02:29.25 | Qwell[] | free wifi |
02:29.40 | *** join/#asterisk coppice (n=chatzill@168.155.17.210.dyn.pacific.net.hk) |
02:29.48 | asteriskmonkey | dude you not talking about your house are you? |
02:29.48 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
02:29.54 | Qwell[] | no |
02:30.17 | Qwell[] | front and back porch, and surrounding areas too |
02:30.17 | asteriskmonkey | lol |
02:30.17 | Vco | heh..heh.. |
02:30.17 | Qwell[] | I mean... |
02:30.18 | mog_home | ooh the back porch |
02:30.18 | Qwell[] | inside and outside castle |
02:30.28 | mog_home | i get charged like 100 dollars a minute for calls there |
02:30.36 | asteriskmonkey | damn |
02:30.40 | Vco | and around the car hold |
02:30.40 | nick125 | wow |
02:30.50 | Qwell[] | Vco: No cars in Qwellsakistan |
02:31.01 | *** join/#asterisk kks (n=kks@202.73.8.130) |
02:31.01 | asteriskmonkey | hey i got to play with some kick as wireless sip phones today |
02:31.04 | Qwell[] | cars must be outsourced...to parkinglotsakistan |
02:31.19 | mog_home | lol |
02:31.24 | asteriskmonkey | they take sim chips so you can use fido then user voip when you pick up a wireless connection |
02:32.17 | trixter | asteriskmonkey: yes I do own it |
02:32.50 | asteriskmonkey | so you have a multi hommed t1/pri then or acuallt mult location multi pri |
02:33.03 | Vco | trixter p0wnz |
02:34.22 | Dr-Linux | asteriskmonkey: http://pastebin.ca/31276 |
02:34.27 | Dr-Linux | please check it out |
02:34.44 | Dr-Linux | bcoz its not working, i believe i'm missing/wrong somewhere |
02:34.57 | *** join/#asterisk camonz (n=camonz@200.8.21.123) |
02:36.29 | asteriskmonkey | apart from your overuse of the background function it should work reload your extensions |
02:37.33 | asteriskmonkey | oh wait try putting include => default affter # |
02:37.33 | asteriskmonkey | exten => s,22,Background(to-reach-operator) |
02:37.42 | asteriskmonkey | then reload |
02:37.51 | asteriskmonkey | then foward payments to digium :P |
02:37.55 | Dr-Linux | okey wait |
02:38.37 | Dr-Linux | include => default affter # << sir i didnt understand this |
02:38.59 | Qwell[] | don't put users in default |
02:39.01 | Qwell[] | thats unsafe |
02:39.43 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
02:39.52 | Dr-Linux | Qwell[]: i will change the contexts once i understand a few things |
02:40.58 | Dr-Linux | where should i put this "#" sign ? |
02:41.07 | Qwell[] | nowhere |
02:42.01 | Dr-Linux | but this doesn't work for me >> http://pastebin.ca/31276 |
02:42.08 | Qwell[] | Why doesn't it work? |
02:42.36 | Dr-Linux | oohhh |
02:42.54 | Dr-Linux | exten => 7777,1,Goto(ivr,s,1) << this extension should under the [default] context? |
02:43.02 | *** join/#asterisk ntwrknggeek (n=ntwrkngg@pool-70-105-186-165.alt.east.verizon.net) |
02:43.16 | Qwell[] | Why would you want to go back to the ivr if somebody dialed an extension? |
02:43.48 | ntwrknggeek | I am looking for some help with an asterisk@home install... |
02:44.02 | Qwell[] | ntwrknggeek: #asteriskathome |
02:44.16 | Dr-Linux | Qwell: yes sir |
02:44.24 | ntwrknggeek | thanks qwell |
02:44.25 | Qwell[] | *@h only barely resembles asterisk |
02:44.36 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
02:44.39 | ntwrknggeek | oh really??? |
02:44.40 | Dr-Linux | my actuall problem is that, i have more then 200 users in [default] context |
02:44.51 | ntwrknggeek | what do you mean quell |
02:45.23 | m160858 | benjk? |
02:45.30 | Qwell[] | Dr-Linux: okay, so whats the problem? |
02:45.51 | mog_home | asterisk @ home isnt horrid just dont expect help here |
02:46.00 | Qwell[] | it is horrid |
02:46.11 | Qwell[] | mog_home: You heard the story about my tech writer, right? |
02:46.24 | mog_home | ? |
02:46.30 | mog_home | i am the biggest a@h hater |
02:46.33 | Qwell[] | I'm having to teach her all of the options in AMP, so that she can mockup a new GUI for me to write |
02:46.36 | mog_home | but the core is the same |
02:46.46 | mog_home | your writing a gui qwell? |
02:46.51 | Qwell[] | supposed to be |
02:46.58 | Qwell[] | It's not gonna suck though |
02:47.06 | Qwell[] | besides being written in C#/ASP.NET :D |
02:47.23 | ntwrknggeek | What is the difference between *@h and *?!? |
02:47.33 | Qwell[] | ntwrknggeek: *@h completely butchers the configs |
02:47.34 | mog_home | BLECH |
02:47.36 | mog_home | what the hell |
02:47.46 | Dr-Linux | Qwell: problem is that, caller can listen all greeting etc, but i want caller if he wanna dial any extension from [default] context during greeting he should |
02:47.49 | Qwell[] | mog_home: mine isn't gonna butcher said configs. :D |
02:47.55 | Qwell[] | Dr-Linux: so let him |
02:48.00 | mog_home | why not do the right thing qwell |
02:48.06 | mog_home | go make rt stuff |
02:48.15 | Qwell[] | mog_home: It is going to use realtime |
02:48.23 | Qwell[] | and not the BS that AMP does, where it has its own tables |
02:48.29 | mog_home | i mean all of configs in rt |
02:48.32 | Qwell[] | yeah |
02:48.33 | mog_home | dont touch flat files |
02:48.37 | Qwell[] | not going to |
02:48.40 | mog_home | if you do its gonna be uggies |
02:49.07 | Qwell[] | gonna even do blah.conf => odbc,asterisk,blah.conf |
02:49.15 | Qwell[] | so I grab the general sections and such too |
02:49.22 | Dr-Linux | Qwell[]: :S |
02:49.39 | mog_home | nice |
02:49.58 | Qwell[] | mog_home: it'll be decent. It'll still suck because it's a gui, but... |
02:50.31 | Qwell[] | my work is actually putting some money behind writing it. It'll have documentation, a few fulltime programmers, testing... |
02:50.58 | *** part/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
02:50.59 | mog_home | im fine with guis |
02:51.06 | mog_home | just as long as i dont have to touch em |
02:51.08 | *** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
02:51.13 | Qwell[] | heh |
02:51.23 | mog_home | what i want though is a phpmyadmin for ncurses |
02:51.26 | mog_home | so if customer has rt |
02:51.32 | mog_home | i can go view mysql tables fast |
02:51.34 | mog_home | as i am lazy |
02:51.37 | mog_home | lazy as all |
02:51.38 | Qwell[] | we're just not going to be able to admin * for all of the different sites |
02:51.44 | *** join/#asterisk jaristizabal (n=jaristiz@69.79.133.185) |
02:51.49 | Qwell[] | so we need something that doesn't completely suck, for them to config stuff on their own |
02:51.54 | Dr-Linux | asteriskmonkey: sir really thanks, you were right, i'm done now it works what i want :) |
02:52.10 | ntwrknggeek | Qwell[] but for the basic home use and someone who is not all that good with linux *@h is ok? |
02:52.17 | Qwell[] | ntwrknggeek: not really |
02:52.29 | Qwell[] | You'll never learn * or Linux if you use AMP |
02:52.30 | mog_home | doing asterisk is really easy |
02:52.37 | mog_home | exactly |
02:52.48 | camonz | hi, i've a question, since * is OSS isn't the bussiness to provide administration for the service? |
02:52.57 | Dr-Linux | <Qwell[]> You'll never learn * or Linux if you use AMP << i gree |
02:52.57 | Qwell[] | camonz: huh? |
02:53.02 | *** join/#asterisk wildcard0 (n=generic@S0106006097e16040.vc.shawcable.net) |
02:53.07 | camonz | installing, configuring the dialplan, etc |
02:53.14 | Qwell[] | camonz: What business? |
02:53.17 | camonz | at least that's the way it's done over here |
02:53.20 | camonz | voip |
02:53.26 | Qwell[] | huh? |
02:53.30 | camonz | setting pbx for enterprises and such |
02:53.31 | wildcard0 | hey. im trying to connect asterisk to an emergent system and im getting 'Got SIP response 400 "Bad Request" back from <ip>' |
02:53.36 | wildcard0 | any ideas on how i could fix this? |
02:53.37 | Qwell[] | You saying the provider should setup your pbx? |
02:53.41 | camonz | nope |
02:53.56 | camonz | another enterprise |
02:54.02 | Qwell[] | camonz: well, people aren't going to do it for you for free |
02:54.02 | ntwrknggeek | Qwell[] Do you recommend a good resource for learning * with AMP and a little bit of linux? |
02:54.06 | camonz | of course |
02:54.18 | Qwell[] | ntwrknggeek: There are no resources for learning * when using amp. amp breaks things |
02:54.30 | Qwell[] | If you want to actually learn asterisk, ... |
02:54.32 | Qwell[] | ~docs |
02:54.33 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
02:54.35 | camonz | that's what i mean, in making a GUI that makes easy to configure things for the end user isn't that going to hurt your bussiness as a consultant |
02:54.44 | benjk | I recommend learning about networking first if you have to learn "a little bit of Linux" |
02:54.57 | benjk | I was helping this German chap last night |
02:55.07 | benjk | he got all sorts of SIP error messages |
02:55.15 | camonz | by example over here there's a company that charges about 9000$ for instalation |
02:55.21 | benjk | turns out he diodnt have any DNS |
02:55.27 | wildcard0 | was one of them '400 "Bad Request"'? |
02:55.32 | camonz | plus you have to contract them for maintenance |
02:55.37 | benjk | but he managed to get on to IRC |
02:55.47 | ntwrknggeek | Thanks Qwell[] |
02:55.51 | benjk | though he couldn';t tell me how he did it |
02:55.55 | camonz | i.e. adding mailboxes, configuring the dialplan, and things like that |
02:56.07 | benjk | he had no clue whatsoever about networking |
02:56.26 | benjk | if you don't know about networking on Unix, forget about Asterisk |
02:56.32 | mog_home | buy a router, plug things into said router |
02:56.41 | mog_home | set asterisk box as static |
02:56.46 | mog_home | and port forward or dmz |
02:56.47 | mog_home | your done |
02:57.05 | benjk | mog, if you cant' condigure your Asterisk box to connext to the router then the router will do you no good |
02:57.30 | benjk | if your Asterisk box cannot resolve any DNS names, you won't get very far |
02:57.42 | mog_home | you pull a dhcp |
02:57.48 | mog_home | and then you just set it static |
02:57.52 | mog_home | networking is easy |
02:58.00 | mog_home | if you need help |
02:58.03 | benjk | well, that's why I said, learn about networking on Unix |
02:58.03 | mog_home | buy a linksys |
02:58.09 | mog_home | ^+^ |
02:58.22 | benjk | so that you know 1) what you have to do and b) how to do it |
02:58.51 | benjk | what good does that Linksys do you if your Asterisk box won't talk to it |
02:59.04 | mog_home | <PROTECTED> |
02:59.06 | mog_home | oops |
02:59.10 | file[laptop] | ;) |
02:59.19 | mog_home | ^_^ |
03:00.34 | wildcard0 | so...SIP errors? anyone? anyone? |
03:00.55 | camonz | i mean, the real question is wich bussiness models are you using to make profits with * |
03:01.10 | wildcard0 | mine's working :) |
03:02.31 | asteriskmonkey | what make money using * ??? |
03:02.34 | asteriskmonkey | tell me more |
03:03.20 | wildcard0 | ya. unfortunately im not allowed to talk about it |
03:04.59 | camonz | :->, here in vzla there are just 1 or 2 bussiness doing the research on how * works so they can begin doing installations |
03:05.31 | camonz | so, the concept of voip is kind of new over here |
03:05.35 | Vco | i'd suspect your math may be off |
03:05.58 | Vco | see...you need to actually know of the other ones out there, and count them too |
03:06.27 | camonz | sure, maybe more than 1 or 2, but still not many |
03:07.22 | *** join/#asterisk ritesha (n=ritesha@c-24-6-80-22.hsd1.ca.comcast.net) |
03:07.33 | *** part/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
03:07.54 | ritesha | hello. |
03:08.59 | Dr-Linux | asteriskmonkey: if the caller doesn't press anything in end of greeeting, it says "goodbye" but user is still connected, |
03:09.02 | Dr-Linux | here |
03:09.03 | Dr-Linux | exten => t,1,Playback(vm-goodbye) |
03:09.04 | Dr-Linux | exten => t,2,Hangup() |
03:09.06 | ritesha | I need to understand how asterisk connects to database. Specifically, how to add SIP extenstions to a database so that asterisk can read it. Do I need to have a standard table name e.g. sipfriends or is there some toher way to specify a table name to asterisk? |
03:09.22 | Dr-Linux | i want the user Hangup |
03:10.04 | Dr-Linux | or start again from first |
03:10.32 | ritesha | i saw some examples where the sip.conf just specifies the database name, username and passwd. I am curious to to know how does the asterisk know whihc table belongs to sip extensions for bunch of other tables that my database may have? |
03:11.11 | ritesha | don't see details description about asterisk + database anywhere include voip-info.org |
03:11.15 | ritesha | please help... |
03:14.58 | Qwell[] | ritesha: realtime |
03:15.23 | ritesha | Qwell: realtime meaning? |
03:15.32 | Qwell[] | realtime is what you want to google |
03:15.41 | ritesha | oh!! thanks... let me try |
03:18.51 | jaristizabal | ritesha, maybe help you: http://www.asteriskguru.com/tutorials/realtime_pgsql.html |
03:21.03 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
03:21.09 | *** join/#asterisk rowter (n=SilverDr@201.135.26.195) |
03:21.20 | ritesha | thanks Qwell and jaristizabal. Found the information on voip-info.org. Reading through it. |
03:21.50 | jaristizabal | ok |
03:28.09 | *** join/#asterisk BleedingMe (n=Bleeding@ppp-71-137-216-107.dsl.scrm01.pacbell.net) |
03:29.13 | BleedingMe | anyone know an easy way to generate a random number variable in asterisk... like, generate a random number between 1 and 100 |
03:29.16 | BleedingMe | ? |
03:29.23 | mog_home | hrmm |
03:29.27 | mog_home | well |
03:29.37 | mog_home | you can execute system commands |
03:29.37 | Vco | flip some coins? |
03:29.48 | mog_home | and then read them off with readfile |
03:29.51 | mog_home | hmm |
03:30.00 | mog_home | oh wait |
03:30.01 | mog_home | i got it |
03:30.05 | mog_home | the unique id |
03:30.06 | mog_home | of the call |
03:30.08 | mog_home | its random |
03:30.15 | BleedingMe | unique id? |
03:30.25 | Chuji | app_mysql |
03:30.37 | Chuji | grab a key |
03:31.05 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
03:31.10 | mog_home | bbz |
03:31.12 | mog_home | begone |
03:32.35 | alephcom | bbz: I messaged you and you didn't respond. You mustn't want to be rid of them too badly |
03:34.14 | phsdshft | Does anyone have experience working with Sipura devices behind NAT connecting to an asterisk server that has a public IP? |
03:35.08 | alephcom | I little bit. What's going on? |
03:35.18 | phsdshft | well.. I keep getting one way audio |
03:35.42 | Vco | get a second phone |
03:35.44 | Vco | hahahaha |
03:35.49 | phsdshft | I can hear the phone menu on the asterisk server (or person off of the asterisk server) but they can't hear me |
03:36.06 | phsdshft | me meaning the phone off of the sipura device |
03:36.19 | alephcom | What kind of firewall? |
03:37.04 | phsdshft | its a linksys broadband router |
03:37.25 | phsdshft | I have ports 5060, 5061, and 10000 - 20000 forwarded to the sipura |
03:38.22 | phsdshft | however |
03:38.30 | phsdshft | I get one way audio even if its directly off of my cable modem |
03:38.59 | phsdshft | the other side has an access list that drops traffic going to the asterisk server... but it allows ports 10k - 20k, 5060 and 5061 |
03:39.34 | phsdshft | (udp) |
03:44.41 | phsdshft | soo.. yeah.. heh |
03:46.51 | alephcom | Hmmm, I'm not sure what the problem is. If it does the same thing directly off of the cable modem is very interesting. |
03:46.53 | benjk | phsdshft: its not as simple as forwarding |
03:47.15 | phsdshft | benjk: Right.. the sipura has to be configured to do nat as well.. |
03:47.17 | benjk | because it matters what's inside the SIP messages |
03:47.26 | phsdshft | I specified the ext ip in the sipura as my public ip |
03:47.30 | phsdshft | set nat mapping to yes |
03:48.04 | benjk | the TO: FROM: and the CONTACT: fields must match the *endpoint*, not your router |
03:48.08 | phsdshft | and set ... handle via received, substitute via addr and everything but STUN to yes |
03:48.34 | phsdshft | benjk: endpoint meaning the private, or public address? |
03:48.52 | *** join/#asterisk nxtw (n=matt@adsl-69-221-114-151.dsl.akrnoh.ameritech.net) |
03:48.57 | benjk | the private address |
03:49.28 | benjk | ultimately |
03:49.48 | benjk | but it depends on the sip engine how the device handles the traffic |
03:50.29 | phsdshft | well... on the asterisk side I have (changing it to one line to make it not scroll the room): |
03:50.41 | benjk | if it gets the traffic looks in the field and discovers that the address isn't its own, then says "hey, that's not meant for me" and drops it, well then you get problems |
03:50.49 | BleedingMe | is there a way to play an audio file directly to a channel while on the phone? |
03:51.37 | JunK-Y | BleedingMe: u mean with the CLI? |
03:51.53 | BleedingMe | sure |
03:51.59 | phsdshft | [2001]; type=friend; username=2001; host=dynamic; context=customersin; allow=all; dtmfmode=inband; nat=yes; qualify=yes; |
03:52.09 | phsdshft | (the ; means new line, they arent actually in the config) |
03:52.54 | benjk | didn't you say sipura? |
03:53.32 | phsdshft | yes.. thats from the asterisk side |
03:53.42 | phsdshft | I have control over both endpoints |
03:54.03 | benjk | the asterisk box is on a public address? |
03:54.06 | phsdshft | yes |
03:54.22 | benjk | and the sipura is behind NAT |
03:54.33 | phsdshft | it has a router with an acl in front of it... thats it.. and that acl allows 10k - 20k, 5060 and 5061 through (udp) |
03:54.41 | phsdshft | correct |
03:55.05 | benjk | the asterisk is behind a NAT router? |
03:55.31 | benjk | forwarding addresses is not the same as having a public address |
03:56.00 | phsdshft | no.. it has a public (not private) address, and no nat is being done |
03:56.05 | phsdshft | its behind a cisco 7204 |
03:56.06 | benjk | ok |
03:56.18 | phsdshft | the other side is behind a linksys (the side with the sipura) |
03:56.25 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
03:56.36 | benjk | did you configure rtp.conf to make sure it doesn't use any ports outside of your range |
03:57.56 | phsdshft | yes.. it says 10k to 20k, which is inside of the range |
03:57.56 | benjk | ok |
03:58.02 | phsdshft | of course, at the moment I'm noticing that after upgrading the sipura's firmware I no longer am receiving RTP packets... heh doh |
03:58.04 | benjk | and the sipura registers with the Asterisk server? |
03:58.05 | phsdshft | (or sending) |
03:58.09 | phsdshft | yes |
03:58.23 | phsdshft | but it didnt work before upgrading the firmware either |
03:58.31 | phsdshft | although at the moment it has no audio at all |
03:59.12 | benjk | have you done a sip debug on the asterisk to see whats going on? |
03:59.30 | phsdshft | yes.. but it doesnt seem to be helpful |
04:00.13 | *** join/#asterisk robl^ (n=robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
04:00.23 | benjk | what does it say as contact in the incoming SIP messages from the sipura |
04:01.29 | benjk | also, does the sipura user rport ? |
04:01.36 | benjk | er use rport |
04:01.49 | phsdshft | from, to or via? |
04:02.30 | benjk | well at first when the connection is established, the important field is "contact:" |
04:02.45 | benjk | that tells the other side where to reply to |
04:03.07 | phsdshft | cool now audio works (after restarting the asterisk server.) |
04:03.10 | phsdshft | one side |
04:03.12 | phsdshft | now let me check |
04:03.12 | benjk | so if that is off, then you have no chance of getting a two way converation going betwen the devices |
04:05.24 | benjk | I have had a whole array of problems with SIP over cable here in Japan -- I am being told it's better in the US |
04:05.35 | *** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn) |
04:05.45 | phsdshft | Contact: 2001 <sip:2001@66.189.25.218:212> |
04:06.04 | Qwell[] | kram: evening |
04:06.06 | phsdshft | ok, so it has my public IP |
04:06.21 | benjk | anyway, one of the problems I frequently encountered was that the cable guys seem to drop packets or delay them in a way that the NAT entries time out, that's something to watch out for |
04:06.39 | *** join/#asterisk sudhir492 (n=sudhir@pool-71-114-48-29.washdc.dsl-w.verizon.net) |
04:06.42 | sudhir492 | Hi all |
04:06.42 | Vco | benjk where are you in japan? |
04:06.49 | benjk | Tokyo |
04:07.10 | sudhir492 | Anyone here has pap2-sp2k.bin ? |
04:07.47 | benjk | phsd: is that the address of the linksys? |
04:07.54 | phsdshft | the public ip of it yes |
04:08.40 | benjk | ok, so Asterisk knows where to send the packets off and they are likely to reach the linksys (unless there are troubles on the cable network) |
04:09.22 | benjk | do you have a means to see (at the sipura said) if those packets show up at the sipura? |
04:09.26 | phsdshft | well, I'm getting audio from the asterisk server |
04:09.29 | Vco | any idea if you can use fritz isdn card with ntt? |
04:09.33 | phsdshft | its audio from the sipura to the asterisk server that doesnt work |
04:09.49 | benjk | Vco: Most likely not |
04:10.01 | Vco | thats what i figured..boo |
04:10.04 | benjk | Japanese BRI is different from Euro BRI |
04:10.09 | Vco | **nod** |
04:10.34 | benjk | Vco: Mark Spencer is looking into getting the Japanese Caller ID working for Zaptel |
04:10.49 | Vco | whats the easiest way to get multi line service out there plugged into a server? |
04:10.51 | Vco | like.. |
04:10.51 | benjk | so you could use a TA and use a Zaptel card |
04:10.53 | phsdshft | hmm that depends if this is a switch or a hub on the back of the linksys.. one sec |
04:10.55 | Vco | 6 or so |
04:11.12 | coppice | benjk: do you have any lines with japanese caller ID? |
04:11.28 | benjk | coppice: yes I do |
04:11.47 | coppice | good. I will want someone to test japanese caller ID in openpbx soon |
04:12.00 | benjk | And I also have the English documentation (PDF) in which the NTT line interface is documented (in great detail) |
04:12.11 | Vco | o.O |
04:12.22 | benjk | coppice: just let me know, you can log in here remotely |
04:12.39 | kram | benjk: we already have almost everything |
04:12.45 | coppice | yuo gave me that a long time ago, and I implemented it in spandsp. I just never got a chance to try it out before. |
04:12.52 | benjk | Mark, that's terrific! |
04:12.59 | kram | we already have the demodulator |
04:13.03 | kram | it already supports the mode proposed |
04:13.06 | benjk | cool |
04:13.10 | kram | chan_zap already supports using it |
04:13.27 | kram | the only things that are different are going off hook and the semi-parity bit |
04:13.34 | kram | none of which should be so complex |
04:13.39 | benjk | Vco: you should be looking for a TA and get a Zaptel card ;-) |
04:13.59 | mog_home | man i thought you said semi-party marko |
04:14.03 | mog_home | i was gonna head to office |
04:14.10 | benjk | for 6 lines you could probably use Digium's new tdm2400 card |
04:14.18 | benjk | populate it with 6 FXO modules |
04:14.37 | Qwell[] | multiples of 4 I thought |
04:14.41 | Vco | 6 |
04:14.46 | coppice | the japanese caller ID is really weird, mixing parity and CRC in the same block |
04:14.47 | Qwell[] | oh, 6 modules...right |
04:14.49 | benjk | semi-parity bit |
04:14.59 | benjk | geez that sounds like a paradoxon |
04:15.00 | Vco | i thought it was 6X4 modules |
04:15.04 | kram | i figure it's easiest to run the demodulator without parity turned on |
04:15.09 | kram | then manually do the parity |
04:15.13 | Qwell[] | Vco: 6 modules carrying 4 ports each |
04:15.15 | kram | since it's 7 bits and not 8, it's probably just easier this way |
04:15.32 | kram | we go ahead and run it as an 8-bit modem, and just calculate the 8th bit as parity when it should be and leave it alone when we do the crc-16 |
04:15.33 | Vco | oh..wait..ya |
04:16.03 | benjk | sounds like a plan |
04:16.04 | phsdshft | no its a switch so I cant see if the rtp packets hit the sipura |
04:16.43 | benjk | phsd, how about using some other device on that end first? |
04:17.00 | phsdshft | like a soft phone? |
04:17.00 | benjk | for example a Linux box with a SIP spftphone and Ethereal |
04:17.29 | benjk | then see if you get it working with that since you can see what traffic is coming in and going out |
04:17.38 | benjk | then try the sipura again |
04:18.10 | phsdshft | whats a good sip client for windows (my linux box does not have a microphone, whereas my laptop has one built in)? |
04:18.13 | Vco | know of anyone doing termination/origination over there then? |
04:18.21 | Qwell[] | phsdshft: xlite is decent I hear |
04:18.24 | benjk | Vco: did you mean virtual phone lines? DIDs? |
04:18.54 | Vco | Well, I'm lookign for Osaka DID's for now.. |
04:19.02 | Vco | which hasn't been fun |
04:19.11 | benjk | phsd: X-lite has a debug window that shows you the sip messages but it can be a bitch to configure |
04:19.27 | benjk | Vco: how many do you need? |
04:19.38 | Vco | not many for now. |
04:19.43 | Vco | like less than 10 |
04:19.59 | Vco | maybe 3 to start and test with |
04:20.15 | benjk | The trick is to get multiple parties who will be able to share a PRI |
04:20.27 | Vco | or 1 that can support multiple channels |
04:20.30 | phsdshft | benjk: I can use windump while using a sip phone probably |
04:21.11 | benjk | phsd: wouldn't know about that, I don't use Windoze, but yeah, use whatever you are familiar with |
04:22.25 | benjk | Vco: if we can find three parties who are interested in Osaka virtual phone numbers or DIDs we can set up a PRI there |
04:22.49 | Vco | need a number or two for inbound dialing to canada, and sounds like some guy that has 11 locations for his shop around osaka that wants to possibly do some |
04:22.59 | Vco | business |
04:23.09 | Vco | inlaw are still over there... |
04:23.16 | benjk | Vco: where are you located? |
04:23.20 | Vco | Canada |
04:23.26 | benjk | Oh I see |
04:23.28 | Vco | Inlaws are still over there |
04:23.38 | Vco | aparantly pimping my skills to their customers |
04:23.40 | benjk | ok |
04:23.47 | benjk | :-) |
04:23.50 | Vco | heh..heh. |
04:24.06 | benjk | well, if you need somebody to do ground work at location, let me know |
04:24.11 | Vco | nishnomia city around there.. |
04:24.15 | Vco | will do.. |
04:24.21 | Vco | saves me a 18hr flight |
04:24.25 | ritesha | Realtime mapping for 'queues' found to engine 'mysql', but the engine is not available....anyone knows what's going on? |
04:24.25 | benjk | benjamin at sunrise-tel dot com |
04:24.32 | Vco | ahhhhhhhh |
04:24.46 | Qwell[] | ritesha: Did you setup the mysql engine? |
04:24.47 | Vco | a familiar domain name |
04:24.49 | Vco | :) |
04:24.55 | benjk | :-) |
04:25.10 | ritesha | hmm. what do I need to setup the mysql engine |
04:25.20 | ritesha | i started the mysqld service |
04:25.21 | Qwell[] | ritesha: mysql |
04:25.27 | ritesha | isn't that enough? |
04:25.57 | Qwell[] | You need to create a database for asterisk, create the tables, create users, setup the configs in * to know all of this information |
04:25.57 | benjk | Vco: the telco I am working with here in Tokyo has datacentres in Osaka too |
04:26.24 | benjk | they can provide PRI in Osaka and they are higher quality and lower price than NTT |
04:26.46 | ritesha | i did all the setup. but the engine is not available is something of a trouble. |
04:27.21 | ritesha | if it's somethign to do with the * setup, I could look again. Else I could check for the mysql gotchas? |
04:28.18 | benjk | Vco: still there? |
04:28.29 | Vco | yo |
04:28.47 | Vco | is kddi only for international? |
04:28.53 | benjk | In the short term, your client could get a VoIP service from OCN |
04:29.13 | benjk | I know how to hook Asterisk up to that directly, without any adapters ;-) |
04:29.22 | Vco | yea, been having fun navigating for info from here... |
04:29.23 | benjk | officially you cant do that |
04:29.51 | Vco | or getting my wife to explain to my father in law to get info from him about what he pays etc |
04:30.15 | benjk | kddi used to be internationaly only but I think they now do long distance too |
04:30.47 | Vco | i had a link from something i thought was ntt west, for what sounded like voip kinda stuff.. |
04:31.31 | benjk | Anyway, if you get ADSL from OCN on a single NTT analog line, then you can have their bundled VoIP service for about 500 yen per month |
04:31.43 | benjk | this gets you a 050 telephone number |
04:31.52 | benjk | ]which can be called from the PSTN |
04:32.17 | benjk | the cost of calling a 050 number from PSTN lines is 10.5 yen per 3 mins |
04:32.19 | Vco | inlaws have a yahoobb line upsatirs at the house, but down in the salon they have a business line |
04:32.30 | benjk | YahooBB sucks |
04:32.34 | Vco | tell me about it |
04:32.45 | benjk | false hangups all the time |
04:32.51 | benjk | bad line noise |
04:32.56 | Vco | had 3 modems go tits up |
04:32.56 | benjk | bad sound |
04:33.01 | benjk | yep |
04:33.20 | benjk | and you cannot ever hope to get Asterisk working with that directly |
04:33.34 | benjk | always need to go through the analog port |
04:33.37 | Vco | yea, thats why i was wondering about isdn |
04:33.44 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
04:33.44 | Vco | i think thats what they have downstairs |
04:33.53 | benjk | No, get ADSL from OCN on their business line |
04:33.54 | coppice | what is it with broadband modems? their reliability makes windows look solid, and all makes seem the same |
04:33.54 | Vco | is it net64 or something |
04:33.59 | benjk | then cancel YahooBB |
04:34.08 | Vco | ocn eh?? have an english link? |
04:34.22 | Vco | meh, i'll google |
04:34.29 | benjk | OCN is the Internet service of NTT COmmunications |
04:34.52 | benjk | Your inlaws will know how to sign up for OCN |
04:34.58 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167049176.nb.aliant.net) |
04:35.18 | benjk | just go to an electroncis store, and they have the form |
04:35.45 | Vco | or any streetcorn ein umeda on a saturday afternoon in the summer ;) |
04:35.52 | benjk | yep |
04:35.54 | Vco | streetcorner in ... |
04:35.56 | Vco | even |
04:36.07 | benjk | OCN is about on par with YahooBB now |
04:37.06 | Vco | now..inbound calls are free on landling too? |
04:37.11 | Vco | or just mobiles? |
04:37.13 | benjk | many smaller ISPs who think they are too small to afford their own VoIP service (because they only know NEC and Fujitsu and have never heard of Asterisk) resell OCN service under their own name without making any profit (just so their customers dont churn) |
04:37.33 | Vco | or does it depend on plan/carrier/time of day/prefecture/populatin density in the area |
04:37.48 | Vco | and a million other factors for determining call rates there.. |
04:37.49 | benjk | inbound calls are alwasy free |
04:37.51 | Vco | insane |
04:38.06 | benjk | unless you have a toll-free number of course |
04:38.45 | benjk | in Japan VoIP service is always bundled with your ISP |
04:38.59 | benjk | if you want to change, you have to switch your ISP |
04:39.15 | phsdshft | cool.. I fixed my voip problem |
04:39.27 | benjk | phsd: what was the trouble? |
04:39.42 | Vco | wakatta |
04:40.56 | phsdshft | benjk: I changed a bunch of things at once... but... part of it was I had dtmf set to inband instead of rfc.... so, the way I was testing if audio got from me to the asterisk server was flawed.. so it might have worked correctly (except dtmf) after I upgraded the asterisk server code.. |
04:41.37 | phsdshft | I had upgraded the firmware on the sipura, the version of asterisk on the server side, reset the sipura to defaults and reconfigured it and disabled 4 options in the nat support parameters on the sipura |
04:41.45 | phsdshft | so one of those fixed it heh |
04:42.00 | Vco | what about for solid internet connection there? dsl/fiber? |
04:42.07 | Vco | fibre |
04:42.16 | Vco | whatever |
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04:43.08 | *** part/#asterisk mcadory (n=mcadory@208-149-64-246.adsl.nexband.com) |
04:43.26 | *** part/#asterisk shmooz (n=shmooz@H142.C72.B0.tor.eicat.ca) |
04:43.51 | benjk | Vco: FTTH is best of course |
04:43.59 | Vco | :) |
04:44.08 | Vco | as far as price/perf |
04:44.10 | benjk | and the best value for money you get from USEN |
04:44.44 | benjk | about 50 USD per month for 100Mbit full duplex with a block of 8 ip addresses included |
04:44.51 | Vco | shit |
04:44.52 | benjk | which is a steal |
04:44.55 | Vco | ya |
04:45.05 | Vco | last i heard it was still $85-$100ish |
04:45.08 | benjk | because in Japan ip addresses are worth more than diamonds |
04:45.38 | benjk | on OCN if you get fibre, you pay 350 USD per month just for the IP addresses |
04:46.27 | Vco | maybe when i'm off this contract i 'll need to head back over there instead of going snowboarding.. :/ |
04:46.33 | benjk | Vco: it is still 85-100 with other providers |
04:46.48 | benjk | haha |
04:47.13 | Vco | haven't been over since spring '04 |
04:48.31 | benjk | BTW, OCN have a business VoIP paclkage |
04:48.50 | benjk | 4 lines on a single ADSL circuit |
04:48.53 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
04:49.02 | Vco | hmm.... |
04:49.17 | benjk | so you could get four 050 numbers directly into Asterisk |
04:49.31 | Vco | interesting, i have a spave tdm400 card too |
04:49.38 | benjk | or a single number with 4 channels |
04:49.43 | Vco | **nod** |
04:49.46 | benjk | you don't need it |
04:49.57 | Vco | ahh. |
04:50.03 | benjk | as I said, I can make Asterisk connect to OCN directly |
04:50.05 | PakiPenguin | Vco: send it to me :p |
04:50.07 | Vco | yea.. |
04:50.23 | benjk | officially you cannot do that |
04:50.35 | benjk | but I have it working here ;-) |
04:50.40 | Vco | cool. |
04:50.49 | benjk | works very nicely |
04:50.53 | benjk | G711 |
04:50.58 | benjk | and caller ID |
04:51.04 | benjk | crisp sound |
04:51.26 | benjk | not like YahooBB, which sounds like your'e calling somewhere in Africa |
04:52.21 | Vco | Wife is pretty active in the cultural center here, so she knows a fair number of people from her prefecture, want to get a local inbound number there, since a lot of other people there we also know only use a mobile phone.. |
04:52.28 | Vco | so calling here is a pain in the butt |
04:52.35 | h3x | with a bluebox, and a operator line |
04:52.35 | h3x | heh |
04:53.59 | Vco | i guess DID's are the only real need tho.. |
04:54.03 | phsdshft | I'm getting a 404 not found error when doing a sip debug for calls incoming to the sipura (outbound work fine)... and the asterisk server reports 'everyone is busy/congested' ... the line I have in my extensions.conf is: exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@2001,30,r) |
04:54.06 | phsdshft | any ideas? |
04:55.05 | Vco | calling there is 3.5/min without looking for the cheapest possible rates around, so i don' tknow if there is a grounbreaking savings by having a local number there |
04:55.17 | Vco | at least on the scale we'd be using it inititally |
04:57.14 | benjk | four inbound you will find that most people now either user a mobile, or they will use a VoIP line |
04:57.47 | benjk | if they are on YahooBB, then the call to your 050 OCN number will be 10.5 yen per 3 mins like it is on NTT analog lines |
04:57.49 | phsdshft | benjk: any ideas? |
04:58.13 | Vco | what about toll free stuff there? |
04:58.14 | benjk | if they are on any other VoIP service, then their call to your 050 number will be FREE |
04:58.40 | benjk | toll-free is prohibitive |
04:59.12 | benjk | last time I checked on toll-free you had to pay for inbound long distance charges |
04:59.33 | phsdshft | my toll free is flat rate per minute ... higher intrastate than interstate |
04:59.56 | Vco | and then there's calling mobiles as well.. |
05:01.14 | benjk | they best way to do toll free in Japan, maybe you should get 1x YahooBB and connect via Zaptel to the Yahoo modem/ata, and 1x OCN connected directly over SIP, then you tell your callers to call the 050 number of your YahooBB line if they are on YahooBB (free) and the others you tell to call you on your OCN 050 number (free) |
05:01.35 | Vco | ya.. |
05:02.00 | Vco | been dealing with a company, totally insane setup right now.. |
05:02.03 | Vco | ok.. |
05:02.09 | benjk | unfortunately there is no way to tell by the 050 number if it is YahooBB or OCN (and allied services) |
05:02.23 | Vco | they have offices in Toyko, Hawaii, and Sydney |
05:02.46 | phsdshft | I'm getting a 404 not found error when doing a sip debug for calls incoming to the sipura (outbound work fine)... and the asterisk server reports 'everyone is busy/congested' ... the line I have in my extensions.conf is: exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@2001,30,r) |
05:04.03 | Vco | most of the calls in hawaii are to tokyo, they call the customer, but for the customer to call them back is overseas, so they leave the tokyo number, which may or may not still be long distance for the caller, the customer calls back, the receptionist asnwers, ....instant messages the person in Hawaii, and make arrangments for the person in Hawaii to call backteh person in japan right away |
05:04.19 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
05:04.39 | Vco | all i could ask them when they explained this to me was |
05:04.43 | Vco | "WHY?" |
05:04.45 | benjk | phsd: did you configure inbound on your sipura? those sipuras can be a bitch to get set up right |
05:05.10 | phsdshft | I /think/ so... |
05:05.51 | benjk | phsd: again, back to the softphone refernece ;-) |
05:06.12 | phsdshft | ben: I got it working for outbound calls now... so just inbound /calls/ (not audio) are a problem now |
05:06.15 | benjk | Vco: sounds like a perfectly sound Japanese business process |
05:06.23 | Vco | yup..heh. |
05:06.31 | phsdshft | so I have two way audio now... inbound calls dont work at all at the moment and I think its likely the way my asterisk config or something is |
05:08.17 | benjk | Vco: set up an OCN ADSL/VoIP in their Tokyo office, route that 050 to Hawaii, problem solved |
05:08.26 | Vco | thats what i'm looking at |
05:09.22 | Vco | they do a lot of inter-office as well, so be likely doing 2 severs over iax or the like etc |
05:09.29 | Vco | servers |
05:09.43 | Vco | goddammit |
05:09.56 | Vco | whoa..i guess there is a lot crud stuck in my keyboard |
05:10.08 | benjk | yes, and perhaps replace their office telephone system along with it ;-) |
05:10.17 | Vco | yup |
05:10.51 | Vco | the kinda of setup they have can't go anywhere but up quite frankly.. |
05:11.05 | benjk | as Mark is now fixing the Japanese Caller ID, it will be possible to actually deploy some Asterisk servers as PBXes in Japan |
05:11.40 | coppice | benjk: do you still face serious permission to connect issues, or is japan loosing up? |
05:11.59 | phsdshft | doh... the dial line needed to be userid@contextinsip.conf not number@contextinsip.conf |
05:13.04 | benjk | coppice: I have cultivated a relationship with JATE, been there many times for meetings, explained Asterisk to them and the Zaptel stuff |
05:13.42 | benjk | I am also familiar with the process and what is needed to fill out the (immense) paperwork |
05:14.19 | benjk | I also got a Japanese engineer who I can call upon to help with preparing some of the documents |
05:14.51 | benjk | so, I feel confident to take on the task of getting JATE approval, at least for digital |
05:15.01 | coppice | japanese seem to like paperwork. buying a train ticket actually gets you several which machines along the way swallow one by one. weird |
05:15.05 | benjk | analog is more elaborate |
05:15.24 | Vco | it's the same ticket |
05:15.41 | Vco | it spits out the other end |
05:16.27 | benjk | coppice: the type approval does cost a bit of money though |
05:16.31 | coppice | nope. I had a two step journey. I got three tickets. I had to put all three through each machine, and less came out each time |
05:17.05 | benjk | haha that's because you took the Shinkansen |
05:17.10 | Vco | heh..heh. |
05:17.20 | benjk | one ticket is for the distance (basic fee) |
05:17.33 | Vco | "zones" |
05:17.34 | benjk | the other is for the express fare |
05:17.45 | benjk | a surcharge so to speak |
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05:17.55 | Vco | "luxury tax" |
05:17.55 | benjk | and the third one is for the Shinkansen |
05:18.03 | benjk | another surcharge |
05:18.10 | coppice | the two tickets for the stages had prices which didn't add up to the overall price on the third ticket. i never figured that one out |
05:18.26 | benjk | and you had to use the shinkansen ticket to get into the shinkansen area of the station |
05:18.39 | Vco | *sigh* |
05:18.50 | Vco | i want to go back |
05:18.59 | h3x | them japanese must have been hanging out with the us government to come up with that shit |
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05:19.51 | benjk | h3x: trust me, the Japanese need nobody's help to come up with their weird processes |
05:20.12 | h3x | so, you guys use T1s, and is it ulaw? |
05:21.46 | benjk | what in Japan? |
05:21.52 | h3x | yeah |
05:21.53 | benjk | we don;t use T1s |
05:21.58 | benjk | we use J1s |
05:22.01 | h3x | i thought it was north america, |
05:22.02 | h3x | oh |
05:22.19 | h3x | but isnt it electrically the same as t1? |
05:22.28 | benjk | that's why those interface cards (some of them) are branded T1/E1/J1 |
05:22.44 | benjk | it is a futzed up T1 |
05:22.48 | h3x | right |
05:22.52 | coppice | its electrically the same, its just twisted for incompatibility |
05:23.01 | h3x | but it is ulaw right? |
05:23.16 | coppice | they got that into Taiwan too, since NEC basically build their early digital networks |
05:23.27 | h3x | i thought all that MFC R2 signalling did a good enough job of fucking up compatibility |
05:23.42 | h3x | on E1s |
05:23.47 | coppice | they don't use R2 |
05:24.10 | h3x | yea i know but seeing as ericcson thought it would be a great idea to come up with a different R2 in each country |
05:24.25 | h3x | im suprised you had T1 influence there |
05:24.30 | coppice | that was nothing to do with ericsson |
05:25.50 | coppice | actually, just one form of R2 in each country would be an improvement :-) |
05:25.57 | h3x | haha |
05:26.13 | h3x | man i cant believe aculab had the balls to sell me a $13k card in 2000 |
05:26.19 | h3x | knowing i was using sun solaris |
05:26.27 | h3x | and they didnt even have any ulaw algorithyms working yet |
05:26.33 | h3x | on the DSPs |
05:26.35 | h3x | for like 2 years |
05:26.44 | h3x | i think they were counting on me not getting along that far |
05:26.47 | h3x | before them . hah |
05:27.16 | h3x | but whats even more balsy |
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05:27.27 | Vco | bah.. |
05:27.39 | coppice | aculab keep far more of their customers happy than the competitors do :-) |
05:27.50 | h3x | is their "asterisk driver". all it does is nails up channels to record and playback |
05:27.54 | h3x | like the dialogic drivers |
05:28.04 | *** join/#asterisk alphaque (n=alphaque@218.208.239.119) |
05:28.07 | h3x | aculab is doing terrible in the US |
05:28.12 | h3x | has always been |
05:28.20 | alphaque | they seem to be doing ok in oz |
05:28.23 | alphaque | or so i hear |
05:28.43 | h3x | they actually engineered the stuff well but they have too many parallel development projects that arent compatible with each other |
05:30.46 | h3x | like, i was fortunate enough to start with the TiNG drivers on solaris, which was ported to linux eventually |
05:31.00 | h3x | the previous generation drivers died off on os/2, sco, linux, etc. |
05:31.07 | h3x | then TiNG took over |
05:31.22 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
05:32.03 | h3x | They have TERRIBLE code samples |
05:32.11 | h3x | im glad its all ANSI C though |
05:32.33 | h3x | but i guess it dosent matter anymore |
05:32.46 | h3x | coz the only thing it does better than asterisk is fax and asr/tts. |
05:32.57 | h3x | but not for long |
05:33.29 | coppice | why not for long? |
05:33.53 | h3x | we have faith in your t.38 and spandsp skills :D |
05:33.59 | sudhir492 | exit |
05:34.14 | coppice | that doesn't address 2 out of the 3 things you listed |
05:34.14 | h3x | and i donno, im sure cepstreal has something cooking, ... |
05:34.23 | sudhir492 | ooops, sorry. wrong window |
05:34.43 | h3x | Well, actually you are best off with some type of HMP with nuance or scansoft stuff |
05:35.03 | coppice | TTS is easy to cover. the asr part is the problem |
05:35.09 | h3x | yeah |
05:35.23 | h3x | i mean, the aculab boards dont do anything for asr/tts besides playback/record |
05:35.48 | h3x | they do have a "free" asr and tts, but the asr connected word recognition cant do that big of a vocabulary at once |
05:36.47 | coppice | I think they do V.34 fax now, which * will not do |
05:38.03 | h3x | yeah well they seem to have it togehter for faxes but those cards cost way too much to use just for faxing |
05:38.07 | *** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw) |
05:38.23 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
05:38.31 | h3x | although |
05:38.39 | h3x | the new voip board can do 500 transmits on ethernet |
05:38.42 | h3x | and its like 13 grand |
05:38.50 | h3x | so thats reasonable |
05:39.00 | h3x | as g.711 of course |
05:39.03 | coppice | they provide cheap SS7 too |
05:39.31 | h3x | it has a lot of limitations |
05:39.42 | h3x | tcap is terrible, you may as well start from scratch |
05:40.24 | h3x | i dont remember if its still limited to 8 T1/E1s per linkset or not |
05:40.30 | h3x | for ISUP |
05:40.32 | coppice | really? I don't know anyone who has used it, but I thought their SS7 was supposed to be very good. |
05:40.57 | h3x | it is, but its really low level. although ISUP is the same general interface as doing any other CAS/CCS signalling |
05:41.32 | h3x | Oh, I know. ANSI ISUP is non-existant |
05:41.46 | h3x | that pretty much brought all my ideas for it to a screeching halt |
05:49.28 | h3x | using sangoma for ss7 is probably cheaper and a smarter thing to do |
05:49.49 | phsdshft | anyone doing distinctive ring with the sipuras? |
05:49.57 | phsdshft | (and asterisk) |
06:17.46 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
06:25.06 | Dr_Ray | why not post to the asterisk-biz mailing list |
06:34.02 | *** join/#asterisk mrgoby (n=knoppix@pcp05307400pcs.wanarb01.mi.comcast.net) |
06:39.14 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-236-213.37-151.net24.it) |
06:41.55 | h3x | <PROTECTED> |
06:41.56 | h3x | heh |
06:44.33 | *** join/#asterisk lorinc (n=ang@caracas-0247.adsl.interware.hu) |
06:48.28 | benjk | my ignore didn't seem to have worked |
06:48.38 | benjk | otherwise I wouldn't have seem his post |
06:48.43 | benjk | :( |
06:49.10 | h3x | <PROTECTED> |
06:49.17 | benjk | ah |
06:49.30 | benjk | thanks |
06:49.49 | Dr_Ray | I regulated Katty to the ether long ago |
06:50.07 | benjk | bbz is Katty? |
06:50.11 | Dr_Ray | no |
06:51.00 | benjk | still I think bbz should be banned from freenode |
06:51.12 | Dr_Ray | why? |
06:51.30 | benjk | because its very bad ettiquette |
06:51.47 | benjk | and its automated |
06:52.00 | h3x | dude hes been spamming every hour for a week |
06:52.03 | Dr_Ray | freenode/irc would be empty based on ettiyuette |
06:52.03 | h3x | its like the guy went on vacation |
06:52.23 | benjk | if somebody did this while personally around and participating, well I guess it would still be acceptable |
06:52.30 | *** join/#asterisk diamon (n=diamon@c-66-176-91-189.hsd1.fl.comcast.net) |
06:52.54 | Dr_Ray | I did not know it was automated |
06:53.04 | Dr_Ray | hence my suggestion to take it to the biz mailing list |
06:53.06 | benjk | h3x: that's why I said its automated, its a bot, a spam bot |
06:53.20 | benjk | spam bots should be banned |
06:53.46 | diamon | Ok, I've finally got some time (courtesy the holiday!) to get * working. Is there any suggested distro of linux to use with it? My two main prefs are Fedora and Slackware. |
06:54.01 | Dr_Ray | I use fedora |
06:54.17 | benjk | whatever distro you are familiar with works best |
06:54.22 | diamon | Is there an rpm for it, or did you build it? |
06:54.24 | benjk | usually, that is |
06:54.29 | Dr_Ray | I build it from cvs |
06:54.32 | Dr_Ray | it's ez |
06:54.44 | *** join/#asterisk tessier_ (n=treed@wsip-68-15-4-13.sd.sd.cox.net) |
06:54.56 | Dr_Ray | if you search teh wiki for fedora you will find all the install help you need |
06:55.09 | benjk | pretty much every serious astmaster builds his asterisk himself |
06:55.14 | diamon | DR: Yeah, compiling is easy, I was just trying to get myself used to using RPM's whenever I can. It makes yum updates less likely to break things by accident. |
06:55.19 | h3x | well it might be a regular irc client |
06:55.21 | h3x | with a timer |
06:55.38 | diamon | benjk: Ideally, I'd build my own rpm for a FC4 system. :) |
06:55.48 | benjk | h3x: in my book that still counts as a spam bot |
06:55.55 | Dr_Ray | you can, but asterisk is a moving target |
06:55.55 | diamon | But anything that works is good enough for me. |
06:56.20 | benjk | diamon: sure, I build my own installer for MacOSX |
06:56.50 | diamon | Dr: I don't upgrade unless there's a security problem or a bug I'm impacted by, or a feature I feel I must have... Learned my lesson about arbitrary upgrades on my MythTV system. Now it actually works, and keeps working even. |
06:57.02 | Dr_Ray | diamon - that's how I do it |
06:57.17 | benjk | same here, my PBX still runs a version of Asterisk that is more than 1 year old |
06:57.25 | benjk | if it aint broken don;t fix it |
06:57.28 | Dr_Ray | my asterisk is from june |
06:57.35 | diamon | Though now that I remember, I'll have to hit #mythtv-users and ask about disabling the auto-eat-all-my-crap feature... |
06:57.39 | Dr_Ray | since it's off the internet, I'm not super concerned |
06:57.51 | benjk | but I have plenty of other asterisks to play with newer stuff |
06:58.27 | Dr_Ray | there are asterisk rpm's if you want, but from cvs is just simple |
06:59.31 | diamon | Ok, so here's what I wanted to do in general... I've got an X100p Digium card clone which seems to be doing well enough so far. I want * to answer my POTS, but to do mixed VoIP and POTS answering. I'd think that's a common setup, but my thoughts and reality often don't mix well. :) |
06:59.55 | benjk | there is no such thing as a X100P Digium card clone |
07:00.07 | diamon | Dr: The ones I saw were 1.0 of *, but I'll keep looking for a 1.2. Seems to be a lot of features and functionality in 1.2 that I'd like to play with eventually. |
07:00.12 | IronHelix | yes there is, the x100 is just a rebadged intel voice modem |
07:00.30 | IronHelix | if you buy the same kind of voice modem with the right chipset, its a 'x100 clone' |
07:00.31 | benjk | Diogium's own cards were rebadged Intel modems |
07:00.36 | diamon | IronHelix: Sure of that? It's an intel modem? |
07:00.43 | Dr_Ray | it's not a clone |
07:00.46 | IronHelix | yeah, the x100 is |
07:00.48 | mog_home | hey now |
07:00.55 | mog_home | we made a small change ^_^ |
07:00.58 | IronHelix | diamon- if you do that, i recommend connecting your other pots phones to asterisk too |
07:01.06 | mog_home | and we never claimed they were anything else |
07:01.06 | diamon | Bwahahaaaa! I'd been trying for years to get the command set for voice modems to do more-or-less what this card will now do. Irony... |
07:01.15 | IronHelix | hehehe |
07:01.17 | benjk | so if you say that some card is a clone of Digium's card than that implies that Digium's card is an original, which it is not |
07:01.24 | IronHelix | no definately not bumping your products Mog <3 |
07:01.30 | mog_home | well |
07:01.33 | mog_home | benjk |
07:01.36 | mog_home | digium wrote the driver |
07:01.46 | mog_home | thanks iron its all good |
07:01.46 | benjk | Yes, the driver is original |
07:01.51 | diamon | Iron: I don't have the card to provide ring (FXS?) for my lines, nor an ATA. |
07:01.52 | benjk | but the card is vanilla |
07:02.03 | mog_home | well driver is 99% of the work in this case |
07:02.10 | mog_home | and in most cases regarding zaptel hw |
07:02.12 | IronHelix | diamon- i would recommend getting that, either an ata or some fxs ports |
07:02.16 | IronHelix | it will be much more useful |
07:02.17 | mog_home | the genius is in the code not the card |
07:02.23 | IronHelix | very true |
07:02.27 | benjk | mog, nobody disputes that |
07:02.28 | Dr_Ray | the genius is asterisk |
07:02.28 | IronHelix | the card is just a dumb interface |
07:02.32 | IronHelix | and zaptel |
07:02.32 | Dr_Ray | not the zaptel driver |
07:02.43 | diamon | Is there any kind of cheap FXS card, or should I just suck it up and get an ATA? |
07:02.43 | mog_home | i dont dr_ray |
07:02.46 | IronHelix | the problem is a mixed setup diamon, if you pick up another phone it will put you on the line with * |
07:02.47 | mog_home | there is a lot of code in zaptel |
07:02.52 | mog_home | that gets over looked |
07:02.56 | IronHelix | diamon- depends on your definition of cheap |
07:03.05 | benjk | diamon, get the real thing |
07:03.06 | IronHelix | digium tdm400 series is nice |
07:03.10 | IronHelix | pretty easy to setup too |
07:03.10 | diamon | I just want to do a proof-of-concept type thing for myself, and maybe a client/friend or two of mine who might get good use out of this... |
07:03.12 | benjk | get an IP phone |
07:03.14 | mog_home | like asterisk is one of the few pbxes that can generate and understand rotery |
07:03.18 | benjk | analog sucks balls |
07:03.42 | diamon | Iron: Those looked nice, but aren't they like $400USD? |
07:03.43 | IronHelix | diamon are you trying to make a really cool answering machine or do something more complicated? |
07:03.51 | IronHelix | $400 fully loaded with ports |
07:03.52 | benjk | proof of concept use phones: Grandstram BT100 for 50$ |
07:04.07 | IronHelix | you can get one that has only 1 port for like $200 |
07:04.17 | IronHelix | yeah benjk might have a better idea |
07:04.17 | diamon | Iron: More complicated eventually, for now I want an uber-spiff answering system with some PBX-type features. |
07:04.21 | Dr_Ray | my proof of concept was the asterisk dev kit (back when it included a x100 card) and a bugetone |
07:04.30 | benjk | you can get 4 Grandstream IP phones for 200 $ |
07:04.40 | IronHelix | grandstream bt100 barbietel phones are cheap and they wrok well |
07:04.48 | IronHelix | plus if you forward your ports and configure * correctly |
07:04.52 | IronHelix | you can demo it from your friends house |
07:04.55 | *** join/#asterisk rustyPixel (n=rustyPix@cpe-024-162-252-094.nc.res.rr.com) |
07:04.56 | IronHelix | just plug in the phone and go |
07:05.15 | IronHelix | without bringing the whole server |
07:05.17 | Dr_Ray | I'd not marry myself to 4 budgetones |
07:05.29 | benjk | However, if you must do FXS for something like -say- a cordless base, then you may want to get an IAXy |
07:06.08 | Dr_Ray | I was happy with the budgetone for testing, but not an everyday phone |
07:06.10 | IronHelix | or get that new uniden thingy |
07:06.12 | benjk | but FXS ports are just dumb |
07:06.20 | IronHelix | i agree, when possible |
07:06.22 | diamon | benjk: I'll probably just get some kind of ATA and hook it to the whole house, I don't have much inside, but I'd prefer not to tear everything up. :) |
07:06.32 | IronHelix | diamon- some tearing will be useful |
07:06.40 | IronHelix | if you tear enough that each phone gets its own fxs port |
07:06.45 | IronHelix | then you can dial from one phone to another |
07:06.45 | benjk | unless you have a client who is stubbornly braindead and demands analog phones |
07:06.49 | diamon | Iron: I'll preserve tearing up for when I try for IVR. :) |
07:07.09 | benjk | or a client with thousands of analog phones that havent been written off yet, don't use FXS |
07:07.10 | IronHelix | benjk- or has no cat5 |
07:07.21 | IronHelix | for a small one at least |
07:07.42 | IronHelix | id much rather install pots channels than get out the drill and run cat5 through walls |
07:08.04 | benjk | Iron: that no cat5 thing falls under stubbornly braindead client |
07:08.05 | diamon | benjk: I choose my clients carefully. If I suggest it, they know to get exactly that or I don't support it at all, and whatever I suggest that they get, I support as best I can at no charge for simple stuff. My usual rates cover a bit of support, and most of them are bright folks. |
07:08.08 | Dr_Ray | we've been very happy with channel banks and zap |
07:08.48 | benjk | diamon: then do yourself and your clients a big favour and get IP phones |
07:09.00 | IronHelix | agreed |
07:09.03 | benjk | there is plenty of decent IP phones to choose from |
07:09.06 | diamon | The main client I was thinking of had hell trying to get a Cisco setup in and working, but best I could tell it was the moron (read: nepotism!) that the CFO hired, some friend of his or something. |
07:09.07 | IronHelix | diamon- at the very least get ONE barbietel and play with it |
07:09.29 | mog_home | spas are great too |
07:09.31 | diamon | Iron: You'd suggest a Barbitel over an ATA? |
07:09.32 | mog_home | no barbietel |
07:09.50 | IronHelix | depends on the use. to play with and learn i'd suggest a barbietel any day of the week |
07:10.04 | diamon | Ok, I'll consider that. |
07:10.04 | IronHelix | if you need to be using it constantly AND you have a good pots phone, then get the ata |
07:10.04 | benjk | diamon: if they want Cisco phones (cause they impress visitors) then give them Cisco phones, Asterisk can handle those |
07:10.14 | Dr_Ray | we drive 7960's |
07:10.19 | Dr_Ray | via asterisk |
07:10.21 | mog_home | other than a cisco or polycom, id rather have something like a spa or iaxy hooked into my 900mhz nice phone |
07:10.22 | IronHelix | yeah from what i hear asterisk has made a bunch of progress with chan_sccp |
07:10.26 | mog_home | only thing that sucks is cid |
07:10.35 | mog_home | but i get that from jabber these days anyways |
07:10.44 | diamon | benjk: Yeah, I told them to get the unlock access, so we're good there, AFAICT the guy is a buffoon who has nearly no idea what he's doing. Even the wiring was screwy... |
07:11.16 | benjk | Oh BTW, did you see that the Grandstream GXP-2000 has PoE ??? |
07:11.23 | benjk | I was baffled by that |
07:11.43 | mog_home | isnt that phone a legitament one |
07:11.49 | mog_home | not like the barbies |
07:11.52 | IronHelix | hehe i know the type. i once came on a job where some kid was trying to wire stuff up, he was stripping the cat5 wire before crimping it (i mean like stripping the individual wires). his first Q to me was why the switch was broken (slow steady flash on all lights = bad cable) |
07:11.56 | benjk | legitament? |
07:12.01 | IronHelix | yeah the gxp is pretty good |
07:12.05 | IronHelix | i use one on my desk |
07:12.10 | IronHelix | with the new firmware its actually pretty cool |
07:12.12 | mog_home | Good |
07:12.14 | mog_home | i thought those were quality ones |
07:12.19 | mog_home | unlike the barbie crap |
07:12.28 | IronHelix | in another two months or so it will be awesome once they get some more bugs/features in |
07:12.28 | diamon | They laid down a second network just to be totally sure of available bandwidth (they do have brutal usage spikes at times), so that was sensible, but that moron wired it odd, and cross-linked stuff, and didn't set up anything... Very odd. |
07:12.41 | diamon | Iron: Arrgh, I hade that kind of stuff... |
07:13.22 | mog_home | man any of yall watch late night tv |
07:13.37 | IronHelix | hell no |
07:13.39 | IronHelix | get a tivo |
07:13.39 | mog_home | they have the wierdest stuff on |
07:13.47 | mog_home | heh |
07:13.49 | mog_home | i dont get cable |
07:13.55 | benjk | mog, our late night tv is your early bird morning program |
07:13.58 | mog_home | i just was bored this week |
07:14.04 | mog_home | so i flipped it on |
07:14.07 | mog_home | heh |
07:14.11 | IronHelix | yeah, once i explained to him that having the wires touch each other inside the thingy is bad, hes like OHHHHHH that explains it, goes and takes a huge scissors and cuts off the last ~50 cable ends that he did |
07:14.13 | mog_home | this guy is selling knives |
07:14.24 | mog_home | and he is acting like it is the most amazing thing ever |
07:14.30 | IronHelix | cutco? |
07:14.35 | mog_home | oh no |
07:14.40 | mog_home | this guy is a dealer |
07:14.41 | diamon | Iron: Though that reminds me... Once I had a call from a client who had a LED they didn't recognize turning off in flickers on their network switches. I went up to take a look. It wasn't a switch, it was an 8-hub stack, daisy-chained together. The light that was turning off? COL. |
07:14.48 | mog_home | and he sells all different kinds of knives |
07:14.54 | mog_home | i could buy 200 knives |
07:15.00 | mog_home | at 99cents a knife |
07:15.03 | benjk | the guy who messed up the wires is selling knoves on tv? |
07:15.05 | mog_home | and that includes a sword |
07:15.35 | mog_home | hrm? |
07:15.42 | mog_home | oh noes |
07:15.48 | mog_home | only 1 minute to buy the set |
07:15.50 | diamon | I was like, "Uh, that's only supposed to flicker on." They got worried when it was flickering *off*! I split it up with an 8-port 10/100 switch attached to the rack, one hub per switch. Wow did that fix a lot of slowness... |
07:16.04 | IronHelix | hahahahahahaha |
07:16.47 | benjk | mog, in Taiwan, they make famous knives of the steel from Chinese mortar shells that the Chinese fired at them over the years from the mainland across Taiwan straits |
07:17.15 | benjk | and they then export those knives to China, they are a big hit in Chine |
07:17.19 | diamon | I countead heads after that (new client) and found I had a collision domain of over 300 systems, including all the servers. Nothing was switched. I had a lot of work to do for that place for sure. Good money and they were thrilled with the results, so I was happy. |
07:17.28 | mog_home | heh |
07:17.33 | mog_home | I just dont get it |
07:17.36 | benjk | they even come with a certificate that they are made from steel from China |
07:17.38 | mog_home | who buys knives off tv |
07:17.51 | diamon | benjk: Wow, that's funny! I'd call it IRONic, but hey... |
07:17.51 | mog_home | this show is on every night 5 nights a week |
07:17.53 | mog_home | for an hour |
07:17.55 | benjk | mog, people who are bored |
07:18.06 | mog_home | how can you have 5 hours a week of knives |
07:18.20 | diamon | mog: There's a lot of bored people out there... |
07:18.25 | mog_home | yeah |
07:18.28 | mog_home | i mean im watching it |
07:18.35 | mog_home | but its a far cry from buying |
07:18.37 | benjk | shame on you |
07:19.00 | benjk | wait another two weeks and you'll start buying ;-) |
07:19.05 | diamon | Iron: So the uber-answering-rig I was asking about is no big deal then overall? |
07:19.15 | mog_home | nah i only am watching it as i am frustrated with my code |
07:19.20 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-45-84.dsl.irvnca.pacbell.net) |
07:19.24 | mog_home | and i have given up for the night |
07:19.26 | IronHelix | yeah it shouldnt be that bad just make some architecture decisions before you do anything |
07:19.29 | diamon | mog: Errr, if you are watching a knife infomercial, do your brain a favor and turn it off, then go get a book... |
07:19.34 | benjk | mog, that's how it always starts |
07:19.41 | mog_home | heh i have many a book |
07:19.42 | IronHelix | what you can always do to start out at least is wire your whole house to a single FXS port or ATA |
07:19.53 | benjk | and it ends with buying useless stuff off the TV |
07:19.57 | mog_home | i am reading specs right now |
07:19.58 | IronHelix | then split off extensions later or run cat5 cable for IP phones if you decide to |
07:20.05 | IronHelix | thus making the decision that you arent going to decide |
07:20.05 | mog_home | i will never buy stuff off tv |
07:20.07 | mog_home | although |
07:20.14 | mog_home | my woman has the ronco oven thing |
07:20.27 | IronHelix | i do highly recommend however that you pick up a grandstream budgetone 101 or grandstream gxp2000 and play with it |
07:20.32 | benjk | no, if you wire your houise, start with cat5 |
07:20.35 | IronHelix | www.voipsupply.com i've had good luck with them |
07:20.44 | diamon | Iron: That's exactly what I was thinking to do. Moreso because my phone lines are badly damaged in half the house, and it'll be a PITA to fix, so I'll likely just run cat5 pairs to each outlet, do a network and phone line at once. |
07:20.49 | IronHelix | if you RUN wires then run 5x cat5 everywhere |
07:20.53 | benjk | Iron: how fast do they deliver? |
07:21.05 | IronHelix | i never get rush shipping so 3-5 days |
07:21.09 | benjk | I was looking at them this week |
07:21.10 | diamon | 5 strands?! |
07:21.25 | IronHelix | the more the better |
07:21.30 | IronHelix | running ANYTHIN Is a PITA |
07:21.33 | IronHelix | so if you run 5 at once |
07:21.35 | IronHelix | then you're set |
07:21.40 | benjk | A customer of mine ordered some ATAs and stuff from Voxilla and it took two months |
07:21.43 | IronHelix | leave 4 of them hanging there maybe |
07:21.43 | benjk | scary |
07:21.49 | IronHelix | wow |
07:21.51 | IronHelix | thats horrible |
07:22.05 | IronHelix | try voipsupply, they're reputable, if it makes any difference bandwidth.com uses them as a supplier |
07:22.09 | mog_home | i bought some iaxys from digium |
07:22.12 | mog_home | took a week |
07:22.16 | IronHelix | lol |
07:22.19 | mog_home | and then i just walked down the hall |
07:22.22 | mog_home | and grabbed em |
07:22.26 | benjk | he called Voxilla every day and therew was always only answering mode |
07:22.27 | mog_home | ^_^ |
07:22.32 | diamon | Iron: A triple is my max, 5 would be huge, and with 5 outlets in the house to rewire, my main point would have 25 cat5 sets coming to it! |
07:22.32 | IronHelix | hehehe |
07:22.39 | benjk | a week is ok |
07:22.43 | benjk | but 2 months |
07:22.52 | IronHelix | diamon- it doesnt have to be 5 but it should def. be more than one |
07:23.05 | IronHelix | run good cable too, cat6 if you can afford it |
07:23.08 | IronHelix | so upgrading is easy |
07:23.28 | IronHelix | or better |
07:23.32 | IronHelix | this is HARD |
07:23.36 | IronHelix | but if you run conduit |
07:23.41 | IronHelix | all your problems will be solved forever |
07:23.48 | diamon | Yeah, a pair is my usual, or a triple. Can't do cat6 yet. First, it's not a spec; second, the folks who claim to be cat6 are HUGELY expensive per foot. I'll just use my cat5e. It'll do. |
07:24.13 | diamon | Iron: I was toying with doing that, but I think the house's design will prevent me from being able to do a proper conduit. |
07:24.13 | IronHelix | i thought it had been nailed down |
07:24.24 | mog_home | lol |
07:24.36 | diamon | Iron: If it was, it's VERY recent, I looked into that just a month ago or so. |
07:24.36 | mog_home | they say if the knife isnt up to the right level of density |
07:24.46 | mog_home | forget name of scale |
07:24.50 | mog_home | but if it doesnt meet it |
07:24.53 | mog_home | you can send it back |
07:25.00 | diamon | Iron: Most folks are using the base pre-spec that they think will become the standard. Might, might not. |
07:25.08 | IronHelix | diamon- yeah, then triple cat5e should be fine then |
07:25.11 | diamon | mog: Moh's hardness? |
07:25.17 | mog_home | yeah |
07:25.28 | IronHelix | how the hell are you going to test that |
07:25.32 | diamon | What rating are they saying? |
07:25.39 | IronHelix | i highly doubt that most people happen to have a materials tester laying around :\ |
07:25.40 | mog_home | they didnt say |
07:25.45 | mog_home | its just guaranteed |
07:25.46 | diamon | Iron: With items higher and lower on the scale, silly! :) |
07:26.00 | mog_home | they sell like 80 knives |
07:26.07 | mog_home | so some are harder than others |
07:26.17 | mog_home | but if its not hard enough they will rma it for ya |
07:26.19 | IronHelix | lol |
07:26.31 | mog_home | i wonder how do you test it |
07:26.35 | mog_home | do you crush it? |
07:26.53 | mog_home | or put it in water and condense water pressure? |
07:27.04 | mog_home | err increase not condense |
07:27.59 | diamon | Nah, scratch it. Moh's is a hardness scale. Higher hardness will hold a tighter edge, but is more likely to break. |
07:28.22 | mog_home | oh so go get some constant and try to scratch it |
07:28.34 | mog_home | lol, they just went to commercial |
07:28.44 | diamon | So if they claim it should be a 7.2 Moh's metal, you can get something a bit higher to test it, and a bit lower to see if it fails to scratch it... |
07:28.46 | mog_home | how can you sell ads during an ad |
07:28.52 | mog_home | yeah |
07:28.55 | mog_home | that makes sense |
07:29.01 | diamon | Like, diamonds are a 10, and talc is a 1 or something... |
07:29.15 | mog_home | yeah i remember 6th grade now |
07:30.07 | *** join/#asterisk L|NUX (i=linux@202.5.131.28) |
07:32.12 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
07:32.39 | diamon | My poor computer... It's showing ECC errors in the IDE SMART info. That's not good... |
07:32.58 | mog_home | eep |
07:33.19 | IronHelix | eek |
07:33.25 | IronHelix | hope you got backups :) |
07:33.34 | jayk- | might just be your ide controller |
07:35.29 | diamon | I think it's my cable. Stupid SATA rev1 cables don't have clips to keep them in place, always wiggling loose. |
07:35.50 | IronHelix | blah |
07:35.56 | diamon | No problems yet, I was just running SpinRite to check the drive. |
07:36.07 | diamon | I love SpinRite... It's my friend. |
07:37.57 | diamon | So, assuming I get an ATA for my phone system so I can keep my current cordless, is there any simple way to pass a code to specify usage of VoIP or POTS for dialling outbound? Like, if I just dial it uses VoIP, and if I do *9 then dial it's VoIP? |
07:38.36 | Dr_Ray | yes, in your extensions.conf |
07:38.43 | mog_home | okay ive decided |
07:38.46 | mog_home | enough with voip |
07:38.51 | benjk | diamon: get an IAXy |
07:38.51 | diamon | Dr: I'll read up on that then. |
07:38.52 | mog_home | im going into the knife business |
07:39.03 | h3x | mog, kyocera? hehe |
07:39.12 | mog_home | ahhh |
07:39.13 | BladeRunner05 | IronHelix: Hola |
07:39.19 | IronHelix | blade! |
07:39.22 | IronHelix | hows it goin |
07:39.24 | Dr_Ray | get a cnc machine |
07:39.34 | diamon | benjk: That was the leading one so far in my list, though codec support seems a bit limited... Can that be extended? |
07:39.39 | mog_home | who has a cnc machine |
07:39.42 | IronHelix | diamon- yeah you can do that |
07:39.47 | benjk | Dr_Ray: a CNC machine for the cordless? how would that work? |
07:39.48 | mog_home | and how can i borrow it for an hour..... |
07:40.00 | Dr_Ray | get a cnc machine for the mogknifefacotry |
07:40.06 | diamon | LoL. |
07:40.09 | benjk | diamon: no it cannot |
07:40.12 | mog_home | ooh thats a great idea |
07:40.25 | Dr_Ray | emachineshop.com |
07:40.32 | benjk | diamon: Atcomm makes IAX ATA's with more codecs |
07:40.33 | mog_home | why resell someone else's knives when i can make my own |
07:40.46 | Dr_Ray | but if you are going to make more than 1 then a cnc machine would pay for itself |
07:40.54 | mog_home | yeah |
07:41.01 | mog_home | how much is a cnc machine |
07:41.05 | diamon | benjk: Foo. Thought so, but I was hoping. Can I have the * system change codecs for it? |
07:41.06 | mog_home | i know its massively expensive |
07:41.06 | IronHelix | $10k |
07:41.10 | Dr_Ray | ~1000 or so on ebay |
07:41.12 | mog_home | yikes |
07:41.13 | diamon | mog: If you have to ask, you can't afford one. |
07:41.16 | mog_home | lol |
07:41.23 | mog_home | just cheap |
07:41.30 | Dr_Ray | emachineshop.com is an online one |
07:41.32 | mog_home | well not really high |
07:41.34 | Dr_Ray | wiht cad software |
07:41.43 | Dr_Ray | er, cadlike software |
07:41.44 | benjk | diamon: sure |
07:41.46 | mog_home | ooh |
07:41.58 | mog_home | i saw a project like asterisk for cnc stuff i want to try |
07:42.03 | Dr_Ray | emachineshop is big fun |
07:42.04 | IronHelix | anyway |
07:42.05 | IronHelix | im out |
07:42.07 | IronHelix | nite all |
07:42.10 | diamon | Yeah, emachineshop is nifty. I was toying with having them carve me a few GPU heatsink water blocks, but someone finally made a non-crappy one. |
07:42.11 | mog_home | i dont know what i would make with a machine like that |
07:42.15 | IronHelix | diamon- good luck wiring your pad |
07:42.21 | mog_home | i think i would just make cases for my toys |
07:42.37 | diamon | Iron: Yeah, it's gonna be fun... |
07:42.53 | Dr_Ray | I've yet to pay for any parts to be made, but I've enjoyed making stuff with the software and then pricing it |
07:43.51 | diamon | Heh, good thing it's so automated! I did have a client who wanted to figure out how to get a custom-made crankshaft for an old-ass motorcycle, I pointed him there. Worked out nice. |
07:44.10 | Dr_Ray | yeah, I figured out how to make some custom payphone parts |
07:44.13 | Dr_Ray | it was nifty |
07:44.18 | mog_home | cool |
07:44.44 | mog_home | i want just want to make cool cases for my projects |
07:45.07 | mog_home | how long would it take a cnc machine to make something as complicated as like a linksys router case |
07:45.21 | Dr_Ray | in plastic or in metal? |
07:45.43 | diamon | That's one thing my customers always love. If they're willing to pay my rates, I'm willing to do most anything. Had a guy pay me an hour a day to drive to his place and spend 30 mins feeding and playing with his dogs when he had to leave town on no notice. He paid for the 15-min drive and my 30 mins there, so 1hr/day, and he was happy. Nice dogs, too. |
07:46.41 | mog_home | plastic |
07:46.53 | diamon | mog: For plastic stuff, you might be better served with some kind of vacuum molding or injection system, especially if you want to do small sets of more than one. |
07:47.12 | mog_home | well what about a pc case |
07:47.13 | Dr_Ray | emachineshop will do vacume and plastic injection |
07:47.24 | Dr_Ray | they have a bunch of tools |
07:47.28 | mog_home | how hard is something like that out of curiosity |
07:47.37 | mog_home | i just wonder how fast i could make like 1000 units |
07:47.44 | diamon | mog: Heck, with a metal roller and folding sheetstock set you can make one cheap. |
07:47.52 | jayk- | we have a cnc machine |
07:47.58 | mog_home | ooh your lucky |
07:48.00 | jayk- | 3 & 5 axis |
07:48.01 | Dr_Ray | with injection molding (you need molds) and with vaccum you need a blank to form it on |
07:48.16 | Dr_Ray | once that is created it's a snap |
07:48.20 | jayk- | we specialize in vacuum molding too |
07:48.25 | mog_home | so like a few minutes |
07:48.28 | mog_home | or seconds |
07:48.44 | jayk- | we are a plastics, tooling, and composites manufacturer |
07:48.48 | diamon | mog: It'd probably need more time to cool enough to remove than anything, once the forms exist. |
07:48.52 | Dr_Ray | emachine shop is good for 1's and 2's, if you wanted 10k or whatever you'd be better off going overseas |
07:49.06 | mog_home | jayk can you make me something ^_^ |
07:49.10 | jayk- | sure |
07:49.15 | mog_home | really? |
07:49.17 | jayk- | yeah |
07:49.21 | mog_home | i want a case for my soekris |
07:49.24 | mog_home | and my x100p |
07:49.38 | mog_home | that looks like a linksys router case basically |
07:49.49 | jayk- | how many do you want |
07:49.53 | mog_home | 1.... |
07:49.59 | jayk- | heh |
07:50.02 | jayk- | probably not worth it |
07:50.05 | mog_home | yeah |
07:50.10 | mog_home | not worth it to you |
07:50.10 | diamon | mog: Uniques are always wildly pricey. |
07:50.12 | jayk- | nope |
07:50.13 | mog_home | ^_^ |
07:50.16 | Dr_Ray | emachineshop |
07:50.19 | mog_home | yeah i figured as much |
07:50.20 | Dr_Ray | check em out mog |
07:50.21 | jayk- | maybe if you wanted 500 |
07:50.22 | jayk- | :) |
07:50.41 | mog_home | i need to go round up 500 friends |
07:50.43 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
07:50.45 | mog_home | hrm only 300 in channel |
07:50.50 | mazzanet | ok |
07:50.55 | mazzanet | crazy question of the day |
07:51.04 | jayk- | heh |
07:51.04 | jayk- | well |
07:51.06 | jayk- | maybe 50 |
07:51.06 | diamon | And one less; I'd best go ask my Mythtv questions before it gets too much later... |
07:51.08 | mog_home | how much is it per unit at 500? |
07:51.12 | jayk- | beats me |
07:51.16 | Dr_Ray | I priced making brass payphone keys, and 1 was $200 and 200 was $310 |
07:51.30 | mog_home | okies |
07:51.30 | Dr_Ray | the first one is the hardest |
07:51.31 | mazzanet | i need an app say like hyperterminal, that can work over SIP |
07:51.46 | mog_home | umm |
07:51.46 | diamon | Dr: Exactly. |
07:52.10 | mog_home | hmm |
07:52.11 | diamon | Mazza: You just sprained my mind... That's like wanting to do VoIP over SMTP or something... |
07:52.16 | mog_home | you could do a modem connection over sip |
07:52.18 | mog_home | in ulaw |
07:52.22 | mazzanet | thats what i meant |
07:52.24 | mazzanet | modem connection |
07:52.24 | Dr_Ray | emachineshop is great for prototyping.. then you send them off to jayk after the design is done |
07:52.29 | mog_home | but it will work rarely at best |
07:52.39 | mog_home | well my problem is its just for me kinda stuff |
07:52.48 | mog_home | so at most i would need 10-25 for friends and family |
07:52.49 | Dr_Ray | then emachineshop is the way to go |
07:53.01 | mog_home | so it would never make sense for me to get it really done |
07:53.05 | mog_home | i just dream |
07:53.13 | Dr_Ray | or you could learn to do it yourself |
07:53.23 | Dr_Ray | vacumm forming is not difficult |
07:53.23 | mog_home | yeah |
07:53.24 | diamon | Dr: Oooh, hand-carved! |
07:53.30 | mog_home | heh |
07:53.31 | mazzanet | ie. the software equivalent of going hyperterminal -> 56k modem -> ATA -> asterisk -> pstn |
07:53.35 | mog_home | i did a wood case once |
07:53.39 | diamon | mog: Make it from wood! Give it a niiiice finish. :) |
07:53.42 | mog_home | dont do it mazz |
07:53.45 | mazzanet | i just want to go hyperterminal or similar -> asterisk -> pstn |
07:53.48 | mog_home | it over heated |
07:53.52 | mog_home | and wood warped over time |
07:54.15 | mog_home | i mostly leave stuff open or in tuperware containers |
07:54.18 | jayk- | i got my keyboard wet one time |
07:54.22 | jayk- | and set it up on the heat register |
07:54.23 | mog_home | as you can get em in most shapes and sizes |
07:54.26 | benjk | mog: is the tdm2400 a 6port card only? |
07:54.31 | jayk- | after about 12 hours, it had melted |
07:54.34 | mog_home | no |
07:54.34 | jayk- | but very very slowly |
07:54.35 | mog_home | 24 |
07:54.36 | jayk- | it looked really weird |
07:54.39 | mog_home | 6 modules |
07:54.40 | diamon | mog: More cooling, and a better-cured wood then... I did a 50W car amp cover in wood, with two fan ports to keep the MOSFETs cool. |
07:54.44 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
07:54.47 | mog_home | 4 ports per module |
07:54.58 | mog_home | well i didnt have fans |
07:54.59 | Dr_Ray | does digium have new toys? |
07:55.01 | mog_home | that was the problem |
07:55.04 | shido6 | yes |
07:55.04 | mog_home | tdm2400p |
07:55.06 | shido6 | check the site |
07:55.09 | diamon | jayk: Press the Escher key to continue! |
07:55.35 | benjk | mog: you may want to talk to whover is in charge of making the content for Digium;'s website, the tdm2400 page is not really clear on this |
07:55.38 | diamon | (Pun on melting-flowy keyboard image I have in my mind) |
07:55.42 | Dr_Ray | ! |
07:55.57 | mazzanet | mog_home: i can use G.729 by the way |
07:55.59 | mog_home | hrmm i will pass word along |
07:56.00 | jayk- | diamon :) |
07:56.02 | jayk- | i should have taken a picture |
07:56.03 | mog_home | what? |
07:56.30 | diamon | jayk: I had a fun one trying to remove a keyboard's melted remains out of a fancy dishwasher. It went....poorly. |
07:56.40 | mog_home | heh |
07:56.44 | Dr_Ray | does this have IRQ problems? |
07:56.44 | benjk | so these are different modules than the ones for the tdm400 then, I see |
07:56.52 | mog_home | i have only really messed up one keyboard |
07:56.54 | mog_home | yes |
07:56.59 | mog_home | well slightly |
07:57.07 | mog_home | the fxs modules are more different |
07:57.15 | diamon | A customer decided my cleaning method was too slow and dainty, so they just tossed it in the dishwasher. Heated dry did it in I think, though the sanitizing heat-wash did it no good I'm sure. |
07:57.16 | mog_home | the fxo is very similar |
07:57.22 | mog_home | just more on one board |
07:57.26 | Dr_Ray | but you can put multiple of these cards in a box? |
07:57.29 | jayk- | did it ruin the dishwasher? |
07:57.31 | mog_home | yes |
07:57.36 | mog_home | we ran 4 in a machine for grins |
07:57.38 | benjk | what's that strange looging connector sticking out at the back of the card? |
07:57.45 | mog_home | digium wont ever reccomend 2-3 cards in a machine |
07:57.53 | diamon | jayk: Yeah, turned out it had jammed the pump impellers. I wish I could have seen it in action. |
07:57.57 | mog_home | that is an amphenol (sp) |
07:58.00 | mog_home | connector |
07:58.00 | Dr_Ray | benjk tel;co punchdown |
07:58.07 | mog_home | you plug a punchdown block |
07:58.07 | benjk | I can recommend up to 6 cards in a single machine |
07:58.11 | mog_home | or breakout box |
07:58.19 | mog_home | heh we do it in testing lab |
07:58.21 | mog_home | and it works |
07:58.25 | benjk | works flawlessly, no problems |
07:58.27 | jayk- | the mesh screening in mine had a gaping hole |
07:58.30 | mog_home | but you wont catch us reccomending it |
07:58.33 | benjk | but it has to be a Mac |
07:58.36 | jayk- | it was sickening all the junk that passed through it and clogged up my drain lines |
07:58.38 | benjk | a vintage Mac |
07:58.41 | benjk | 9600 |
07:58.41 | mog_home | i have seen 5 quad cards in a machine |
07:58.44 | mog_home | and seen it just work |
07:58.45 | diamon | If you're going to have that many cards, wouldn't you want to just give up and get a DS/T1 card and a channel bank for it and be done? |
07:58.49 | *** join/#asterisk grimse (n=grimse@p5481CCBF.dip.t-dialin.net) |
07:58.50 | mog_home | but i wouldnt tell peopl to do it |
07:58.58 | Dr_Ray | what's the price of this monster? |
07:59.04 | mog_home | well it is nice to just have one box diamon |
07:59.06 | benjk | mog: no problem on the Mac architecture |
07:59.16 | mog_home | yeah |
07:59.21 | diamon | jayk: I won't even ask, I was toying with food and want to keep my appetite. |
07:59.22 | mog_home | ppc arch is better at irqs |
07:59.22 | benjk | unfortunately these days Macs only have 3 PCI slots |
07:59.24 | Dr_Ray | if you need to drive fxo lines, then a channel bank stinks |
07:59.29 | mog_home | yup |
07:59.35 | mog_home | yeah that too |
07:59.40 | Dr_Ray | stinks. well, less than ideal |
07:59.43 | mog_home | its cheaper than channel bank, at least rhinos |
07:59.56 | diamon | Wouldn't it be better to split the cards between systems to have a bit of physical redundancy anyway? |
07:59.58 | mog_home | well fxo lines arent fun |
08:00.01 | mog_home | via la pri |
08:00.04 | mog_home | YES |
08:00.07 | Dr_Ray | so $1000 fully populated? |
08:00.13 | Dr_Ray | ish? |
08:00.15 | mog_home | ermm no |
08:00.55 | mog_home | I think its on our site and voip-supply |
08:00.55 | mog_home | as well as our resellers |
08:00.55 | mog_home | i dont deal with the moneys though sorry |
08:00.55 | Dr_Ray | I don't see it at the asterisk store |
08:01.01 | mog_home | :( |
08:01.04 | mog_home | someone must die |
08:01.23 | benjk | so does this card also come with a converter to RJ11? |
08:01.39 | benjk | or do you have to make your own adapter cable? |
08:01.41 | Dr_Ray | benjk - you use the punch down block |
08:01.42 | mog_home | no not from digium, but we are going to offer breakout boxes |
08:01.54 | mog_home | i imagine some people will sell it with it |
08:01.58 | mog_home | its a standard interface |
08:02.07 | Dr_Ray | greybar sells them for $20 |
08:02.09 | benjk | I don't know what a punch down block is, but anyway, does it come with the card? |
08:02.23 | mog_home | yeah greybar rocks |
08:02.31 | mog_home | depends who you get it from |
08:02.32 | diamon | Agreed. |
08:02.34 | benjk | thats very bad news then |
08:02.40 | mog_home | digium will sell breakout boxes as well |
08:02.43 | Dr_Ray | greybar is great to walk in and say I need this, and have them |
08:02.51 | benjk | it will increase the cost of the card here in Japan by at least 100 USD if not 150 |
08:03.03 | mog_home | eep benjk |
08:03.14 | mog_home | you have to be able to get breakout boxes in japan |
08:03.32 | benjk | yes because if you havbe to order the cable somewhere else then iot is additional FedEx, customs and import duties |
08:03.35 | mog_home | its an age old standard |
08:03.40 | Dr_Ray | ok, that's awesome |
08:03.48 | mog_home | ? |
08:03.52 | Dr_Ray | this new card |
08:03.57 | benjk | nothing telco related in Japan is compatible with anywhere else |
08:03.59 | mog_home | yeah its pretty cool |
08:04.03 | Dr_Ray | in a 1U case, that's studly |
08:04.12 | mog_home | we rolled some of digium over to it a while back |
08:04.15 | mog_home | it works great |
08:04.26 | mog_home | im sorry ben |
08:04.26 | Dr_Ray | I'm going to keep using adit 600's but that will have it's uses |
08:04.27 | *** join/#asterisk Joe2000x (n=jo@202.155.89.122) |
08:04.45 | Joe2000x | hi |
08:04.47 | Dr_Ray | esp, the abilty to reconfigure on the fly |
08:04.57 | Joe2000x | is anyone ever use asterisk realtime ? |
08:05.01 | mog_home | make sure you get it bundled or order with your card, they usually arent that expensive |
08:05.02 | mog_home | yup |
08:05.22 | benjk | mog, you better get your sales/marketing gurus to ship that card with an adapter cable for overseas customers |
08:05.24 | Joe2000x | why i dont have the realtime command in my CLI? |
08:05.32 | Dr_Ray | I was hoping you guys were going to make a 24 port iaxy type device, but this is good enough |
08:05.58 | Dr_Ray | I don't get the impression that mog is running the joint over there |
08:06.01 | Dr_Ray | :) |
08:06.05 | benjk | Ah, so Digium *will* seel it bundled with some RJ11 solution |
08:06.10 | mog_home | lol |
08:06.22 | Dr_Ray | benjk - I'm sure if you axed them, they would set you up |
08:06.23 | mog_home | we are gonna have breakout boxes available ben |
08:06.28 | mog_home | just buy one with it |
08:06.46 | mog_home | and i dont run the joint, but we are a family opperation |
08:06.47 | Dr_Ray | but that is studly |
08:06.58 | jayk- | what do you run, mog |
08:07.11 | mog_home | away |
08:07.12 | mog_home | ^_^ |
08:07.14 | benjk | mog makes the coffee |
08:07.19 | mog_home | heh |
08:07.22 | benjk | hence the name mug |
08:07.22 | mog_home | we dont drink coffee |
08:07.26 | mog_home | just redbull |
08:07.26 | Dr_Ray | making the copies |
08:07.31 | benjk | ah no that s the wrong spelling |
08:07.46 | mog_home | i work in dev |
08:08.02 | diamon | Ugh, Red Bull makes my heart skip after just a few... |
08:08.10 | Dr_Ray | so the echo cancelation is one per card? |
08:08.15 | benjk | I work in the factory |
08:08.19 | benjk | On the chain? |
08:08.29 | benjk | No, we are allowd to walk around freely |
08:08.53 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
08:08.54 | mog_home | yes dr_ray it has an optional echo can |
08:08.57 | *** part/#asterisk diamon (n=diamon@c-66-176-91-189.hsd1.fl.comcast.net) |
08:09.07 | mog_home | like the echo can for te405/410 |
08:10.43 | newmember | I cant seem to dial out on my FX0 interface, where is a good place to watch the call process for debugging? |
08:10.53 | Dr_Ray | asterisk -r |
08:10.53 | mog_home | easiest way |
08:10.56 | mog_home | is zap barge |
08:10.56 | Dr_Ray | set verbose 10 |
08:11.05 | mog_home | if you actually want to hear the audio |
08:11.08 | mog_home | otherwise do that |
08:11.48 | newmember | I have my phones registering and they can call each other, but my 'dial 9' does nt pick up a line to call out |
08:11.50 | benjk | maybe its because its an FX0 card |
08:12.29 | benjk | the dialing goes zero |
08:12.46 | benjk | if it was an FXO card on the other hand ... |
08:12.46 | newmember | FX0 goes to the PSTN? |
08:12.59 | newmember | right |
08:13.02 | newmember | FXO |
08:13.07 | benjk | FXO does, but FX0 I am not so sure ;-) |
08:13.14 | newmember | right |
08:13.48 | newmember | So it doesnt pick up the FXO port |
08:13.59 | newmember | when I dial nine and a number |
08:14.26 | benjk | what does it say when you do zap show channels |
08:14.34 | benjk | does it show up? |
08:15.04 | newmember | pseudo from-internal en |
08:15.04 | newmember | <PROTECTED> |
08:15.04 | newmember | <PROTECTED> |
08:15.04 | newmember | <PROTECTED> |
08:15.04 | newmember | <PROTECTED> |
08:15.28 | newmember | I think thats a goos sign |
08:15.33 | newmember | good sign |
08:16.57 | mog_home | what does your dial statement look like |
08:17.38 | shido6 | amp? |
08:17.46 | newmember | either |
08:18.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
08:20.43 | Dr_Ray | thanks mog |
08:21.35 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
08:22.08 | mog_home | no problema |
08:24.34 | newmember | I see this when I call out |
08:24.36 | Dr_Ray | so is the echo cancelation for the tdm2400 an add on module? |
08:24.36 | newmember | <PROTECTED> |
08:24.36 | newmember | <PROTECTED> |
08:24.36 | newmember | <PROTECTED> |
08:24.52 | newmember | it looks liek it going to the zap trunk |
08:25.40 | newmember | I think my FXO is port 4, I think it should look like this -- Zap/1-4 |
08:25.55 | vira | ah |
08:25.56 | vira | er |
08:25.58 | Dr_Ray | Zap/4-1 |
08:26.42 | newmember | which is easier to change, the fxo on the card or the zaptel.conf? |
08:26.56 | Dr_Ray | zaptel.conf |
08:27.08 | Dr_Ray | IMO |
08:27.45 | *** join/#asterisk P4C0 (i=1000@201.224.107.47) |
08:27.47 | P4C0 | hello guys |
08:28.00 | mog_home | hi |
08:28.12 | mazzanet | how can i put a call on hold? |
08:28.42 | mog_home | flash |
08:28.44 | P4C0 | does anyones knows a good softphone client that works in linux KDE with artsd? |
08:28.47 | newmember | Intersting, when I ran compiled the zapel I thought it look correct, it looks like it has the fxo and fxs mixed up |
08:28.49 | newmember | fxoks=1 |
08:28.49 | newmember | fxoks=2 |
08:28.49 | newmember | fxoks=3 |
08:28.49 | newmember | fxsks=4 |
08:28.49 | mog_home | kphone |
08:29.06 | Dr_Ray | they are reversed in zaptel.conf |
08:29.11 | mog_home | and in zapata |
08:29.50 | newmember | revesred on purpose? |
08:29.54 | mog_home | yes |
08:30.01 | mog_home | fxo modules speak fxs signalling |
08:30.04 | benjk | so mog, what do you work on? asterisk, libpri or drivers? |
08:30.06 | mog_home | and fxs modules speak fxo signalling |
08:30.07 | Dr_Ray | reversed by perspective of where you are in the pbx |
08:30.27 | newmember | ty |
08:30.27 | mog_home | its intentionally reversed to match standard, not to confuse you |
08:30.31 | Dr_Ray | the fact that it is confusing is free |
08:30.34 | mog_home | i do internal things at digium, and i work on asterisk code |
08:30.39 | mog_home | nice side effect |
08:30.42 | newmember | gotta like oss |
08:31.20 | newmember | in the real world you call that the 'network' side of the voice link |
08:31.24 | Dr_Ray | extensions you are looking in to the asterisk server, zapata and zaptel, you are looking out, so it will be reversed |
08:31.34 | mog_home | no newmember |
08:31.49 | mog_home | i mean its not userfriendly |
08:31.56 | benjk | Dr_Ray, it's not "reversed" |
08:31.59 | mog_home | but its friendly to anyone that knew telephony pre asterisk |
08:32.13 | newmember | So I am looking to change from Zap/1-1 to Zap/4-1 |
08:32.31 | benjk | and FXO *interface* has the purpose to connect to an exchange office, hence the name |
08:33.05 | Dr_Ray | and it uses fxs signalling |
08:33.19 | benjk | the fxo interface has to speak FXS signaling to the office because the office only talks to stations |
08:33.36 | benjk | its perfectly logic |
08:33.44 | benjk | logical |
08:33.52 | Dr_Ray | mog - is the echo canceling for the tdm2400 built in or is it a drop in module? |
08:33.53 | mog_home | once its explained... |
08:33.58 | Dr_Ray | once you get it |
08:33.59 | Dr_Ray | :) |
08:34.00 | mog_home | its a drop in module |
08:34.08 | mog_home | just like 410 and 411 |
08:34.15 | mog_home | you could upgrade it if you needed it later |
08:34.16 | Dr_Ray | so I can buy it later if needbe |
08:34.21 | Dr_Ray | perfect |
08:34.21 | mog_home | yup |
08:34.31 | mog_home | the software echo can rocks in asterisk |
08:34.47 | Dr_Ray | I've been blessed with damn near no echo |
08:34.51 | mog_home | personally i think echo can stuff is only needed in serious cases, and in places where you need cpu |
08:35.01 | mog_home | but analog lines are more prone to echo than t1s |
08:35.11 | mazzanet | but say i have a phone that doesn't have a hold button and it's plugged into an ATA |
08:35.15 | mazzanet | can i put people on hold? |
08:35.22 | mog_home | if you flash the line |
08:35.25 | Dr_Ray | yes, via flashookl |
08:35.30 | mog_home | i bet the ata will put that caller on hold |
08:35.34 | benjk | mog: in Japan there is always trouble with echo |
08:35.41 | benjk | on analog lines |
08:35.51 | mog_home | man ben if japan is that bad, id leave ^_^ |
08:35.58 | mog_home | yeah its that way in the south too |
08:35.58 | benjk | in some cases it was so bad that we had to back out of the deal |
08:36.05 | mog_home | yikes |
08:36.13 | mazzanet | flash the line? |
08:36.15 | benjk | mog: I'll consider it |
08:36.19 | benjk | ;-) |
08:36.21 | mog_home | go on and off hook real fast |
08:36.23 | mog_home | heh |
08:36.37 | mog_home | usually there is a button that says flash |
08:36.40 | mog_home | that will do that for you |
08:36.44 | benjk | yeah, that's a moron scheme of a caller ID system isn't it? |
08:37.11 | mog_home | meh, i do find it funny how locked down your whole telephony stuff is |
08:37.45 | mazzanet | ah cool |
08:37.46 | benjk | mog it used to be like that pretty much everywhere other than in the US and Finland |
08:37.53 | mazzanet | now how do i add on hold music |
08:38.01 | mog_home | asteirsk will do it for you |
08:38.09 | benjk | national monopolies locked everthing down |
08:38.13 | mog_home | yeah |
08:38.21 | newmember | ok getting close |
08:38.28 | newmember | I just need it to drop the 9 |
08:38.30 | mog_home | you have to cut the 9 off |
08:38.32 | benjk | Japan is just changing its ways only very reluctantly |
08:38.38 | mog_home | do ${EXTEN:1} |
08:38.48 | mog_home | instead of ${EXTEN} newmeber |
08:38.52 | mog_home | that will trim it |
08:39.06 | mog_home | also you want this www${EXTEN:1} |
08:39.19 | mog_home | that will wait 300 millaseconds so the line waits for dial tone |
08:39.46 | benjk | it will dial your extension via the web browser instead of your land line |
08:39.47 | benjk | :-) |
08:39.55 | mog_home | heh |
08:40.12 | mog_home | one day ben |
08:41.14 | newmember | _9.,1,Macro(dialout-trunk,1,${EXTEN:1}) |
08:41.25 | mog_home | yeah that will work |
08:41.32 | newmember | hmmm |
08:41.44 | mog_home | i only reccomend the www s as sometimes it takes some time for the line to sieze |
08:41.56 | newmember | righ t good idea |
08:43.03 | newmember | <PROTECTED> |
08:43.03 | newmember | <PROTECTED> |
08:43.03 | newmember | <PROTECTED> |
08:43.12 | newmember | didnt drop the 9 |
08:43.15 | mog_home | do a noop |
08:43.24 | mog_home | i dont think you are doing that macro |
08:43.32 | mog_home | you are probably doing something else |
08:43.49 | benjk | is that AMP? |
08:43.56 | mog_home | yeah it is i feel |
08:44.05 | benjk | Oh dear |
08:44.08 | mog_home | i can feel the amp /A@H |
08:44.49 | benjk | probably some hidden value in the database that sends the dialplan somewhere else |
08:44.54 | P4C0 | if I want to register into an asterisk server (voip provider) with my asterisk server in which conf file should I put it? |
08:44.56 | mog_home | lol |
08:45.04 | mog_home | how are you connected |
08:45.06 | mog_home | sip or iax |
08:45.09 | mog_home | or h323 |
08:45.32 | P4C0 | sip |
08:45.33 | newmember | sorry just reading the call log |
08:45.37 | mog_home | sip.conf |
08:45.40 | mog_home | yeah |
08:45.43 | mog_home | do a noop in there |
08:45.44 | newmember | ya I installed aah |
08:45.51 | mog_home | its likely not going through that macro |
08:45.56 | mog_home | or that macro sucks |
08:46.01 | newmember | I anm rading this |
08:46.02 | newmember | <PROTECTED> |
08:46.06 | P4C0 | but isn't sip.conf for the clients? |
08:46.10 | newmember | I am reading that |
08:46.12 | P4C0 | I mean, local clients |
08:46.20 | mog_home | just do _9.,1,Dial(zap/g1/www${EXTEN:1}) |
08:46.22 | mog_home | there you go |
08:46.23 | newmember | I use SIP to my cisco 7960 phones |
08:46.25 | benjk | no sip.conf is for SIP |
08:46.25 | mog_home | its for both |
08:46.30 | mog_home | all SIP |
08:46.38 | P4C0 | mog_home: thanks |
08:46.42 | mazzanet | well when i flash the line, it definately appears i get put on hold |
08:46.46 | P4C0 | asteriskguru.com will be my friend :p |
08:46.52 | mazzanet | but i get no music |
08:47.16 | mog_home | well then your ata is not relaying the hold |
08:47.23 | mog_home | id need sip debug to know for sure |
08:47.49 | mazzanet | now thats an idea |
08:48.07 | mog_home | but i tend to not debug sip for fun ^_* |
08:48.33 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
08:50.27 | mazzanet | well from sip debug |
08:50.31 | mazzanet | when i flash the line |
08:50.54 | mazzanet | i just get the usual INVITE -> ACK routine |
08:51.47 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
08:54.41 | mazzanet | hmm |
08:54.42 | Dr_Ray | the redphone thing looks neat |
08:54.49 | mazzanet | theres an option on my ATA called |
08:54.53 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
08:54.57 | mazzanet | "Call Hold using c=0.0.0.0 (RFC 2543) in SDP" |
08:55.12 | mog_home | it is a nifty idea dr_ray |
08:55.21 | mog_home | i prefer clustering with iax rather than tdmoe |
08:55.43 | Dr_Ray | well, I like the magic box aspect of it |
08:55.47 | Dr_Ray | drop and walk away |
08:56.07 | Dr_Ray | but my asterisk server is not the weak link in the chain |
08:56.56 | mog_home | yeah |
08:57.04 | mog_home | we dont have any hw failover at digium |
08:57.26 | Dr_Ray | my asterisk modules don't even autoload |
08:57.27 | Dr_Ray | :) |
08:57.32 | Dr_Ray | it's just not been a problem |
08:57.33 | mog_home | and we have had more problems with our pri going down then we have had with asterisk |
08:57.37 | mazzanet | well |
08:57.50 | mazzanet | when i flashhook |
08:58.20 | mazzanet | the receiving phone get some buzz/crackles every now and then |
09:00.00 | mog_home | if you do sip show channels |
09:00.05 | mog_home | or show channels |
09:00.10 | mog_home | does it say they are on hold |
09:00.17 | mog_home | i cant remeber where |
09:00.26 | mog_home | but somewhere asterisk will show you |
09:00.34 | mog_home | its likely you dont have musiconhold setup |
09:01.40 | mazzanet | 192.168.1.73 ata jxry0-tth84 00101/00145 ulaw Yes Rx: ACK |
09:01.48 | mazzanet | the Yes is for Hold |
09:01.53 | mog_home | yeah |
09:02.04 | mog_home | do you have mpg123-.59-r installed? |
09:02.08 | mazzanet | but the outgoing trunk isn't on hold |
09:02.17 | myke420247 | moh is funky |
09:02.21 | myke420247 | you need a timing device |
09:02.25 | myke420247 | and even then it may not work |
09:02.29 | myke420247 | it works fine on my grandstreams |
09:02.33 | myke420247 | but not on x-lite |
09:02.36 | myke420247 | duno why |
09:02.43 | mog_home | it will always work with timing device |
09:02.59 | benjk | hmmm, maybe we should go to Macedonia ... |
09:03.00 | benjk | http://www.theregister.co.uk/2005/11/24/worlds_biggest_wlan |
09:03.06 | mazzanet | i have mpg123 |
09:03.09 | mog_home | it just needs timing and that version of mpg123 but not with the newer stuff |
09:03.12 | mog_home | do you have that version |
09:03.16 | mog_home | it has to be exact |
09:03.33 | myke420247 | it comes bundled with asterisk so you can always build and install that version |
09:03.34 | mog_home | if you go into asterisk |
09:03.39 | mog_home | and type make mpg123 |
09:03.43 | mog_home | it will install it for you |
09:04.26 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
09:04.34 | mog_home | bbz DIE |
09:04.40 | mog_home | you are a bot |
09:04.43 | mazzanet | Version 0.59q (2002/03/23). |
09:04.43 | mog_home | and if not |
09:04.46 | mog_home | you need to respond |
09:04.51 | mog_home | it wont work with that version |
09:05.01 | mazzanet | ..... |
09:05.03 | mog_home | do you have a card in the machine |
09:05.07 | mog_home | or ztdummy loaded? |
09:05.25 | mazzanet | i'm not using zaptel. |
09:05.36 | mog_home | you need timing to do musiconhold right |
09:05.59 | mog_home | you can install ztdummy real easy if its a 2.6 kernel |
09:06.26 | *** join/#asterisk chapeaurouge (n=chap@85.201.80.249) |
09:15.19 | P4C0 | which is the conf file where I can put logic stuff like if and code? |
09:16.32 | Dr_Ray | extensions.conf? |
09:16.43 | Dr_Ray | dialplan logic? |
09:20.56 | P4C0 | dialplan logic that one |
09:21.20 | *** join/#asterisk genuix (n=genuix@62-167-18-224.adslpremium.ch) |
09:21.36 | asterboy | hola |
09:23.03 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
09:25.14 | asterboy | Anyone get through to Atacomm.com for orders? |
09:30.19 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
09:32.12 | mazzanet | well i now have on hold music |
09:32.21 | mazzanet | but it sounds really crap. |
09:34.26 | mazzanet | then again |
09:35.00 | mazzanet | resampling a 190-320kbps VBR 44.1khz mp3 down to 8khz has gotta hurt |
09:35.04 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
09:37.22 | *** join/#asterisk nemisus (n=nemisus@203-206-119-211.dyn.iinet.net.au) |
09:39.27 | mazzanet | now thats just freaking weird |
09:39.56 | mazzanet | ok i'm going internal phone -> ata -> asterisk -> pstn -> my other phone |
09:40.06 | mazzanet | when i put [my other phone] on hold |
09:40.32 | mazzanet | the hold music only plays when [my other phone] is 'transmitting' something |
09:40.34 | mazzanet | ie. making noise |
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09:47.27 | *** part/#asterisk genuix (n=genuix@62-167-18-224.adslpremium.ch) |
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09:54.50 | pooh_ | coppice: Nin Zou ;-) |
09:56.36 | *** join/#asterisk shadebob (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
09:56.41 | shadebob | hi |
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09:59.31 | shadebob | can I detect (with zap channel) answer of th callee party? Answeronpolarity is just for hangup ...? |
10:03.39 | pooh_ | shadebob: ${dialstatus} |
10:05.12 | shadebob | pooh_ : my problem is with an digital line (pri, bri) answer and hangup of callee give good cdr informations. With zap channel, cdr start with ring tone and not answer of callee |
10:06.16 | coppice | shadebob: most types of analogue line do not inform the caller when the called party answers |
10:08.49 | shadebob | there a no asterisk apps to detect the end of the ring tone? |
10:09.20 | coppice | the end of ring tone has no specific meaning |
10:12.14 | *** join/#asterisk bmg505 (n=leon@rndf-146-26-48.telkomadsl.co.za) |
10:14.17 | shadebob | so cdr information will be always false? |
10:14.56 | coppice | yep. nothing reliable you can do to avoid that |
10:15.37 | coppice | thank digital exchanges. all analogue lines had proper supervision in the strowger days. A few countries have it now. most do not |
10:15.53 | shadebob | how analog pbx can give good cdr information? |
10:16.39 | coppice | sometimes you can pay a higher rental and get supervision. PBX users like that. in many places that is not available, and PBXs cannot produce an accurate CDR |
10:18.03 | *** join/#asterisk patzub (n=patzub@109.9.101-84.rev.gaoland.net) |
10:18.42 | shadebob | how analog public payphone detect callee answer? |
10:19.19 | coppice | they do not use standard lines. they get supervision in one of several ways |
10:20.37 | shadebob | 2 cases : they use lines with pulse metering. In the other case they use standard line... |
10:21.41 | coppice | often privately owned payphones do not start charging correctly. telco owned ones have answer supervision |
10:22.34 | shadebob | maybe I can tell telco to have a line with pulse metering |
10:22.55 | shadebob | but I will code apps to detect 12khz or 16khz sign |
10:23.25 | shadebob | and I thing digium card have filter to supress such signal... |
10:23.44 | coppice | you need a card that can detect 12kHz or 16kHz |
10:23.50 | h3x | voip payphones... heh |
10:24.13 | coppice | i saw a VoIP payphone on a website somewhere |
10:24.20 | shadebob | It's for a gateway with existing payphones h3x ;) |
10:24.38 | h3x | hack an ATA to do that shit |
10:25.13 | coppice | how exactly would an ATA do this? |
10:25.18 | shadebob | I install an asterisk box with tdm card |
10:26.11 | h3x | why couldnt it |
10:26.22 | shadebob | linksys ata have an open firmware I thing.... but electronics componements are not present for this task.... |
10:26.22 | coppice | explain how |
10:26.46 | h3x | isnt the only electrical difference, ground start? |
10:27.06 | coppice | if you know of open fimware for any VoIP other than PA1688, do tell |
10:27.21 | h3x | well thats what i mean |
10:27.25 | coppice | they don't usually ground start |
10:27.25 | h3x | make an ata for payphones |
10:27.40 | h3x | using a 1688 would be a good start |
10:27.55 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
10:29.53 | coppice | shadebob: it seems the chips on the digium TDM cards can generate 12kHz and 16kHz, but I don't think they can detect those tones. |
10:30.44 | shadebob | old tdm card can.... but newer have new si3215 chip that can not generate |
10:31.24 | coppice | well, its detection you need, so it doesn't make a lot of difference |
10:31.42 | shadebob | but it seem si3215 can detect 12 or 16k signal |
10:31.56 | h3x | and turn those tones into sip info messages or whatever |
10:32.18 | h3x | the reason im thinking its a good idea is coz i was at a payphone convention here in vegas recently |
10:32.24 | coppice | so that should be OK |
10:32.42 | h3x | and a lot of the guys were talking about how they are getting out of the business because they are still getting raped with 1980's prices from the ILECs |
10:32.46 | h3x | for payphone access lines |
10:32.55 | h3x | and they dont have as much usage as before because of cell phones |
10:33.12 | h3x | bypassing it with wireless, cable modem, whatever they can get their hands on |
10:33.13 | coppice | and getting broadband access lines would be better? |
10:33.16 | h3x | hell even DSL is cheaper than a payphone line here |
10:33.26 | h3x | they are paying 60 bucks a month plus taxes ! |
10:33.35 | h3x | in some regions |
10:33.38 | coppice | I thought most new payphones were cellphones |
10:33.53 | h3x | ive never seen those here, it would make more sense tho |
10:34.34 | coppice | they are all over the place. often very fancy with big LCD touch screens that show ads when nobody is calling |
10:35.21 | Dr_Ray | the idea is to put 4 or 8 phones in a busy area with one dsl line |
10:35.28 | Dr_Ray | like at a hospital |
10:35.33 | Dr_Ray | where poor people are |
10:36.01 | coppice | is anyone really too poor to have a cellphone these days? |
10:36.05 | Dr_Ray | yes |
10:36.10 | Dr_Ray | or cellphone batteries die |
10:36.18 | Dr_Ray | we price our long distance at 10 cents a minute |
10:36.27 | Dr_Ray | better than a ghetto cell phone and calling card |
10:36.46 | *** join/#asterisk dudes (n=dudes@12-215-32-62.client.mchsi.com) |
10:37.07 | Dr_Ray | it's also an amenity |
10:37.28 | coppice | china has 1.2B people and 350M cellphone users. if you discount the poor farmers far from coverage, its pretty much 100% penetration in the cities |
10:37.51 | Dr_Ray | it's not 100% penetration |
10:37.57 | Dr_Ray | some people have multiple phonex |
10:37.59 | Dr_Ray | s |
10:38.10 | *** join/#asterisk mesut (n=power@unaffiliated/mesut) |
10:38.13 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
10:38.14 | coppice | it really is 100% |
10:38.20 | Dr_Ray | no, it really is not |
10:38.33 | Dr_Ray | the poorest people have cell phones that don't work all the time |
10:38.36 | Dr_Ray | run out of minutes |
10:38.36 | Dr_Ray | etc |
10:38.42 | mesut | hi |
10:38.44 | shadebob | ok coppice : in fxo module of tdm card the couple of chips si3050/si3018 is present. That chips can detect polarity reversal and billing tones (http://www2.silabs.com/public/documents/tpub_doc/dsheet/Wireline/Silicon_DAA/en/si3050.pdf) |
10:38.51 | coppice | are you a china expert? |
10:39.12 | Dr_Ray | are you? just saying it is, does not make it so |
10:39.26 | Dr_Ray | our payphones do fairly well when priced fairly |
10:39.51 | trixter | it only takes 1 person to not have a cell phone for it to not be 100% |
10:39.56 | coppice | I spend much of my life in china. no janitor cleaning to toilets would be without their cellphone |
10:40.00 | Dr_Ray | no, of course it's 100% |
10:40.07 | Dr_Ray | how could we be so stupid to argue with him |
10:40.12 | Dr_Ray | it's clearly 100% |
10:40.12 | trixter | heh |
10:40.16 | Dr_Ray | I apologize |
10:41.02 | trixter | how much of china is rural? cell phone usage would most likely be less in rural areas |
10:41.05 | Dr_Ray | I like how 350M of 1.2B is 100% |
10:41.10 | trixter | much like it is in all other rural areas around the world |
10:41.25 | trixter | oh I just walked into it is that what was said? |
10:41.35 | coppice | the main places payphone exist these days seem to be tourist spots, where people don't have roaming coverage |
10:41.49 | trixter | ahh he qualified with in the cities |
10:41.54 | trixter | how many 2 year olds have cell phones? |
10:42.01 | trixter | it only takes one for it not to be 100% in the cities :P |
10:42.02 | coppice | Dr idiot seems to have a reading problem |
10:42.03 | Dr_Ray | within the cities he's been too |
10:42.21 | Dr_Ray | hey, I apologized and said you right, no need for name calling |
10:43.37 | trixter | I would argue that the vast majority of china lives in the cities rather than in the rural areas though.. he siad 350M out of 1.2B - exclude the farmers and you are left with the city population (in effect) and I think it might be the other way around |
10:43.45 | coppice | in china 350M means cellphones are already spreading well into rural areas, with their number determined to a large extent by the speed with which coverage spreads |
10:44.10 | trixter | look at america as one example - this type of demographic is fairly constant -- the city of sacramento has 600k people (not counting the cities immediatly adjecent to sacramento) and the state of nebraska has 600k people |
10:44.38 | trixter | why is that? farms take a lot of space and you cant pile a lot of people in that space. by definition that is how it works. as a result the rural areas have far fewer people |
10:44.48 | trixter | I live one county away from sacramento and the whole county only has 40k people |
10:44.53 | trixter | same reason, its farm land |
10:44.55 | coppice | china still has a huge farming population. city dwellers are predominantly around the coast |
10:45.18 | trixter | now if 'market penetration' was defined as 'anyone who wants one has one' then perhaps I would agree |
10:45.50 | trixter | but I do not know if even that is accurate. I do know that not everyone who can have one does indeed have one - which is the marketing wet dream of any company |
10:46.12 | coppice | HK now has significantly more cellphone subscribers than people. i guess quite a few places are like that |
10:46.14 | trixter | yeah but if you look at *any* farming vs rural situation you will find that the vast majority of people are in the urban areas |
10:46.30 | mog_home | gnite |
10:46.34 | trixter | the state of california is 90% rural by geography, but 90% of the 35M people live in urban areas which comprise 10% of the geography |
10:46.43 | trixter | by definition that is what makes those areas urban vs rural |
10:47.16 | coppice | about 50% of the world's population lives in cities. that's nearly 100% in places like US, and something much lower in less developed places |
10:47.17 | trixter | night moggy |
10:47.38 | trixter | um what is nearly 100% in the US specifically |
10:47.49 | trixter | the worlds population or the US population or ... |
10:48.00 | Dr_Ray | the same as 100% of people have cell phones |
10:48.11 | trixter | perhaps 100% of the US is urban? |
10:48.14 | trixter | it wasnt clear what you meant |
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10:48.46 | coppice | I think a very high percentage of the US population and certainly a high percentage of western europe live in cities. balancing that places like china are way below 50% |
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10:49.52 | trixter | the problem with the majority of people in any given society living in rural areas is that farms (whether dirt farming or livestock) takes a lot of land, which implicitly means that people cannot occupy the same per sq foot or however its measured density |
10:50.43 | trixter | the majority of the US is rural however, with the clear majority living in urban areas, NY, LA, CHI are the top 3 cities, that comprises over 10% of the US population in just those 3 cities |
10:50.49 | trixter | or close to 10% anyway |
10:51.15 | trixter | that same type of thing must occur everywhere there is a rural area, if it does not the area is not rural |
10:51.23 | trixter | it is rural because relatively few people live there |
10:51.46 | trixter | look at ireland, 50% live in dublin county, the other 2M people are spread throughout the country, a good chunk of the remaining live in cork county (I think 25% are in cork) |
10:51.47 | coppice | well, if you roam around rural US the homes are far apart. try that in china and homes are never far away. even the rural population is not that thinly spread. there are huge numbers there |
10:52.10 | trixter | discovery channel disagrees with your depiction of rural china |
10:52.23 | trixter | they have video footage to go on their side we have only your word |
10:52.37 | coppice | partly because the 1 child policy only applies to city dwellers |
10:52.51 | coppice | what does discovery channel say? |
10:53.19 | trixter | unless something changed recently the discovery channel disagrees with that too, becuase they showed rural clinics doing testing and showed some rural farming folk that were having issues cause their daughter ran away to avoid being tested |
10:53.24 | trixter | basically the parents get arrested too |
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10:54.00 | trixter | discovery channel showed footage of china which showed houses far apart, they showed large fields for crops and live stock |
10:54.15 | Dr_Ray | but coppice has been to 100% of china |
10:54.30 | trixter | they showed people plowing the fields with old mostly manual equipment (but did note that tractors and other higher tech stuff was coming in it just wasnt everywhere yet) |
10:55.20 | trixter | http://home.earthlink.net/~jackgconrad/travels/china/Rural_Rice2.jpg |
10:55.22 | coppice | i didn't say no rules applied. in the cities anyone with a little money can pay taxes and have more children (kinda, sorta). in the countryside there is a sorta 1 child policy, but you pull the "I am a farmer" card, and it doesn't apply. young girls who may not be able to claim to be a farmer have a problem |
10:55.24 | trixter | looks quite urban if you ask me |
10:56.20 | trixter | there are waiting periods before you can buy the right to have another kid, if your first was a girl or something, that was covered however even farmers had limits |
10:56.28 | coppice | you are seeing one small field. notice there is rather more than one farmhouse beyond it |
10:57.04 | frenzy | hi |
10:57.06 | trixter | http://www.lib.utexas.edu/maps/middle_east_and_asia/china_population_83.jpg |
10:57.09 | frenzy | I have issue with Asterisk |
10:57.21 | frenzy | its hogging CPU |
10:57.29 | frenzy | i've already disabled MOH from loading |
10:57.33 | frenzy | yet I face it |
10:57.50 | trixter | http://www.fao.org/DOCREP/005/AC801E/ac801e09.jpg |
10:57.53 | coppice | china is like 2 countries. 2-300M around the coast which have seem massive development. the interior with the rest of the people looking like the land what time forgot. |
10:58.39 | trixter | the fao.org population density map is pretty good I think anyway, looks better visually than the utexas.edu one |
10:59.47 | trixter | note the chineese girls with green eyes are on the western part of china, kinda up to the north a little |
10:59.59 | coppice | so, ignore the east, which is desert, and the population show few obvious urban hotspots. it really is very spread out |
11:00.08 | frenzy | * is hogging CPU |
11:02.09 | frenzy | ? |
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11:10.41 | *** join/#asterisk Frk2 (n=faraz@202.5.145.13) |
11:10.46 | Frk2 | anybody using asterisk 1.2? |
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11:10.51 | *** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com) |
11:10.57 | Frk2 | im having meetme distortion with anything other than Ulaw |
11:11.16 | delox99 | i have dificulties to load ztdummy |
11:11.35 | delox99 | it was loading ok before i recompiled my kernel |
11:11.55 | delox99 | now i just recompiled zaptel and make install |
11:12.13 | Frk2 | is this a known issue? |
11:12.22 | delox99 | do i need to do depmod or something? |
11:12.36 | delox99 | i had the problem a month ago |
11:12.51 | Frk2 | with meeetme? |
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11:16.49 | Frk2 | hmm |
11:16.52 | Frk2 | everybody sleeps :) |
11:17.08 | trixter | frk2: I know that some stuff has been patched for 1.2 in CVS, I dont know if that specifically is one of the things.. I do plan on instlaling it soon, infact I am specing out the next system as I type this |
11:17.27 | trixter | I never run software that was 'just released' cause there is always a few things that dont work right :) |
11:18.17 | coppice | Frk2: Someone was heavily reworking codec management in meet me, but i don't know if that got into 1.2 or not |
11:18.46 | coppice | trixter: well, that's your fault. you won't run it and test it :-) |
11:19.34 | Frk2 | well i need 1.2 |
11:19.40 | Frk2 | the ael stuff is out of this world |
11:19.44 | Frk2 | and realtime |
11:19.56 | Frk2 | and the new jitter buffer management |
11:19.58 | Frk2 | its great |
11:20.18 | Frk2 | except this screw up with meetme, which is essential for me, everything perfect |
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11:20.36 | coppice | the jitter buffer only works with IAX |
11:20.42 | Frk2 | thats fine |
11:20.45 | demetrio | hello |
11:20.53 | Frk2 | all my server to server comm is IAX anyways |
11:21.09 | coppice | someone is generalising jitter buffering now |
11:21.24 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
11:21.42 | santoshr | is there a way to patch asterisk1.2 to recognize inband dtmf |
11:21.47 | demetrio | how does ${UNIQUEID} work? I see that it's different even from inside the same call |
11:21.52 | trixter | if it hasnt happened already app_meetme needs to take a page out of the app_coinference playbook and do transcoding for each destination type once and only once.. the old (current?) way meetme did transcoding was to transcode for each channel even if its the same codec that was used for the previous channels.. very wasteful |
11:21.58 | trixter | hopefully that is getting fixed |
11:22.28 | Frk2 | crack |
11:22.35 | Frk2 | so i cant use meetme till patched :( |
11:22.46 | coppice | that's just a minor part of meetme's problems. that just affects efficiency, not the results :-) |
11:22.46 | Frk2 | i got the nightly cvs |
11:22.47 | trixter | well I dont know the status of it |
11:22.59 | trixter | that is an obvious performance tweak and I do know that 1.2 had performance as one of its core driving forces |
11:23.02 | Frk2 | tried 1.2 stable as well |
11:23.03 | trixter | so it might have happened |
11:23.19 | Frk2 | but gsm screws up REAL bad, so does ilbc |
11:23.28 | Frk2 | everything but ulaw screws up |
11:23.48 | coppice | what about alaw? does it fall apart with anything at all? |
11:23.54 | santoshr | benjk: asterisk 1.2 doest have suppot for inband.. is there a way (a patch ) or somthing tht can be done for asterisk to recognize it... |
11:27.56 | *** join/#asterisk kaypee (n=kunal@60-240-164-1.tpgi.com.au) |
11:28.53 | kaypee | hi all .... i am trying to "make" zaptel .... but get the following error: |
11:28.55 | kaypee | You do not appear to have the sources for the 2.6.11.12-xenU-rimu1 kernel installed. |
11:29.27 | kaypee | is there a way that i can get the kernel source? |
11:30.15 | {zombie} | sure, it should be provided by your linux distribution |
11:30.19 | frenzy | what is ADSI ? |
11:30.43 | kaypee | can i d/l the kernel source off the internet ? |
11:30.48 | {zombie} | of course |
11:30.57 | coppice | A Dumb Sort of Interface |
11:31.01 | {zombie} | just make sure you have the exact version and patch level your kernel was compiled with |
11:31.08 | kaypee | can i tell apt-get to install them for me ? |
11:31.19 | {zombie} | what distribution are you running? |
11:31.25 | frenzy | coppice: ? |
11:31.31 | frenzy | is it worth the load ? |
11:31.40 | coppice | Analogue Display Services Interface |
11:31.46 | kaypee | FC3 |
11:32.15 | frenzy | crypto.so ? |
11:32.21 | frenzy | Am slimming down * |
11:32.31 | frenzy | for some reason * hogs CPU after few hrs |
11:32.35 | {zombie} | kaypee: you should be able to, but no idea what FC3 calls it.. possibly kernel-source-2.6.11-somethingorother |
11:34.20 | kaypee | am attempting an apt-get .... will tell u how it goes |
11:34.30 | kaypee | thanks for ur help zombie |
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11:37.34 | frenzy | what is chan_modem.so used for ? |
11:38.16 | coppice | i thought chan_modem had been thrown out of 1.2 |
11:38.40 | frenzy | i'm using the old conf |
11:38.53 | coppice | maybe not a good idea |
11:39.30 | frenzy | why ? |
11:39.45 | frenzy | dont think has changes to that |
11:40.01 | coppice | because chan_modem must be from an old install if it isn't in 1.2 |
11:40.06 | frenzy | I havent been chan_modem.so |
11:40.21 | frenzy | am just messing with the module.conf |
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11:45.33 | mesut | err,a very newbie here so it may be a stupid question,but,ild like to know,if i can use voip without any voip hardware,i mean without ipphones etc... |
11:46.02 | trixter | yes |
11:46.08 | {zombie} | sure |
11:46.15 | coppice | probably the commonest way, in fact |
11:46.17 | trixter | software htat acts like a phone is called a softphone. |
11:46.17 | {zombie} | there's several softphones around |
11:46.29 | trixter | depending on the platform you are running there are choices even in the free ones.. |
11:48.36 | mesut | we have asterisk installed on the server,and started very new,even couldnt make it work behind nat yet |
11:48.50 | mesut | we used linksys products |
11:49.10 | trixter | asterisk also can use your soundcard or if you have a bluetooth headset and bluetooth in your asterisk server it can use that as well |
11:49.17 | mesut | but if it can be done in soft way,thatll be better of course,because prolly well need about 5000 of them |
11:49.36 | trixter | but those are typically not newbie installs because its far easier to get a free softphone somewhere and use that for your first time |
11:49.59 | trixter | well the downside of a softphone is that you have to use the soundcard, so you need good quality headsets |
11:50.16 | mesut | i see |
11:51.06 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
11:51.06 | trixter | sjlabs.com has sjphone which is pretty good and can be skinned for a particular setup making it trivial to deploy.. xten.com has xpro which is also pretty good (what I use on my pda) there are others but I have not had any experience with them |
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11:52.01 | mesut | http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1117044308483&pagename=Linksys%2FCommon%2FVisitorWrapper |
11:52.10 | mesut | this is what we used in our first tests |
11:52.38 | mesut | replacing that with softphones will really be great |
11:54.28 | {zombie} | you'll get better results using an ATA like the PAP2 than using softphones in my experience |
11:54.34 | {zombie} | and even better results using "real" IP phones |
11:54.46 | {zombie} | esp if you're going to try and support thousands of them |
11:55.36 | frenzy | Nov 26 06:56:00 WARNING[4259]: file.c:583 ast_readaudio_callback: Failed to write frame |
12:00.04 | trixter | frenzy: I see that if the remote end dies |
12:00.59 | frenzy | hmm |
12:02.58 | mesut | {zombie}, hmm,we can make it optional i think...yes that sounds good to me |
12:03.07 | mesut | g2g thank you guys |
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12:15.00 | Jpc_Lgc | hi all |
12:15.06 | jaristizabal | hi |
12:15.26 | Jpc_Lgc | i am search a lot of answer at y question ;=) |
12:15.38 | Jpc_Lgc | i have buy for test my asterisk a Cisco IP Phone 7910 |
12:16.03 | Jpc_Lgc | anyone know the process for configuring it ? |
12:17.29 | Jpc_Lgc | you sleep ??? hihi ;=)) |
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12:17.51 | *** mode/#asterisk [+o denon] by ChanServ |
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13:11.46 | kernoman | im in uk using grandstream handytone 486 and when i make a call and hang up the phone immediately rings me back - why is this? |
13:14.11 | tzafrir_home | do you really hang up? |
13:18.47 | coppice | so, you have a phoney call problem |
13:22.33 | kernoman | well i put the phone back on the hook... |
13:22.52 | kernoman | how do i fix a phoney call problem? |
13:23.06 | tzafrir_home | set verbose 3 |
13:23.19 | tzafrir_home | what exactly do you see in the CLI? |
13:26.29 | kernoman | weird, even though I hung up my home phone (the one i rang) kept ringing???? |
13:27.19 | kernoman | that didnt read well, what i meant to say is that even though i hung up the phone my home phone (the number i rang) kept ringing |
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13:33.04 | kernoman | anyone? |
13:35.43 | RaYmAn-Bx | sounds like NAT issues |
13:36.36 | coppice | That Nat Issues guy is a real pain to everyone in the VoIP business :-) |
13:36.47 | backblue | does asterisk 1.0 to 1.2 have changes with nat and call termination stuff? in 1.0 sometimes it does not stop the call, and i have calls for hours. |
13:37.50 | kernoman | if i use xlite to call out it hangs up fine however using an analog phone plugged in to a grandstream handytone 486 it does not??? |
13:39.43 | kernoman | i take it the response from coppice was to me?? |
13:40.56 | coppice | it was just as whimsical comment to anyone out there :-) |
13:45.52 | kernoman | sorry i meant RaYmAn-Bx |
13:48.32 | Joe2000x | why i dont have the command "realtime" in my CLI? im using asterisk 1.20.. |
13:48.37 | Joe2000x | can anybody help? |
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13:57.34 | Gourou_fou | salut tout le monde |
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14:05.02 | endre | hi there |
14:05.15 | Gourou_fou | :) |
14:09.34 | Joe2000x | can anybody help me? T_T |
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14:13.34 | frk2 | man- what is WRONG with meetme in asterisk 1.2??? |
14:13.45 | frk2 | nothing but ulaw works :( anybody having this issue other than me? |
14:14.07 | frk2 | sorry to make a rude entry :) just wanna know if this is my issue or a known problem |
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14:16.53 | frk2 | anybody? |
14:16.55 | *** join/#asterisk litage_ (i=litage@CPE-203-45-246-72.qld.bigpond.net.au) |
14:17.02 | litage_ | if *-server-1 only uses h323 to speak to other * servers, can the ATAs, IP phones, softphones, etc that connect to *-server-1 speak iax or sip, or is it best to use the same protocol all the way through? |
14:17.43 | frk2 | i just wanna know if SOMEBODY is running 1.2 w/meetme and gsm without issues |
14:17.44 | *** join/#asterisk jaristizabal (n=jaristiz@69.79.133.185) |
14:20.05 | litage_ | frk2: questions with "anybody" usually don't get answered. try a more direct or specific question |
14:21.43 | frk2 | litage- the question was specific. I'm running asterisk 1.2 - everything works fine, except meetme.. if all conference people use ulaw- all is well.. but if one uses gsm- exceptional distortion happens |
14:22.08 | frk2 | i just wanted to know if this is specific to MY setup or not |
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14:37.46 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
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14:48.02 | h3x | hahahahahahah |
14:48.14 | h3x | theres a John Holmes MD in the phone book |
14:48.24 | h3x | that must be rough |
14:49.12 | litage_ | h3x: ? |
14:50.42 | h3x | you know the porn star that died from aids |
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14:57.56 | litage_ | 'fraid not.. |
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15:08.38 | backblue | anyone here use mcc? |
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15:11.19 | litage_ | backblue: mandrake control centre? |
15:12.08 | wasim | maryleborne cricket club |
15:14.09 | backblue | litage_: mcc = biling system? |
15:14.13 | backblue | http://www.paskambink.lt/mcc/ |
15:14.22 | litage_ | ah =P |
15:14.49 | litage_ | backblue: rather than asking if people are using something, you'll get a better response if you ask a real question |
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15:15.57 | Guest^DJ | hi guys, 1 quick question can 1 * box handles 300 analog phones on channel banks |
15:18.59 | ikarus | Guest^DJ: depends on what it has to do ? |
15:19.00 | backblue | litage_: but i dont have it. why should i ask "a real question" if i just want to share information and talk. |
15:19.03 | ikarus | what else |
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15:20.09 | litage_ | backblue: i'm just trying to be helpful. my experience on irc over the past years is that questions such as "does anyone use xyz?" don't get answered |
15:20.40 | litage_ | is there a difference between G729 and G729a? |
15:20.46 | Dr_Ray | Guest - I would do 3 boxes of 96 ports each |
15:21.02 | ikarus | Guest^DJ: so, if it is just switching between different analog phones, quite likely it can handle it without difficulty |
15:21.05 | Dr_Ray | but that is just my take on it |
15:21.07 | znoG | 300 phones on one box |
15:21.21 | znoG | heh, and here I am thinking it's a bit much to have 20 phones on one box |
15:21.22 | znoG | :) |
15:21.34 | Dr_Ray | well, it depends, a hotel would not have all 300 in use |
15:21.47 | Dr_Ray | I get buy on skimpier hardware |
15:21.50 | ikarus | Guest^DJ: but partitioning might be a good thing, it would enable you to optimise the setup later if you want to add more |
15:21.54 | Guest^DJ | ikarus: just handling extensions to extensions |
15:21.54 | Dr_Ray | er, by |
15:22.16 | Guest^DJ | Dr_Ray: this is a set up for a local hospital |
15:22.33 | Dr_Ray | then I would probably put 3 boxes in |
15:22.34 | ikarus | Guest^DJ: that should work |
15:22.53 | ikarus | Guest^DJ: but I would divide it per 100 just so you have a multi-box setup already and scaling is more easy |
15:23.11 | Guest^DJ | i know it sounds lazy, but planning to use asterisk@home |
15:23.28 | Dr_Ray | asterisk at home is more work than real asterisk |
15:23.33 | Dr_Ray | imo |
15:23.34 | ikarus | Guest^DJ: don't |
15:23.41 | marv | I thought asterisk@home wasn't supported here? |
15:23.43 | ikarus | Guest^DJ: Asterisk is easy enough by itself |
15:24.00 | ikarus | And adding abstraction only makes debugging and modifications harder |
15:24.01 | Guest^DJ | ok |
15:24.30 | Guest^DJ | hrmm |
15:24.31 | ikarus | Guest^DJ: your setup without voicemail, etc would be no work at all to do plain |
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15:25.06 | Dr_Ray | i'd worry about that many cards in one box |
15:25.49 | ikarus | Dr_Ray: channel banks |
15:25.58 | Dr_Ray | is there a cheap way to hook up that ds3 card that digium uses |
15:26.05 | Dr_Ray | to a channel bank |
15:26.26 | Dr_Ray | 30 users is still 4 cards |
15:26.27 | ikarus | Guest^DJ: an idea might be to use a VoIP (SIP possibly) channelbank |
15:26.29 | Dr_Ray | er, 300 |
15:26.37 | ikarus | Dr_Ray: VoIP channel banks |
15:27.00 | Dr_Ray | hmm.. teh adit 600 can do that? |
15:27.18 | Guest^DJ | voip channel banks ? hmm never heard of it, have to do some research |
15:27.33 | Dr_Ray | they have a g.729 card for the adit 600 |
15:27.57 | ikarus | Dr_Ray: I would use mu or alaw (depending on the country) |
15:28.06 | ikarus | the bandwidth on a dedicated network isn't too much |
15:28.07 | Guest^DJ | heard Rhino is the easiest to setup |
15:28.29 | ikarus | And using a channel bank avoids most of the hardware problems with things like IRQ's |
15:28.50 | Dr_Ray | got a brand name for a voip channel bank? |
15:29.10 | ikarus | Dr_Ray: there are some mentioned on the wiki |
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15:39.52 | arcy | if i want a lot of FXS for analog phones why should i buy an expensive TDM (eg TDM2400P with full modules) instead of of many ATA adapters (2 FXS each)? It's almost 3 times the difference in money for 24 FXS ports. |
15:40.03 | arcy | i am sure i am saying something stupid here, but what? |
15:40.52 | ikarus | arcy: ATA's might be more tricky to wire up and maintain (but I agree, the cards are way overpriced) |
15:41.09 | arcy | but it would work, right? |
15:41.15 | ikarus | Yep |
15:41.29 | arcy | thank you ikarus. |
15:41.40 | litage_ | if your asterisk server uses only h323 to connect to its provider, will asterisk performance and/or voice quality degrade if the ATAs, IP phones, and softphones use sip or iax? |
15:42.32 | Guest^DJ | ikarus: ATA vs channel banks on analog phone, which is easier to configure and maintain |
15:42.33 | ikarus | litage_: if they use the same codec, NO |
15:43.36 | ikarus | Guest^DJ: channel banks with VoIP are like HUGE ATA's, so they are easier to configure as they condense alot of configuration issues and ofcourse less parts to go wrong (imagine 150 ATA's each with it's own power brick) |
15:43.49 | litage_ | ikarus: so as long as everything's using the same codec, it absolutely will _not_ matter if ATA->* is IAX, and *->provider, *->other*, etc are H.323? |
15:44.18 | Guest^DJ | ikarus: you are right |
15:44.21 | ikarus | litage_: for 99.9%, yes (there might be some out of band signalling issues, but those don't degrade quality) |
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15:47.04 | znoG | arcy: i'm going to be using 10 x ATAs for 20 extensions, simply cause any TDM card is a rip off |
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15:47.15 | litage_ | ikarus: in that situation where everything "upstream" from your * box is using H.323, are there any advantages in using IAX instead of SIP or H.323 for ATA/IP phone/softphone->* ? or H.323 instead of the others? |
15:47.24 | znoG | if 10 x ATAs were about $300 cheaper than a 24 port channel bank, i'd go the channel bank |
15:47.36 | znoG | but since 10 x ATAs are half the price (or 1/3) of a channel bank, no way! |
15:47.47 | arcy | znoG, i plan to do something like that for the same reason. But since i have never setup an asterix box before i am asking to make sure |
15:48.58 | ikarus | litage_: IAX is mainly important for trunks |
15:49.09 | ikarus | litage_: and H.323 is obsolete |
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15:49.55 | ikarus | znoG: hmmmmmmm, can you give me the cost of your ATA's ? (I wonder if it would be worth adding ATA support to my little box) |
15:50.07 | Guest^DJ | znoG: i have approx 300 ports to support |
15:50.13 | arcy | the way i am calculating it is that with small ATAs i pay 45 Euros per line while with TDMs or large ATAs i pay 125 Euros per line. So it's a _big_ difference |
15:50.59 | litage_ | ikarus: i'm unfortunately in a position where everything upstream from me is H.323 . taking that into consideration, should my ATAs, IP phones, etc use SIP, IAX or H.323? i'm wondering if one of those protocols will perform better in this situation, or if it doesn't matter |
15:51.37 | ikarus | litage_: SIP or IAX (depending on price and trunking nature), H.323 is still obsolete, so don't use it unless you have t |
15:51.56 | ikarus | arcy: 45 euro per line, k, I should be able to beat that |
15:52.38 | litage_ | ikarus: why do you say "depending on price and trunking nature" for IAX? does one need a license to use IAX? |
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15:53.07 | ikarus | litage_: no, but devices supporting IAX are more expensive then those supporting just SIP |
15:53.08 | arcy | really? I found Sipura SPA2002 for 90 Euros (2 FXS each). |
15:53.19 | ritesha | MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. |
15:53.25 | chapeaurouge | arcy, where |
15:53.27 | chapeaurouge | ? |
15:53.28 | ikarus | arcy: I am working on a custom channel bank system |
15:53.33 | ikarus | chapeaurouge: siptronic.com |
15:53.39 | chapeaurouge | k |
15:53.54 | ritesha | i checked my paswd and it looks okay...wonder if anyone has some clues? |
15:53.55 | arcy | www.inkshop.gr (sorry, it's Greek ) |
15:54.09 | trixter | circuit city has a post black friday sale, toshiba $200 laptop doesnt much matter what it is, its a backup system with battery backup :) |
15:54.24 | litage_ | ikarus: ah i see. so unless we'll gain some advantage from using IAX-capable ATAs and IP phones, we might as well use SIP, right? |
15:54.24 | znoG | Guest^DJ: well, i would definately NOT suggest the multi-ATA solution for your environment |
15:54.30 | ikarus | litage_: yes |
15:54.33 | znoG | Guest^DJ: the multi-ATA solution is good maybe for up to 30 extensions |
15:54.38 | litage_ | thanks for that info, ikarus |
15:55.34 | ritesha | MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info....please help....any clues? |
15:55.46 | Guest^DJ | znoG: thanks |
15:56.33 | Guest^DJ | still sourcing the type of channel banks, any recommendation ? |
15:56.55 | znoG | Guest^DJ: besides, if you have 300 extensions, it's likely you have the resources ($$) to afford a channel bank |
15:57.06 | znoG | with 20 extensions like I'm about to support, budget is far more limited |
15:57.24 | chapeaurouge | how are the Granstream handytone ATA in general? |
15:57.31 | znoG | i don't like them much |
15:57.40 | znoG | rather a PAP2-NA or Sipura any day |
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15:57.44 | chapeaurouge | ok |
15:57.52 | znoG | personal preference anyway, don't listen to me :) |
15:57.54 | Guest^DJ | chapeaurouge: i used Linksys, so far so good |
15:57.56 | ikarus | Guest^DJ: Allied Telesyn has one I know of, btw, why don't you upgrade the on desk phones to VoIP, phones come in at under 75 euro, and a decent set of Ethernet equipment should be cheap |
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15:58.11 | ikarus | Guest^DJ: or is the wiring of below cat 3 quality ? |
15:58.14 | chapeaurouge | hello [TK]D-Fender |
15:58.20 | [TK]D-Fender | y0 |
15:59.08 | [TK]D-Fender | I'm on damage recovery today since I screwed my server last night.... |
15:59.08 | Guest^DJ | ikarus: upgrade to Voip will involve new wiring, and there are 6 floors |
15:59.11 | ikarus | Guest^DJ: but if the current wiring is of cat3 qualit or higher 10Mbit Ethernet would work just fine |
15:59.16 | ikarus | or is it really that bad |
15:59.52 | chapeaurouge | ikarus, everywhere I read, they advise no less than cat5 |
16:00.05 | Guest^DJ | i have not check on the wiring, but i belive those wires are at least 3 yrs old |
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16:00.10 | ikarus | chapeaurouge: cat3 is the official design spec for ethernet |
16:00.24 | chapeaurouge | i know, where getting old. |
16:00.28 | ikarus | Guest^DJ: that is not the issue, cat3 or cat5 is pure wiring quality (in use since long ago) |
16:00.29 | chapeaurouge | s/where/but |
16:03.34 | Guest^DJ | i will need to check the cabling |
16:04.42 | ikarus | Guest^DJ: but what is the alloted amount of money per phone, I am just looking up the data and erm, the pricing is such that it would easily exceed 120 euro per phone if you go analog |
16:06.59 | ikarus | And most of those are not VoIP channel banks, but export a E1/T1 interface, which still requires an evilly expensive card in a box |
16:07.22 | ritesha | MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. |
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16:09.13 | Guest^DJ | ikarus: no set budget yet, boss is comparing asterisk vs a ericsson box |
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16:09.37 | Guest^DJ | i do not know how much ericsson cost |
16:09.41 | _shad_ | I'm trying to create sms notifications for incoming calls, but when people call with a & in the callerid, bash takes it literally. It doesn't matter if I enclose the string in " or ' . Am I missing something? |
16:12.43 | Guest^DJ | any one tried zhone channel bank, any good ? |
16:16.26 | ikarus | Guest^DJ: using TDM2400p cards (no channel bank) it would be around 2000 euro per 24 phones and you could probably safely use up to 4 (maybe 5) per box |
16:17.36 | chapeaurouge | ikarus, 4 per box.. better be a _big_ box :) |
16:17.48 | ikarus | chapeaurouge: the physical box, yes :) |
16:18.02 | ikarus | chapeaurouge: actually I consider it a crappy option |
16:18.47 | ikarus | chapeaurouge: I was sure there was a 24, 30, 48 or 60 port ATA for less then that price per port (including the computer, you need a decent box with that card) |
16:19.02 | chapeaurouge | yea |
16:20.02 | ikarus | But appaerently there is a market for such equipment, but no such equipment availible *grin* |
16:20.42 | Guest^DJ | i would need 4 box each with 4 TDM2400p |
16:21.31 | ikarus | Guest^DJ: yep, expensive little joke |
16:21.43 | ikarus | consider a good enough box might also be 2000 euro a piece |
16:21.49 | litage_ | ikarus: from what you were saying earlier, i got the impression that you're not a fan of H.323 . might i enquire as to why that is? |
16:22.18 | ikarus | litage_: because it doesn't allow signalling to travel seperate paths from data, while it's priority is much lower |
16:23.07 | litage_ | ikarus: what do you mean by "separate paths"? |
16:23.12 | ikarus | Guest^DJ: another alternative might be a channel bank, and a few Digium Wildcard E1/T1 cards in a single box, but that might not work |
16:23.15 | ikarus | litage_: SIP |
16:23.28 | Guest^DJ | ikarus: yes, that was my original plan |
16:23.40 | ikarus | litage_: SIP/RTP allowes the RTP to travel a P2P path, seperate from the SIP traffic |
16:23.52 | Guest^DJ | ikarus: maybe i could source some used channel banks from ebay |
16:24.02 | ikarus | Guest^DJ: it depends on how well behaved the Wildcard's are |
16:24.11 | litage_ | ikarus: ah i see |
16:24.37 | Guest^DJ | ikarus: is going to be a night mare |
16:24.39 | ikarus | litage_: this causes lower loads on a SIP proxy, etc |
16:24.45 | ikarus | Guest^DJ: that it is |
16:25.08 | Guest^DJ | better off just buy a off the shelf PBX |
16:25.28 | ikarus | Guest^DJ: not like |
16:25.29 | ikarus | ly |
16:25.33 | ikarus | Guest^DJ: but consider new channel banks aswell, just calculate a few possibilities (including the expensive allied VoIP channel banks), it will probably still be cheaper then |
16:25.40 | ikarus | the Ericsson box |
16:25.52 | ikarus | Considering how much more rip-off those companies are |
16:26.04 | ikarus | for 16 phones they tried to charge us 5000 euro |
16:26.17 | ikarus | I solved it for 1000 using Asterisk |
16:27.29 | Guest^DJ | Rhino cost approx USD 1,400. i would need like 13 of them |
16:27.38 | Guest^DJ | 4 asterisk box |
16:27.59 | Guest^DJ | 4-5 wildcard E1/T1 cards |
16:28.02 | litage_ | why would you want an ATA to have more than 1 ethernet port? |
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16:33.09 | ritesha | MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info....please help....any clues? |
16:33.39 | litage_ | ritesha: did you check the logs? |
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16:42.53 | JonR800 | litage_: maybe it's a port for a pc? |
16:43.16 | ritesha | hmm..let me check the log but how do I set the port? |
16:43.56 | litage_ | JonR800: they're for any ethernet device. i'm just wondering why you'd want to attach 2 ethernet devices (whether they're computers, switches, routers, etc) to an ATA |
16:44.14 | litage_ | ritesha: off the top of my head, /etc/mysql/my.cnf |
16:44.38 | JonR800 | litage_: i dunno.. i could see one, but two seems sort of odd. |
16:45.29 | litage_ | JonR800: my thoughts exactly. but i've noticed that MANY manufacturers sell ATAs with 2 ethernet ports |
16:46.00 | RaYmAn-Bx | a fair amount of ATA's has some sort of simple QoS built in which means you just stuff the ATA before your computer when connecting to the internet and your phone won't be affected by your inet traffic |
16:46.16 | trixter | many of the cheaper ones have a bit of POS built in |
16:46.17 | trixter | :P |
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16:48.50 | netwetrust | Hi, any have probe the rapid configuration on asterisk of xorcom.com |
16:50.02 | litage_ | RaYmAn-Bx: ah, so maybe the ATAs with 2 ethernet ports are implied to be "daisy-chained" between your comp and your comp's router/switch/hub.. |
16:50.46 | RaYmAn-Bx | litage_: often, yes..I believe SPA-1001 has that |
16:51.33 | litage_ | RaYmAn-Bx: the SPA-2000, 2002, and 2100 have 2 ethernet ports, but the 1001 has 1 |
16:51.44 | RaYmAn-Bx | hmm |
16:51.54 | RaYmAn-Bx | I guess it's 2100 then perhaps :) |
16:51.58 | litage_ | ack sorry, only the 2100 has 2 ethernet ports. all of the 2* series have 2 FXS ports |
16:52.07 | RaYmAn-Bx | yeah |
16:52.50 | litage_ | do the extra features that the Cisco ATA-186 (or 188) have justify the price? (~$200) |
16:56.51 | ritesha | hmm...here is the mysql.log.....http://pastebin.ca/31321 |
16:57.12 | ritesha | dont' see any issue. looks like mysql is not even getting the connect request |
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17:00.14 | litage_ | ritesha: increase msyql debug/logging |
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17:21.02 | jahani | hi |
17:21.08 | jahani | is there any asterisk Subscriber Management System ? |
17:22.05 | rking | jahani: forgive my ignorance - what sorts of things would it do? |
17:23.56 | jahani | prepaid card |
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17:24.17 | rking | jahani: ahh... hrmm... |
17:24.26 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
17:25.00 | iDunno | bbz: please fuck off and die. or at least stop sending the same fucking advert to the same channel. |
17:25.29 | iDunno | :) |
17:26.23 | rking | jahani: so, i'm obviously a complete newbie, but it does seem like i've seen someone do something like that with asterisk... i'm now looking through the asterisk/apps/ dir to see if anything rings a bell. |
17:26.49 | jahani | ok thank you |
17:28.15 | sm7syx | Hi, can I play an announcement to the calling party ? In the 'ringing' sequence... |
17:30.01 | rking | sm7syx: how is that different then answering it and playing a message? |
17:30.34 | rking | sm7syx: just that the other end is getting a ring while the message is playing? |
17:32.08 | rking | jahani: http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications # does this help? |
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17:47.12 | rking | this stuff is really fun. |
17:47.34 | rking | i am going to have the baddest answering machine in town. |
17:56.34 | litage_ | rking: if only everyone had video phones, then you could REALLY make an interesting message for your answering machine =P |
17:57.11 | rking | hrmm.. there's ascii art porn... i wonder if i could make .gsm porn. |
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17:57.55 | marv | tt-pr0n? |
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18:02.24 | Brijn | Good morning all |
18:02.29 | Rawplayer | RE |
18:04.30 | Brijn | Something weird happened here. Everything was working fine, but suddenly all voceimail/MoH sounds are very slurred. Played at roughly 1/10th of the speed |
18:04.46 | rking | what is a recommended way, under Linux+(alsa/oss), to record top-quality sounds? i did arecord foo.wav; sox foo.wav foo.gsm; and it's crackletastic |
18:04.58 | Brijn | Calls are OK, retrieved latest cvs, removed modules, recompiled, doesn't make a difference |
18:05.01 | h3x | Brijn: did you load res_alcohol_vodka.so ? |
18:05.19 | Brijn | No, but I did try asperin.so |
18:05.44 | rking | "Hello, my name is *, and I'm an alco.so.holic." |
18:05.48 | h3x | asprin isnt a shared library |
18:06.42 | Brijn | h3x: ahh, could that be the problem? But * had a good nice, long sleep, that should have helped as well? |
18:07.05 | Brijn | From just a headache, the sound shouldn't go weird |
18:07.25 | h3x | i think you need that anti hangover medication |
18:07.26 | h3x | hehe |
18:07.38 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
18:07.58 | Brijn | :) |
18:08.40 | Brijn | If people get my voicemail now, they will probably think I was drunk when I left the message.. Yooooooouuuuuuuuu haaaaaavvvvveeeeee reaaaaaaaaacheeeeeed |
18:09.46 | Brijn | The voicemails that are left are OK, the mail version is perfectly fine. But played over my Polycom or Softphone, it sounds horrible |
18:10.15 | Brijn | And it worked perfectly for a while, I don't know what changed :( |
18:15.59 | rking | Brijn: and it sounds the same with VOIP? |
18:18.26 | Brijn | rking: How do you mean? Calls over the incoming SIP get the same weird sound |
18:18.38 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
18:18.53 | Brijn | Local (same net) both phone and softphone get the wrong sounds, calls are ok |
18:19.25 | rking | Brijn: weird stuff =\ i wish i knew enough to help you |
18:20.22 | Brijn | I'll check in tonight again, maybe one of the supermasters is in/awake/willing to help |
18:27.44 | asterboy | Anyone successfully gets order Atacomm? |
18:28.42 | asterboy | me english no good tdoay |
18:28.58 | asterboy | hung |
18:29.02 | asterboy | ovt |
18:34.24 | kernoman | im using version 1.2 and when i add load => chan_zap.so to my modules.conf so i can use my x100p card i get this error: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call any ideas why? |
18:38.06 | JunK-Y | kernoman: u need res_features too |
18:38.26 | *** join/#asterisk }btorch{ (n=btorch@c-69-180-105-139.hsd1.fl.comcast.net) |
18:39.55 | }btorch{ | If a PBX PRI line works connected to the PSTN line then shouldn't the PBX PRI work when connected to * |
18:40.23 | }btorch{ | do I need something special like PBX -> CSU -< * |
18:45.31 | Rez | you need a crossover t1 cable |
18:45.45 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
18:45.51 | benjk | is mog here? |
18:48.42 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
18:51.23 | }btorch{ | Rez that's it ? |
18:52.22 | }btorch{ | So I assume that the cable that comes with the legacy pbxs are not crossover |
19:00.10 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
19:03.26 | morale | can someone help me with asterisk? i "thought" i had it all working and now my PSTN line is gone.. when i pickup i get a dialtone, if i dial any digit it hangs up right away |
19:03.32 | morale | <PROTECTED> |
19:03.32 | morale | <PROTECTED> |
19:06.45 | arcy | Is this any good? http://www.voip-info.org/wiki/view/AG-468 |
19:07.08 | *** join/#asterisk Dr-Linux (n=loyal@202.59.75.58) |
19:08.23 | *** join/#asterisk alephcom (n=Miranda@207.34.97.130) |
19:11.56 | Dr-Linux | hi alephcom |
19:12.46 | *** join/#asterisk m160858 (n=jsaenz@200.89.12.46) |
19:13.49 | *** join/#asterisk [chico] (n=user@p54913ACF.dip0.t-ipconnect.de) |
19:14.13 | alephcom | Greeting |
19:18.32 | tzafrir_home | morale, to what context does the line go? look at 'zap show channels' |
19:20.26 | *** join/#asterisk Entegrity (n=Entegrit@c-24-34-120-110.hsd1.ma.comcast.net) |
19:20.38 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
19:20.39 | tzafrir_home | kernoman, are you sure that chan_zap was actually built? look at the date of the file in /usr/lib/asterisk/modules |
19:21.19 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
19:22.45 | *** join/#asterisk figits (n=me@217.240.33.65.cfl.res.rr.com) |
19:26.00 | morale | tzafrir_home: 2005-11-26 12:20:40 WARNING[32083]: channel.c:2313 set_format: Unable to find a codec translation path from g729 to gsm |
19:26.15 | morale | any idea what that means? i got outbound calls working, inbound calls don't work though |
19:26.22 | file | that error is pretty self explanitory |
19:26.29 | mog_home | do you have g729 codec installed |
19:26.32 | morale | yes |
19:26.36 | morale | i have it licensed too |
19:26.44 | mog_home | do |
19:26.49 | mog_home | show g729 |
19:26.57 | mog_home | or something like that do you have available lic. |
19:28.09 | *** join/#asterisk t0ke (n=toke@201.Red-81-36-121.dynamicIP.rima-tde.net) |
19:28.17 | t0ke | hi |
19:28.21 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
19:28.37 | morale | ooh.. i don't think i have the module installed now since i did a debian pkg upgrade |
19:28.51 | mog_home | dndndndndnaaaa |
19:28.52 | Telamon | What's it called when two SIP clients send the RTP packets directly to each other? reinvite? |
19:28.56 | mog_home | the truth reveals itself |
19:29.00 | mog_home | reinvite |
19:29.02 | t0ke | anyone know if is possible to use asterisk business edition+astbill ? |
19:29.24 | mog_home | yes, i believe so |
19:29.42 | mog_home | may not be very easy to get it to work together |
19:29.51 | mog_home | but be is very similar to 1.2 |
19:31.06 | Telamon | Damn... Okay, second question. :) I'm using siproxd to proxy SIP phones behind NAT and they can call each other when I send them through the zaptel interface, but on direct extension-to-extension calls, I get no voice. The call connects fine, just no audio. I have reinvite disabled in the user's contexts, so any ideas what the problem is? |
19:31.26 | morale | <PROTECTED> |
19:31.36 | benjk | mog are you here? |
19:31.41 | morale | when i used to start asterisk i would say something about my license |
19:32.11 | benjk | you may want to put entries for the Digium cards on this page ... |
19:32.12 | benjk | http://www.voip-info.org/wiki/view/PSTN+Interface+Hardware+for+Computer+Systems |
19:34.02 | morale | <PROTECTED> |
19:34.05 | morale | <PROTECTED> |
19:34.08 | morale | <PROTECTED> |
19:34.08 | mog_home | maybe |
19:34.10 | morale | there we go |
19:34.54 | mog_home | you want to add it for me? |
19:35.57 | morale | yay it works now |
19:36.36 | benjk | mog, I have run out of steam today |
19:36.55 | benjk | almost 5 am |
19:37.27 | mog_home | eep |
19:37.38 | mog_home | yet another problem with japan -_- |
19:40.45 | robl^ | mmmm.. cookies |
19:41.52 | mog_home | man i could go for a cookie |
19:41.56 | mog_home | i have 0 food at apt. |
19:42.32 | robl^ | I have bagele and cream cheese.... :) |
19:42.46 | mog_home | i have bread and penut butter |
19:42.49 | mog_home | but no jelly |
19:42.50 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
19:42.50 | robl^ | bagels, even |
19:42.58 | pifiu | heyy |
19:43.07 | *** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net) |
19:43.16 | mog_home | QWELL!!!!!!!!!!!!!!!!!! |
19:43.24 | robl^ | QWELL! |
19:43.27 | Qwell[] | omg_home! |
19:43.28 | pifiu | anyone know how to use macros for the config files so its easier to configure? |
19:43.28 | file | yay Qwell[] |
19:55.19 | morale | where can i get a 1-800 number? |
19:55.38 | *** join/#asterisk SwK (n=SwK@12-219-151-128.client.mchsi.com) |
19:58.50 | mog_home | mmmm jelly |
19:59.15 | benjk | morale: NuFone |
19:59.33 | benjk | http://www.nufone.net |
19:59.46 | brimstone | will nufone allow you to call your 1-800 number from a payphone? |
20:00.11 | Qwell[] | yes |
20:01.11 | robl^ | as long as teh payphone allows it! I have seen some non-Bell operated payphones that require you to pay for any call, including toll free (at local call rate) |
20:01.34 | brimstone | thiw was a wally world phone |
20:01.38 | brimstone | i think bell operated |
20:02.53 | morale | Yes, we are seriously working on this mess. |
20:02.54 | morale | Patience is a virtue. |
20:02.55 | morale | heh |
20:03.16 | benjk | I have seen payphones in the UK which required you to throw money in to make a toll-free call, but they didn't charge you for the call |
20:03.48 | robl^ | anyone have any good/bad comments on experiences with Snom 320/360 phones? I am really tempted to buy some for a new project. |
20:04.49 | robl^ | in the uk, you have to pop in coins every 20 secs of a call it seems.. even if you call only 2 km away ;-) |
20:05.06 | CleanerX | ~docs |
20:05.08 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
20:05.12 | gambolputty | pay phones might as well have card readers |
20:05.18 | CleanerX | follow the third link, robl^ |
20:05.22 | benjk | some payphones do |
20:05.41 | morale | hmm.. i just signed up with nuphone.. do i have to wait to request a DID? |
20:05.52 | Qwell[] | morale: didn't used to |
20:06.03 | robl^ | CleanerX: thanks.. already read the wiki's info on the Snom phones.. ;) |
20:06.18 | alephcom | morale: I think you do now. I looked a few days ago and it did not look like they automatically provisioned them. |
20:06.49 | morale | hmm.. do i have to email them to request a 1-800 number? |
20:07.04 | robl^ | when I got DIDs from NuFone (a VERY VERY long time ago), it took about a day to get the numbers |
20:07.31 | morale | erm.. it only provides a US48 1800 number. |
20:07.34 | morale | not canada? |
20:07.46 | morale | as least i only put 2$ on the account |
20:07.53 | alephcom | No, Canadian toll free rates are high. |
20:08.13 | morale | ah |
20:09.02 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
20:09.35 | benjk | ~seen shido6 |
20:09.41 | jbot | shido6 is currently on #asterisk (12h 14m 57s). Has said a total of 3 messages. Is idling for 11h 52m 3s |
20:09.52 | alephcom | I can't think of anybody right now who will automatically provision CDN toll frees. I have a few, I might even have an extra one but they are worth around $0.06CDN. |
20:09.54 | benjk | ~seen JerJer |
20:09.55 | jbot | jerjer <n=JerJer@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 15d 12h 12m 24s ago, saying: 'not really a debian specific question, but someone here should know - Can i merge partitions in Linux? like my / was created way too small and i would like to blow away another partiton and start over, but one issue is I am currently not ... |
20:09.57 | robl^ | ~seen atacomm |
20:09.58 | jbot | atacomm <~dan@69.54.45.98> was last seen on IRC in channel #asterisk, 181d 17h 5m 1s ago, saying: 'lol, well that would be the more specific place, but last time i chceked there's alot of talk about bugs in here, lol...'. |
20:10.39 | benjk | well, morale, the NuFone guys are not around, but if you hang out for a while they may show up |
20:10.39 | robl^ | 181 days?!?!? Gee, I realy have been out of touch |
20:11.06 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
20:11.10 | morale | alephcom: thanks |
20:11.35 | morale | has anyone got postgresql voicemail storage working? |
20:11.47 | alephcom | bbz: lol, how many times have you posted that? |
20:11.58 | benjk | bbz is a spambot |
20:12.11 | benjk | and my /ignore still doesn't work :-( |
20:12.30 | robl^ | mmmmm.. spam... fried with cheese on white bread! american cuisine! :) |
20:12.59 | benjk | white bread, eeek |
20:12.59 | alephcom | lol. Ok, I'll ignore him. |
20:14.11 | robl^ | at least he's not spamming for Viagra, ink jet refills, or home mortgages |
20:14.49 | benjk | i'll have sourdough bread instead |
20:14.55 | benjk | home baked |
20:15.10 | benjk | with Guinness sourdough |
20:15.14 | benjk | yummy |
20:15.40 | robl^ | mmmm.. that sounds good... |
20:15.50 | robl^ | esp the Guinness part |
20:16.13 | benjk | Yes, you use Guinness instead of water to make your sourdough |
20:16.22 | benjk | gets you a very tasty bread |
20:16.46 | benjk | have to let the Guinness go stale and room temperature though |
20:17.01 | robl^ | I knew you could usae beer.. never thought of guineess for the sour dough though.. |
20:17.05 | benjk | the lacto-bacteria don't like the fizz and they don't like it cold either |
20:17.18 | robl^ | right |
20:17.39 | benjk | I have tried many different beers, Guinness makes the best tasting beer bread |
20:18.20 | robl^ | I've used newcastle before |
20:18.42 | benjk | http://www.sunrise-tel.com/misc/Guinness-Bread.JPG |
20:19.55 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
20:20.19 | gordonjcp | alephcom: nine times, by my counting |
20:20.29 | Qwell[] | more like about 30 |
20:22.14 | alephcom | What a waste. Oh, well. I better go help my wife. :-) |
20:22.38 | benjk | waste? |
20:23.39 | robl^ | alephcom: remember when she asks, you reply "No! it doesn't make your bum look big!" ;) |
20:25.46 | morale | when voicemail is working with psql, if it type 'show voicemail users' it should select the people from the database table right? all i am seeing is people in my voicemail.conf |
20:25.58 | alephcom | lol, I learnt that one already. :-) Later, everyone. |
20:26.49 | p1tst0p | exit |
20:37.21 | m160858 | hi |
20:37.26 | m160858 | benjk? |
20:37.30 | m160858 | are 2 there? |
20:37.49 | benjk | yes |
20:38.34 | m160858 | hi |
20:39.01 | benjk | hi |
20:39.04 | m160858 | hey, do yo know some provider of unlimited calls from USA & Canada? |
20:39.26 | benjk | USA, but Canada no |
20:39.41 | m160858 | others countrys? |
20:39.48 | m160858 | i used broadvoice |
20:40.09 | m160858 | but i have some problems, using many broadvoice accounts on my asterisk |
20:40.45 | m160858 | so |
20:40.54 | m160858 | now I am looking for new supplier |
20:41.02 | m160858 | provider |
20:41.31 | benjk | many people seem to have trouble with broadvoice |
20:41.45 | m160858 | yes? |
20:41.50 | benjk | for DIDs or for outbound calls |
20:42.02 | robl^ | I dumped broadvoice about 4 months ago... |
20:42.09 | m160858 | it's my first time using broadvoice on asterisk |
20:42.16 | m160858 | exactly |
20:43.09 | m160858 | hello to all !! |
20:43.13 | m160858 | i'm from peru |
20:43.55 | m160858 | i'm looking for a new provider |
20:44.12 | m160858 | for unlimited calls from USA & Canada |
20:44.16 | tzafrir_home | you're also flooding the channel |
20:44.53 | m160858 | oops |
20:45.38 | m160858 | ok, change the question |
20:46.08 | m160858 | what card digium recommends to me |
20:46.16 | benjk | who is flooding the channel? |
20:46.17 | m160858 | i've 3 E1 |
20:46.54 | benjk | you can use a quad T1/E1 card from Digium or a quad T1/E1 card from Sangoma |
20:47.15 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
20:47.33 | m160858 | but, they say to me that there are problems with cards of 4 ports |
20:47.57 | m160858 | that one uses 3 cards of 1 port each |
20:48.41 | *** join/#asterisk Dr-Linux (n=loyal@202.59.75.58) |
20:49.40 | benjk | who says there are problems with 4 port cards? |
20:49.42 | m160858 | problems with the bus, for example |
20:49.57 | benjk | no, you got that upside down I think |
20:49.58 | Dr-Linux | question, i have new cisco 7960/7940, so can i direct load 7.4 firmware, or thats important to 1st load older SIP firmware? |
20:50.05 | m160858 | a friend that works with asterisk |
20:50.12 | benjk | there are problems with more than two cards in one PC |
20:50.22 | m160858 | sure? |
20:50.34 | m160858 | what kind of computer, i need? |
20:50.40 | benjk | so if you need 4 E1s, then you should have a single 4 port card |
20:50.46 | benjk | depends |
20:50.50 | benjk | what codecs |
20:51.07 | m160858 | i have 3 E1's |
20:51.21 | m160858 | i don't know .. g729 |
20:52.49 | benjk | g729 is heavy on CPU |
20:53.02 | m160858 | i know, but |
20:53.04 | benjk | you cant do that with one server |
20:53.16 | m160858 | here in peru |
20:53.16 | benjk | you need about 4 servers |
20:53.36 | m160858 | the internet doesn't so good |
20:53.40 | benjk | one server can only handle about 80-100 channels with G729 |
20:53.43 | m160858 | sorry by my english |
20:53.50 | benjk | no problem |
20:54.38 | benjk | and for that you already need a dual Xeon |
20:54.46 | m160858 | one server with whatever of processor |
20:55.41 | benjk | if you want to do 4 E1s with G729 on a single server, then you will have to use IBM OpenPower 720 4-way |
20:55.52 | benjk | costs about 20.000 USD |
20:56.14 | benjk | that may be able to handle it, just about |
20:56.15 | m160858 | in peru I do not believe that they sell it |
20:56.20 | mog_home | nah |
20:56.21 | m160858 | hahaha |
20:56.29 | benjk | you can order it in the US, mail order |
20:56.30 | mog_home | you can do 4 E1 with an echo can card |
20:56.34 | mog_home | with dual xeon box |
20:56.43 | mog_home | 3 ghz |
20:56.47 | m160858 | that is very expensive |
20:56.47 | benjk | mog, with G729 transcoding? |
20:56.51 | mog_home | yeah |
20:57.03 | benjk | is the G729 on the hardware? |
20:57.06 | mog_home | no |
20:57.10 | mog_home | software |
20:57.13 | mog_home | but you need echo can board |
20:57.13 | m160858 | and? |
20:57.16 | m160858 | a compatible computer |
20:57.19 | benjk | since when? |
20:57.21 | mog_home | to take off the cpu load |
20:57.26 | mog_home | since rev2 cards |
20:57.35 | benjk | how many concurrent channels? |
20:57.45 | mog_home | 124 channels can be done |
20:57.56 | mog_home | probably a few more |
20:58.02 | benjk | ok |
20:58.05 | mog_home | i just know we have driven 4 e1 spans |
20:58.06 | m160858 | mmmm |
20:58.16 | benjk | that's cool |
20:58.16 | m160858 | 30 i think |
20:58.24 | m160858 | no, impossible |
20:58.43 | brimstone | nothing's impossible |
20:58.50 | brimstone | impractical maybe, but not impossible |
20:58.52 | benjk | mog, you should be doing some testing with the OpenPower 710 and 720 |
20:58.55 | m160858 | haha |
20:59.01 | mog_home | send me one ben |
20:59.08 | benjk | I reckon they can handle 400 or 500 channels |
20:59.15 | Dr-Linux | anybody answer my que |
20:59.16 | m160858 | I am new in this subject |
20:59.18 | Dr-Linux | question, i have new cisco 7960/7940, so can i direct load 7.4 firmware, or thats important to 1st load older SIP firmware? |
20:59.23 | benjk | et least the 720 4 way |
20:59.35 | mog_home | got me ^_^ |
20:59.41 | mog_home | i know we tried those sgi boxes |
20:59.52 | benjk | the IBM uses POWER5 |
20:59.56 | mog_home | and were dissapointed on how many it could drive compared to a standard dell |
20:59.59 | benjk | sual core |
21:00.05 | brimstone | i wonder how many channels my work machine can drive |
21:00.11 | benjk | er dual core POWER5 |
21:00.42 | benjk | IBM has those in their BlueGene super computers |
21:00.57 | benjk | a whole bunch of them in the top ten of the top500 list |
21:01.08 | m160858 | excuse, what about on my question? |
21:01.45 | benjk | m160858: you should try DIgium's new T1/E1 card WITH ECHO CANCELLATION ON BOARD |
21:02.11 | benjk | apparently, that can now handle about 120 channels with G729 |
21:02.14 | m160858 | ok, but it's funcionally on compatible computer? |
21:02.30 | benjk | andy PCI based system |
21:02.47 | benjk | er *any* PCI based system |
21:03.04 | benjk | Dual Xeon 3GHz |
21:03.10 | benjk | for example |
21:03.17 | m160858 | ok, thanks |
21:03.38 | m160858 | I am going to eat something, already I return |
21:04.08 | Dr-Linux | benjk |
21:05.20 | benjk | yes |
21:06.11 | Dr-Linux | benjk: i asked a question |
21:06.14 | Dr-Linux | question, i have new cisco 7960/7940, so can i direct load 7.4 firmware, or thats important to 1st load older SIP firmware? |
21:06.20 | benjk | I don;t know |
21:06.32 | benjk | I would have answered already if I knew |
21:07.05 | benjk | sorry, can't help you with that |
21:07.11 | Dr-Linux | no probelm sir |
21:07.19 | Dr-Linux | benjk: can i go with my other question |
21:07.30 | benjk | :-) |
21:08.19 | Dr-Linux | i have 2 fxo cards (4 port each) that i already disucussed with you, but i have some doubts |
21:08.52 | Dr-Linux | so for that >> in zaptel.conf will be fxsks=1-8 |
21:08.57 | Dr-Linux | is that right? |
21:11.03 | benjk | yep |
21:11.25 | morale | Set(CALLERID(all)="R K McConnachie <4036681593>") - anyone know why that would not work? |
21:12.07 | benjk | is that new syntax from 1.2? |
21:12.12 | morale | i think so |
21:12.24 | benjk | sorry, dont know about that |
21:12.31 | benjk | I use |
21:12.41 | IronHelix | SetCallerID("Name" <number>) |
21:12.46 | IronHelix | but thats 1.0 |
21:12.53 | benjk | SetCallerIDName(Fred Flintstone) |
21:13.11 | benjk | and SetCallerIDNum(555-555-5555) |
21:13.18 | benjk | or that, yes |
21:19.16 | Dr-Linux | benjk: and in my case in zapata.conf will be, [channel] >> signalling=fxs_ks, group=1 channel> 1-8 |
21:19.20 | Dr-Linux | is that right? |
21:20.14 | benjk | yes, but you may want to put a context there too |
21:20.30 | benjk | like context=incoming or so |
21:20.43 | Dr-Linux | oo |
21:21.10 | Dr-Linux | benjk: yeah what i want to know, thats my real question |
21:21.21 | Dr-Linux | if i put there context=ivr |
21:21.24 | Rawplayer | how can i add a testnumber in my config? |
21:21.31 | Rawplayer | to test if my softphone dails |
21:21.48 | IronHelix | ~softphone |
21:21.49 | jbot | something that should be drug out into the street and shot |
21:21.55 | Dr-Linux | bcoz my context is in extension.conf is >> [ivr] where i want the caller to redirect |
21:22.03 | benjk | sure, ivr is fine, as long as you have a context [ivr] in extensions.conf that will handle the incoming calls |
21:22.06 | IronHelix | and/or do exten => 1234,1,echotest() |
21:22.13 | IronHelix | brb |
21:23.13 | Rawplayer | IronHelix: that was to me?:) |
21:23.21 | IronHelix | yeah, both of them |
21:23.57 | Rawplayer | i see "call not approved" |
21:24.03 | Rawplayer | on my softphone |
21:24.08 | benjk | ok, I really have to get some sleep now. cya |
21:24.34 | IronHelix | eh |
21:24.42 | IronHelix | exten => 1234,1,echo() |
21:24.44 | IronHelix | not echotest |
21:24.45 | IronHelix | my bad |
21:25.15 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
21:25.40 | Rawplayer | still the same |
21:25.45 | IronHelix | did you reload |
21:25.50 | Rawplayer | restart even |
21:26.00 | IronHelix | is the softphone in the right context? |
21:26.39 | IronHelix | (in sip.conf for the softphone config block there is context=, is that the same as the [context] in extensions.conf that the echo line is in? |
21:27.45 | Rawplayer | exten => 1234,1,echo(SIP/kevin1) |
21:27.51 | IronHelix | no |
21:27.53 | *** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk) |
21:28.00 | IronHelix | exten => 1234,1,Echo |
21:28.01 | IronHelix | just that |
21:28.21 | p1tst0p | hi, i am trying to build CVS on ubuntu.. and it keeps failing here, "/usr/bin/ld: cannot find -lssl" |
21:28.32 | p1tst0p | any thoughts, on what i could be missing. |
21:28.39 | Rawplayer | libssl |
21:28.42 | p1tst0p | ah |
21:28.43 | IronHelix | p1- install packages openssl and openssl-devel |
21:28.53 | p1tst0p | cheers peeps ! |
21:29.11 | Rawplayer | [kevin1] |
21:29.11 | Rawplayer | type=friend |
21:29.11 | Rawplayer | context=kevin1 |
21:29.11 | Rawplayer | ;callerid=kevin1 |
21:29.11 | Rawplayer | callerid="2045093kok" <101> |
21:29.13 | Rawplayer | host=dynamic |
21:29.16 | Rawplayer | secret=test |
21:29.18 | Rawplayer | username=kevin1 |
21:29.21 | Rawplayer | nat=never |
21:29.23 | Rawplayer | thats from sip.conf |
21:29.32 | chapeaurouge | ~pastebin |
21:29.34 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
21:29.40 | chapeaurouge | ~p2bin |
21:29.41 | jbot | i heard p2bin is a script to paste to the http://pastebin.ca from the standard input (linux/unix CLI) . Can be fetched from http://www.madpenguin.org/blogs/chapeaurouge/?p=92 |
21:30.33 | IronHelix | raw- in put your echo line in the kevin1 context in extensions.conf then |
21:32.08 | *** join/#asterisk marc32422 (n=marc3234@206-248-134-171.dsl.teksavvy.com) |
21:32.37 | pifiu | where is jer jer? |
21:32.59 | *** join/#asterisk ComputerWarm (n=workingg@66.244.235.210) |
21:33.31 | ComputerWarm | Hello; Question is there anyway to do it so if someone calls in with a blocked number it auto sends them to voicemail |
21:34.50 | *** join/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de) |
21:36.24 | ComputerWarm | anyone? |
21:38.32 | file[laptop] | ComputerWarm: of course there's a way, you have the callerid accessible in the dialplan, you can compare strings, voila - there you go |
21:39.48 | iDunno | or use exten => dialed/callerid,blah,stuff |
21:40.22 | ComputerWarm | in the dial plan for example will it handle if-else statements? |
21:40.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
21:40.27 | *** mode/#asterisk [+o denon] by ChanServ |
21:41.13 | file[laptop] | try iDunno's way first, then type show application gotoif and read |
21:41.22 | file[laptop] | type it on the CLI that is... |
21:41.52 | ComputerWarm | ok thanks. the reason i was asking there is a number that calls me and i was wondering if in the dial plan, if the caller id matched that number it would hang up on them |
21:44.04 | IronHelix | you want hte anti ex girlfriend filter :) |
21:44.45 | IronHelix | try this http://www.voip-info.org/wiki-Asterisk+cmd+LookupBlacklist |
21:44.58 | IronHelix | uses the asterisk database to store a list of numbers that are blacklisted |
21:45.12 | IronHelix | you can also blacklist the current call wtih a star code of some kind |
21:45.28 | ComputerWarm | IronHelix oh cool thank you |
21:46.10 | IronHelix | and if they are in the blacklist, lookupBlacklist sends them to priority current+101 |
21:46.25 | IronHelix | and i think *80 is add current call to blacklist |
21:46.31 | file[laptop] | slowly am I going crazy |
21:46.40 | IronHelix | http://www.voip-info.org/wiki-Asterisk+vertical+service+activation+codes |
21:47.02 | IronHelix | file- depens what on you are about talking? |
21:47.25 | IronHelix | might be *60 |
21:47.31 | ComputerWarm | IronHelix wow didn`t know there was so much added to asterisk. |
21:47.41 | IronHelix | oh theres TONS |
21:47.45 | IronHelix | poke around in the voip-info wiki |
21:47.54 | IronHelix | asterisk has become very fully featured |
21:48.15 | ComputerWarm | ya i remember when i first started using it. you where lucky to get the queue`s working lol |
21:48.48 | IronHelix | hehe |
21:49.59 | robl^ | when I first started using Asterisk, I was lucky to get SIP working with my VoIP provider. everytime chan_sip was changed, it broke something for me.. but that was long before 1.0 was released. LOL :) |
21:50.12 | file[laptop] | those service codes are only applicable to zap analog lines you know... just a little fyi |
21:52.21 | IronHelix | mmm good point |
21:52.24 | ComputerWarm | file[laptop] ya i don`t think i have those services on my phone line |
21:52.33 | IronHelix | no they are asterisk services |
21:52.37 | IronHelix | if you have zap channels |
21:52.43 | file[laptop] | they're written into chan_zap |
21:52.43 | IronHelix | like if you have a zaptel fxs port youc an use them |
21:53.00 | file[laptop] | so if you hook a phone up to an FXS port via a channel bank, or the TDM400P card, then they'll work for it... |
21:57.39 | *** join/#asterisk lorinc (n=ang@caracas-2459.adsl.interware.hu) |
22:01.37 | Dr-Linux | if i put there context=ivr in zapata.conf , so doing that caller will be able to redirect in extionsion.conf [ivr] context ? |
22:01.40 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
22:04.15 | Dr-Linux | whats best webbased monitoring problem for * ? |
22:04.39 | IronHelix | FOP (flash operator panel)? |
22:05.13 | IronHelix | drlinux- if you put context=ivr, when a call comes in it will go to extension s (or whatever the user dialled) in the ivr context and any included channels |
22:06.28 | Dr-Linux | ok thanks |
22:06.32 | tzafrir_home | IronHelix, dependeing on the number dialed. s is if no number was dialed |
22:06.38 | IronHelix | exactly |
22:06.41 | *** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net) |
22:07.56 | Dr-Linux | IronHelix: my other included context is default, and my all users are in default context, so can you give me example pateren, if the user wanna make a call outside via PSTN ? |
22:08.04 | Dr-Linux | i have 2 fxo cards (4 port) each |
22:08.31 | IronHelix | well put all your fxo ports in a group (call it g0) |
22:09.12 | Dr-Linux | oo |
22:09.13 | IronHelix | do exten => _XXXXXXXXXX,1,Dial(Zap/g0/${EXTEN}) |
22:09.24 | IronHelix | put as many Xs as you expect to have digits |
22:09.40 | Dr-Linux | IronHelix: sir but in zapata.conf i gonna put >> group=1 and channel > 8 |
22:09.41 | IronHelix | as i recally you arent in the US so you have different dialing patterns than we do |
22:09.46 | Dr-Linux | is that wrong? |
22:09.55 | IronHelix | well channel 8 would be IN group on |
22:09.58 | IronHelix | one |
22:10.10 | IronHelix | so if you dialled group 1 then channel 8 might be picked |
22:10.37 | Dr-Linux | ok sir but where i put g0 ? |
22:10.45 | Dr-Linux | really i dont understand that? |
22:11.07 | IronHelix | http://www.voip-info.org/wiki-Channels+and+Groups scroll down to zap groups |
22:11.16 | IronHelix | if you put group=1 before you define all your fxo channels |
22:11.23 | IronHelix | they will all be in group1 |
22:11.30 | Dr-Linux | okey lemme try |
22:11.48 | IronHelix | note- you DONT wanna put fxs channels (if you have any) in that group |
22:12.13 | IronHelix | so before any fxs ports put just group= to define no group |
22:12.26 | Dr-Linux | yes but sir what you suggest/recommend what should be in group=? |
22:12.36 | IronHelix | a number, say 1 |
22:12.44 | IronHelix | for the fxo ports |
22:12.57 | IronHelix | do you have any fxs? |
22:13.07 | Dr-Linux | i have only fxo cards, i don't wanna use fxs cards |
22:13.11 | Dr-Linux | no sir |
22:13.14 | IronHelix | ok then you're good |
22:13.29 | IronHelix | just before you define the fxo ports with channel= do group=1 |
22:13.30 | Dr-Linux | i don't wanna use them, i have only 2 fxo cards (4 port each) |
22:13.47 | IronHelix | so you have like channel=1-8 |
22:13.58 | Rawplayer | can anyone tell me whats wrong with this? http://www.nomorepasting.com/paste.php?pasteID=53370 i've got the sample from a book |
22:14.21 | Dr-Linux | IronHelix: so sir in thise case what will be the pateren? |
22:14.41 | IronHelix | huh? |
22:14.44 | Dr-Linux | like you intstruct bfore |
22:14.45 | Dr-Linux | <IronHelix> do exten => _XXXXXXXXXX,1,Dial(Zap/g0/${EXTEN}) |
22:14.48 | IronHelix | ahh |
22:14.58 | IronHelix | then Dial(Zap/g1/${EXTEN}) |
22:15.12 | Dr-Linux | only think i didn't understand is g0 |
22:15.12 | rking | Rawplayer: does asterisk -rcvvvv show anything interesting when you do stuff? |
22:15.26 | IronHelix | g0 is group 0 |
22:15.41 | IronHelix | dial zap/g0/exten means dial the called number in any zap channel from group0 |
22:15.46 | IronHelix | it will choose a free one and dial |
22:16.27 | Dr-Linux | oo ic and there will totally 8 channels like g1 , g2 g3 |
22:16.28 | Dr-Linux | right? |
22:16.34 | IronHelix | no |
22:16.40 | IronHelix | g is group |
22:16.45 | IronHelix | if you want to dial a particular channel |
22:16.55 | IronHelix | say you wanna dial port 5 (first port on the 2nd card) |
22:16.56 | Rawplayer | nope nothing |
22:17.03 | IronHelix | you'd dial zap/5/whatever |
22:17.32 | IronHelix | but if you dial g1 that means use the first available zap channel thats associated to group 1 |
22:17.57 | IronHelix | drlinux- pastebin your zapata.conf file |
22:18.06 | IronHelix | i'll show you what you need to do |
22:18.22 | rking | Rawplayer: hrm... nothing interesting or nothing at all? |
22:18.54 | Dr-Linux | IronHelix: can i tell me something in pvt, if u don't mind |
22:19.27 | IronHelix | sure |
22:22.57 | *** join/#asterisk clive- (n=pirch@ndn-165-129-135.telkomadsl.co.za) |
22:23.59 | kernoman | im trying to connect out via pstn (x100p) card however I keep getting the all circuits are busy now message - why is this? |
22:24.38 | file[laptop] | kernoman: you know, that generic is very message... |
22:24.48 | file[laptop] | er that message is very generic... you should post what you see on the CLI on pastebin.com |
22:26.49 | kernoman | sorry, ok ive posted it to pastebin.com |
22:27.29 | Rawplayer | no nothing rking |
22:27.57 | rking | Rawplayer: from that same "asterisk -crvvv" shell, does a "reload" show anything interesting? |
22:27.59 | file[laptop] | kernoman: giving us the link would help so we could see it |
22:29.11 | kernoman | doh! sorry im totally new to IRC and pastebin... |
22:29.23 | *** part/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de) |
22:29.30 | kernoman | think i have it working now though, just going to test to confirm |
22:29.32 | Rawplayer | a lot of stuff with found |
22:29.43 | Rawplayer | everything looks fine |
22:29.52 | Rawplayer | <PROTECTED> |
22:29.52 | Rawplayer | <PROTECTED> |
22:29.57 | Rawplayer | but still not working |
22:30.28 | Rawplayer | setting up sip.conf and extensions.conf is enough right? |
22:30.36 | rking | Rawplayer: i think so |
22:30.44 | *** join/#asterisk p1tst0p (n=admin@82-38-106-54.cable.ubr03.donc.blueyonder.co.uk) |
22:31.15 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
22:31.40 | p1tst0p | hi, i am trying to ring my friends asterisk box, from my own asterisk box ( both with sipgate accounts) but he keeps seeing, ast_set_read_format: Unable to find a path from g723 to g726 |
22:31.47 | p1tst0p | and, ast_set_write_format: Unable to find a path from ulaw to g723 |
22:31.48 | *** join/#asterisk clinthome (n=clinthom@va-chrvlle-cad1-bdgrp5-8d-209.chvlva.adelphia.net) |
22:31.58 | p1tst0p | no matter what codecs we try, it always says this on his side. |
22:32.33 | IronHelix | somewhere, some thing is needing g723 codec |
22:32.36 | IronHelix | blah |
22:32.42 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
22:32.49 | rking | Rawplayer: something is very odd that you don't see anything in the -vvvv console when you try to sip in - it should at least say that it's denying the call for some reason |
22:33.38 | shimi | is there a good location where I can get PSTN termination for asterisk (IAX2) that is cheap (but with good sound quality) to destinations in USA / Europe / Japan ? |
22:33.56 | p1tst0p | IronHelix, if ring his sipgate account from my mobile all is good.. i can get through to him ok.. its just if i ring him from my sipgate, to his sipgate.. |
22:33.57 | *** join/#asterisk fugitivo (n=ajf@209.13.241.110) |
22:34.33 | p1tst0p | IronHelix, and we both have allow=g723 in our outgoing sipgate configs. |
22:34.50 | IronHelix | dont allow it |
22:34.54 | rowter | I been getting lots of this messages Nov 26 17:33:47 DEBUG[2833]: chan_sip.c:7251 handle_request: That's odd... Got a response on a call we dont know about. |
22:35.06 | IronHelix | do disallow=all then allow=ulaw allow=alaw allow=gsm allow=ilbc |
22:35.07 | *** join/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de) |
22:35.19 | IronHelix | if you cant get it going with ulaw alaw ilbc or gsm you have bigger rpbolems |
22:37.12 | *** join/#asterisk kimc (n=freenode@pcp04041197pcs.wbrmfd01.mi.comcast.net) |
22:37.41 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
22:37.50 | clive- | can anyone help me get a new network card going...? |
22:38.14 | p1tst0p | IronHelix, when i do that we get, process_sdp: No compatible codecs! |
22:38.41 | *** join/#asterisk ThePeopleGA (n=kemtram@rev-204.120.18.37.genesiswireless.us) |
22:38.50 | IronHelix | ugh |
22:38.52 | IronHelix | that is very odd |
22:39.28 | fugitivo | p1tst0p: type "show codecs" at the cli |
22:39.32 | robl^ | check the SIP phones being used one each astersik box |
22:39.49 | fugitivo | p1tst0p: what do you get? |
22:39.50 | robl^ | I bet one of the phones is stuck on g723 |
22:40.03 | p1tst0p | fugitivo, about 15 codecs |
22:40.27 | *** join/#asterisk ThePeopleGA (n=kemtram@rev-204.120.18.37.genesiswireless.us) |
22:40.30 | p1tst0p | i see, g723, gsm, ulaw, alaw, g726, g729 |
22:41.02 | p1tst0p | thing is, on the end that im seeing the Unable to find a path from g723 to g726 |
22:41.05 | morale | anyone know why i cannot call gov't of canada numbers with freeworldtel? |
22:41.06 | IronHelix | switch the order around |
22:41.20 | IronHelix | put ulaw, alaw, gsm, g729 in that order |
22:41.22 | p1tst0p | he can successfully ring 10000, which is the sipgate test phone |
22:48.51 | *** join/#asterisk [TK]D-Fender (i=1000@66.11.164.239) |
22:48.57 | [TK]D-Fender | whee! |
22:48.57 | jahani | asterisk work fine on what distribution ? |
22:49.07 | jahani | fedora or centos? |
22:49.10 | [TK]D-Fender | jahani : Just about everything :) |
22:49.30 | [TK]D-Fender | You name it and Asterisk has been there.... |
22:49.44 | [TK]D-Fender | I prefer Slackware, others RH based, others Debian based... |
22:49.52 | [TK]D-Fender | Just use what you're comfortable with |
22:49.58 | [TK]D-Fender | and the rest is just Linux :) |
22:50.40 | Rawplayer | rking: what should be in the config to let sip work |
22:50.53 | Rawplayer | configureing sip.conf and extensions.conf should be enough right? |
22:50.54 | jahani | ok |
22:50.56 | jahani | thank you |
22:51.01 | Rawplayer | i now got the sample from oreilly book |
22:51.06 | Rawplayer | and its still not working |
22:51.21 | *** join/#asterisk Qwell[] (n=chatzill@pool-71-108-28-219.lsanca.dsl-w.verizon.net) |
22:51.29 | rking | Rawplayer: as far as i can think, yep - i'm in the process of config'ing sip with you (i've had IAX working), so i'll let you know |
22:51.43 | rking | Rawplayer: have you been able to dial any other SIP servers with your client? |
22:52.05 | Rawplayer | nope |
22:52.07 | Rawplayer | just installed it |
22:54.16 | Rawplayer | authentications works |
22:55.35 | rking | Rawplayer: authentication to where? |
22:55.42 | Rawplayer | to asterisk |
22:55.44 | Rawplayer | with my client |
22:55.47 | *** join/#asterisk netwetrust (n=sminguel@61.Red-80-24-25.staticIP.rima-tde.net) |
22:55.50 | dudes | for iax trunking does the host have to be static or can it be dynamic |
22:56.02 | rking | Rawplayer: ahh, awesome - you're more than half-way there... maybe now the problem is just your dialplan? |
22:56.35 | Rawplayer | [internal] |
22:56.35 | Rawplayer | exten => 100,1,Dial(SIP/john) |
22:56.36 | Rawplayer | exten => 611,1,Echo() |
22:56.42 | Rawplayer | this is in my extensions.conf now |
22:57.10 | Rawplayer | in sip.conf i have in john's part context=internal |
22:57.22 | Rawplayer | so that it refers to internal in extensions.conf |
22:57.47 | bbz | Would anyone here be interested in some Polycom 501 phones? We got stuck w/ an extra 20 after a project, and we need to unload them at cost---->>> <<<---- [sorry alot of people have been missing the cost of these] if anyone is interested send me a private message (if i dont respond immediately, leave me your email address, and i can send you info) i have a reference in the channel if you need. -- |
22:58.04 | rking | Rawplayer: start of with something simpler to tell if it passed or not - just like, exten => s,1,Background(tt-monkeysintro.gsm) |
22:58.08 | rking | urr, no .gsm |
22:58.19 | Rawplayer | k lets try |
22:58.20 | many | dejadejadejavu |
22:58.36 | Rawplayer | but whats the number then?:) |
22:58.47 | [Airwolf] | Rawplayer, can you also do "sip debug" on you console and then paste the output here ? |
22:59.00 | Rawplayer | it doesnt show anything :\ |
22:59.04 | asteriskmonkey | no paste it in pastebin youll flood the channel |
22:59.16 | asteriskmonkey | do sip intense debug |
22:59.33 | [Airwolf] | Rawplayer, how did you start asterisk or got back at the console ? |
22:59.47 | Rawplayer | hmm it show something now |
23:00.54 | [Airwolf] | And do you here something ? |
23:00.59 | [Airwolf] | hear |
23:02.08 | JunK-Y | bbz: we actually looking for 20 501s. |
23:02.41 | *** join/#asterisk gvag11 (n=g@ppp23-adsl-86.ath.forthnet.gr) |
23:03.19 | gvag11 | hi all |
23:03.22 | jahani | gnophone is for what ? |
23:03.33 | gvag11 | freevoip rocks !!!!!!!1 |
23:03.46 | gvag11 | i just test it and its really fine ... |
23:04.04 | robl^ | jahani: not much. it is a VERY OLD IAX sofphone. Its not mained anymore |
23:04.18 | jahani | ok |
23:04.18 | robl^ | not maintained, even |
23:04.20 | jahani | thank you |
23:04.47 | JunK-Y | robl^: i prefer polycom :) |
23:05.03 | jahani | and gtkiaxyprov ? |
23:05.49 | *** join/#asterisk _Thor (i=_Thor@user-vc8fl7n.biz.mindspring.com) |
23:06.04 | robl^ | Junk-Y: I am considering Snom or Polycom... I am leaning towards Snom. After 1 year with Cisco and some of their "caveats", I am ready to switch on my home system |
23:06.57 | many | snom is cool. not perfect, but cool |
23:07.11 | _Thor | question: is there such a thing as a logoff command on the manager interface? |
23:07.19 | many | they are just a bit bitchy when they figure that your supportrequest is a bug of asterisk :) |
23:07.20 | robl^ | I don't think any phone is perfect. :) |
23:08.52 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
23:09.08 | robl^ | many: at least snom supports Asterisk, firmware is Linux based, and its easy to get firmware and order equipment. Ordering from Cisco can be a pain.. You have to order the phone, firmware license, support contract, and power supply seperate. |
23:09.09 | JunK-Y | polycom is aint bad. |
23:09.21 | JunK-Y | never tried a snom phone yet. |
23:09.33 | *** part/#asterisk volkerli (n=volkerli@port-212-202-0-231.dynamic.qsc.de) |
23:09.45 | [TK]D-Fender | JunK-Y : I've just reinstalled my server and am approaching normalcy :) |
23:09.47 | robl^ | JunK-Y. I have been playing with their softphone / emulated Snom 360, and I am quite impressed |
23:10.38 | morale | why does this not work in my extensions.conf - exten => _101,1,VoicemailMain (101) |
23:10.41 | many | robl: we work with a complete asterisk/snom installation at work |
23:10.42 | [TK]D-Fender | ok, got to reboot, back later... |
23:10.49 | morale | i want to dial 101 and it transfer me to my voicemail |
23:10.58 | *** part/#asterisk _Thor (i=_Thor@user-vc8fl7n.biz.mindspring.com) |
23:11.11 | robl^ | many: do you use 320s and 360s? any major complaints or "gotchas"? |
23:12.13 | many | 360s, no major complaints. tiny things which may seem annoying, but does not affect working with them. from the 13 phones ordered, i sent two back since they were... fuzzy? when doing network stuff (id guess defective nic) |
23:13.50 | robl^ | hrmmm.. I had network trouble with one of 7960s and had to have it replaced.. so that's not really a Snom only problem :) |
23:13.51 | many | for example the display shows a little snom.com logo, which i wanted replaced or removed which is not possible |
23:13.58 | JunK-Y | tk: after 3 days? :P |
23:19.59 | *** join/#asterisk stillbourne (n=stillbou@c-24-9-8-59.hsd1.co.comcast.net) |
23:22.24 | robl^ | many: ahh.. ok.. but sound quality and reliability is pretty good? |
23:22.51 | many | IMVHO yes. |
23:22.59 | many | atleast when you upgraded apps to 4.3 |
23:23.19 | morale | this asterisk voicemail wants me to register my password evertime i call it |
23:23.21 | many | when you get'em they usually have 3.6 which seems to have memleaks, but 4.3 seems pretty fine |
23:25.09 | *** join/#asterisk Little-L (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk) |
23:25.52 | robl^ | many: yeah. firmware upgrade is no big deal for me. esp since I am only talking about maybe 5-6 phones. This will be my home system. I have Ciscos here now.. and I have ciscos on a couple other servers I put into production.. only looking to switch here now.. |
23:26.09 | *** join/#asterisk Dr-Linux (n=loyal@202.59.75.58) |
23:26.42 | Dr-Linux | IronHelix: ufff i got dc and i'm was trying to connect since that time , but had no luck |
23:26.54 | IronHelix | gah |
23:26.59 | Dr-Linux | http://pastebin.com/439135 |
23:27.13 | IronHelix | ok gimme a few |
23:27.24 | Dr-Linux | IronHelix: i don't know how to copy the whole extensions.conf |
23:27.43 | Dr-Linux | thats why i miss somethings there |
23:27.49 | many | robl: thats easy anyway. snoms have a http interface, tell them to load a new firmware, reboot them and then you can even config the rest via dhcp // http-download of configfiles. |
23:27.53 | many | pretty nice, i think |
23:28.20 | many | the http interface also allows dialing new numbers and also numbers from lists which is nifty. |
23:29.06 | many | its just not so nice with acls, you cant enable it at work without having the cow-orkers play tricks on their cow-orkers. 8) |
23:29.14 | robl^ | many: is there a way to config a large number via config files.. like I could with Cisco? I found no reference to that in the manuals |
23:29.26 | *** join/#asterisk delphiuk (n=Richard@host86-128-157-3.range86-128.btcentralplus.com) |
23:29.32 | many | You do. you only have to configure this once for every phone |
23:29.48 | *** part/#asterisk delphiuk (n=Richard@host86-128-157-3.range86-128.btcentralplus.com) |
23:30.17 | robl^ | many: ahh. ok. |
23:30.18 | many | Since it downloads from http, rather than tftp like cisco, you can even write some php-or-whatever script to do complex operations |
23:31.40 | robl^ | many: that is what I was thinking... if they work well, I really will consider them for other production systems.. |
23:31.44 | *** join/#asterisk heath__ (n=root@12-215-32-62.client.mchsi.com) |
23:31.44 | many | sadly enough, some features seem to work only with latest asterisk + latest bristuff patch |
23:32.49 | robl^ | many: I am rebuilding my server as I type.. anything special beyond the lastest 1.2 cvs and bristuff I should add for snom? |
23:32.58 | many | i think not. |
23:33.13 | many | there are quite nice hints pages on voip-info.org |
23:33.46 | robl^ | I am finally getting off my bum and upgrading everything on this box from 1.0.7 to 1.2 |
23:33.50 | rking | http://pastebin.com/439169 # i'm trying to dial * using the X-Lite softphone, and this is the result. =\ |
23:34.08 | many | i geuss my server wont see 1.2 anymore *hehe* |
23:34.32 | robl^ | won't see 1.2? why not? gone to head? |
23:35.59 | many | no, i guess i wander off to openpbx |
23:36.02 | Qwell[] | rking: Why are you setting context=default, but using [sipincontext] in extensions.conf? |
23:36.22 | robl^ | isn't openpbx just a fork of Asterisk? |
23:36.45 | many | isnt netbsd just a fork of freebsd? :) |
23:36.56 | Qwell[] | Isn't Linux just a fork of sco unix? |
23:37.02 | rking | Qwell: good question - i had it the other way before - i've tried dozens of iterations of the configs - let me re-set it to that |
23:37.23 | many | qwell :) |
23:37.23 | robl^ | Qwell: NO NO NO!! Linux is a stolen from SCO! Not a fork! |
23:38.08 | robl^ | many: I just mean I don't understand how OPenPBX and Asterisk differ besides name.. |
23:38.10 | rking | Qwell: the result is the same |
23:38.20 | Qwell[] | and why two [sipin]'s in sip.conf? |
23:38.45 | many | robl^: They will differ when opo first releases. |
23:39.20 | many | to give you an idea, since the fork opo had ~1100 commits to svn |
23:39.32 | robl^ | many: what is the different "goal" for opo? |
23:39.35 | rking | Qwell: i wasn't sure where the username piece was supposed to come from, so i made one user "st" and then later added the "2345" without thinking to change that title |
23:40.35 | rking | same result with the \names set to different values |
23:40.36 | Qwell[] | rking: http://www.asteriskguru.com/tutorials/xlite_softphone.html |
23:40.40 | many | robl: www.openpbx.org/ wiki.openpbx.org lists some, and to quote the japanese lifestyle: competition is good for your own quality. anyway, i guess this should be taken off here. :-) |
23:40.59 | ritesha | MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info....couldn't resolve yet. any suggestions on how to increase debug dump for mysql |
23:41.44 | ritesha | anyone experienced similar problem would care of comment. I really need to fix this asap and any help would be appreciated |
23:41.47 | ritesha | Thanks a lot. |
23:42.08 | Qwell[] | ritesha: pastebin the exact errors you're getting from asterisk |
23:42.12 | Qwell[] | and your configs |
23:42.15 | ritesha | sure. |
23:44.03 | ritesha | http://pastebin.ca/31358 |
23:44.13 | ritesha | which other config do you need? |
23:44.23 | Qwell[] | lemme see |
23:45.06 | Qwell[] | Is mysql listening on the network, or just the socket? |
23:45.16 | ritesha | just the socket |
23:45.28 | Qwell[] | Then comment out the dbhost line |
23:45.31 | Qwell[] | let it use dbsock |
23:45.39 | Qwell[] | probably dbport too |
23:46.00 | Qwell[] | if it's listening on only the socket, it won't be able to connect to localhost |
23:46.59 | Qwell[] | odd, it says if the host is "localhost", it'll connect through the socket...that's not cool at all |
23:47.21 | m160858 | hi |
23:47.49 | m160858 | here i'm agaim |
23:47.56 | ritesha | hmm..this helped a little. I am not getting the connect error. but... http://pastebin.ca/31359 |
23:48.03 | m160858 | i'm looking for a new international calls provider, for a unlimited calls to USA & Canada |
23:48.09 | m160858 | do you know someone? |
23:48.17 | ritesha | so I am wondering if it actually got to mysql? |
23:48.29 | Qwell[] | ritesha: it did |
23:48.34 | ritesha | Thaks a lot Qwell for helping me on this. really need it going... |
23:48.36 | Qwell[] | I'm not seeing any errors there |
23:48.54 | ritesha | Oh!! Meaning I might have wrong dbase values for it not to get to the queue |
23:49.05 | Qwell[] | or the table could be missing, or the table is empty, or... |
23:49.14 | Qwell[] | ritesha: paypal is my friend, if you're *really* thankful... ;) |
23:49.37 | ritesha | sure...IM me and I will be glad to |
23:49.52 | ritesha | seriously...no jokes |
23:54.25 | m160858 | question: the panasonic telephones are compatibility with asterisk? |
23:55.14 | mog_home | probably not |
23:55.27 | mog_home | are they "digital" |
23:55.38 | Qwell[] | panasonic? probably analog |
23:55.49 | Qwell[] | or perhaps IP |
23:56.01 | m160858 | i'm confused |
23:56.14 | Qwell[] | m160858: Do you have a web address for the phone? |
23:56.44 | m160858 | they are analog |
23:57.08 | m160858 | they working actually with a panasonic pbx |
23:57.31 | dudes | has anyone tried t38 on openpbx |
23:57.43 | mog_home | then no |
23:57.47 | dudes | I imagine it's suppose isn't any diff than /w * |
23:58.02 | mog_home | but it is probably similar |
23:58.09 | m160858 | but i want to replace that panasonic PBX by one asterisk |
23:58.51 | m160858 | that me force to change all the telephones? |
23:59.05 | mog_home | connect the pasanoic to asterisk |
23:59.10 | mog_home | over t1 |
23:59.30 | Qwell[] | panasonic>tincan>string>tincan>asterisk |
23:59.58 | m160858 | in my country doesn't exists T1 |