00:00.21 | kn0x | yes... |
00:00.26 | *** join/#asterisk PepOSX (n=angeldav@190.72.147.233) |
00:01.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
00:01.43 | *** join/#asterisk zippytech (n=chatzill@dns2.zippytech.com) |
00:02.28 | zippytech | any one seen this TRUNK Dial failed due to CHANUNAVAIL |
00:03.42 | JT | LemensTS: no, it converts SIP to FXS |
00:04.05 | JT | why would you need another channel bank? |
00:05.12 | zippytech | failing through to other trunks |
00:05.39 | *** join/#asterisk ninazu (n=who@cpe-67-10-211-160.satx.res.rr.com) |
00:06.26 | jameswf-home | ~freeswitch |
00:06.26 | jbot | freeswitch is probably an open source soft switch that is *not* a fork of asterisk http://www.freeswitch.org/interview2.htm |
00:06.32 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
00:09.36 | zippytech | <PROTECTED> |
00:09.37 | zippytech | <PROTECTED> |
00:09.39 | zippytech | <PROTECTED> |
00:10.04 | zippytech | why would it not show an channels |
00:10.12 | zippytech | there should be 4 |
00:10.33 | ManxPower | maybe your /etc/asterisk/zapata.conf is not correct? |
00:11.49 | jameswf-home | imagine that |
00:15.50 | robmac67 | zippytech: what card do you have installed ? |
00:16.25 | angryuser | gn all ;) |
00:17.39 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
00:22.21 | Greek-Boy | is there a channel state that is the opposite of CHANUNAVAIL? I want to use a GotoIF statement provided there is one available channel in a trunk. |
00:24.10 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@76-204-200-226.lightspeed.hstntx.sbcglobal.net) |
00:25.19 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
00:28.43 | *** join/#asterisk duri (n=mduregon@c-76-105-157-51.hsd1.or.comcast.net) |
00:29.11 | duri | what kind of load can a WRT54GS handle ? |
00:29.34 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0470144e074db085) |
00:38.21 | ManxPower | Greek-Boy: No. |
00:38.33 | ManxPower | But you can use the app ChanIsAvail |
00:39.02 | ManxPower | Greek-Boy: you sure do like to make things complicated |
00:40.42 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
00:48.05 | ManxPower | They're not killers, just misunderstood. |
00:48.11 | ManxPower | I blame it on the schools. |
00:48.28 | jameswf-home | I blame the parents |
00:48.31 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
00:49.21 | JerJer | i blame canada |
00:50.00 | JerJer | i can hear them up there at night sharpening their ice skates, just waiting for the perfect time to attack |
00:50.19 | ManxPower | I wish they would hurry up. We could use some decent beer. |
00:51.54 | Greek-Boy | ManxPower once an extensions reaches hangup() does it ignore priorities after that? |
00:53.01 | Greek-Boy | i need to do something after a call is hung up |
00:53.25 | lunaphyte | what is smdi? |
00:54.07 | JerJer | i think smdi is a serial connection to legacy PBXs |
00:54.15 | lunaphyte | oh, ok. |
00:54.53 | JerJer | Station Message Detail Interface |
00:55.58 | lunaphyte | ah, thanks. |
00:56.27 | lunaphyte | oh, is it maybe Simplified Message Desk Interface? http://en.wikipedia.org/wiki/SMDI |
00:56.28 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
00:56.38 | lunaphyte | or is that something else with the same acronym? |
01:02.00 | cleone | any one here have iaxcomm? |
01:02.58 | Greek-Boy | when an extension reaches hangup() does it ignore priorities after that? |
01:03.19 | adeel | why does the polycom digit mapping feature suck so hard? |
01:06.17 | *** join/#asterisk RoyK (n=roy@ti211110a081-7661.bb.online.no) |
01:16.39 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
01:18.15 | *** part/#asterisk beek (n=klinebl@65.211.106.243) |
01:19.25 | Greek-Boy | someone please take a look at http://pastebin.com/d2f881e |
01:19.44 | Greek-Boy | i am trying to setup a macro in my dial plan to handle emergency calls |
01:19.57 | Greek-Boy | but I seem to be going off track big time |
01:24.34 | Greek-Boy | forget that url. check http://pastebin.com/d45525822 |
01:26.25 | rajiv | whats the equivalent of 'core show locks' in 1.2.x ? |
01:30.00 | ManxPower | rajiv: there isn't. |
01:30.34 | ManxPower | that feature (invented by rusellb, I think) is new in 1.6.x, it may have been backported to 1.4. |
01:31.12 | rajiv | any ideas what to do about: WARNING[14800]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x81aae58', 9 retri |
01:33.49 | Greek-Boy | ManxPower can u check http://pastebin.com/d45525822 for me? |
01:33.50 | Greek-Boy | pls |
01:33.58 | ManxPower | no |
01:34.02 | Greek-Boy | lol |
01:34.03 | Greek-Boy | k |
01:34.04 | Greek-Boy | :) |
01:35.34 | Greek-Boy | ManxPower: Can you tell me if hangup() waits until the call is over before going to the next priority? |
01:36.19 | lunaphyte | why might asterisk not load a module after upgrading from 1.2 to 1.4? http://rafb.net/p/qaTgnw89.html |
01:38.26 | ManxPower | 1.2 binary modules do not work with 1.4 |
01:39.26 | ManxPower | One of the critera FOR an increase in the tenth's digit is breaking binary compatability. |
01:40.56 | lunaphyte | oh, ok. |
01:44.27 | lunaphyte | i have an older 12sp+ that i was using with chan_sccp, which was (is?) a third party module. might i be able to use it with asterisk without the need for the additional module anymore? |
01:47.40 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
01:47.42 | ManxPower | most modules have newer releases that support 1.4 |
01:47.56 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
01:49.15 | lunaphyte | yeah, i was just looking at the site for this module. it seems to be a bit neglected. |
01:51.35 | *** join/#asterisk akira2014 (n=chatzill@172.Red-88-8-198.dynamicIP.rima-tde.net) |
01:51.44 | akira2014 | hello |
01:51.57 | jameswf-home | oh snap |
01:52.50 | *** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com) |
01:52.52 | akira2014 | jameswf-home: ? |
01:53.06 | *** join/#asterisk BeeBuu (n=beebuu@219.135.42.4) |
01:54.04 | jameswf-home | ~random |
01:54.20 | akira2014 | :) |
01:54.49 | akira2014 | jameswf-home: can you help me ( another time ) with zaptel |
01:54.51 | akira2014 | ? |
01:54.58 | akira2014 | or some one else, plz |
01:55.46 | BeeBuu | can i dial someone and invite he/she to meet? |
01:56.53 | akira2014 | as tzafrir and others tell me i've compiled zaptel, libpri & Asterisk from sources.... |
01:59.04 | akira2014 | but i continue without being able to pick up calls |
01:59.08 | akira2014 | any ideas ? |
02:01.28 | jameswf-home | now killer jelly fish |
02:03.58 | *** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
02:04.03 | anonymouz666 | Asterisk native wav format is 16bit or 8bit? |
02:05.10 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
02:05.11 | jameswf-home | usualy 8 is the magic number |
02:06.26 | drmessano | 3 is the magic number, per Schoolhouse Rock |
02:06.44 | jameswf-home | 1 is the lonliest number |
02:06.56 | _charly_ | 42 is the answer |
02:07.16 | drmessano | Hmm.. if only I knew the question |
02:09.05 | *** join/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
02:19.42 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
02:25.07 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
02:26.32 | *** join/#asterisk jmacz (n=jmacz@190.25.38.26) |
02:33.25 | jmacz | Hi everyone, I'm having some problems with an * box which is terminating some incoming calls over a PRI because of the Telco sends a Destination out of order (27) HANGUP cause. |
02:33.38 | jmacz | This does not happens with the old PBX nor the ISDN Tester. Any ideas what it might be? |
02:35.06 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
02:37.22 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
02:41.35 | jameswf-home | jmacz: http://www.cisco.com/warp/public/121/cause_codes.html |
02:42.13 | *** join/#asterisk [hC] (n=hardcore@S0106001346a4b813.vc.shawcable.net) |
02:42.29 | Greek-Boy | I think I finally figured out a solution for emergency calls |
02:42.54 | Greek-Boy | I have this in my dial plan [macro-dialemergency] |
02:42.55 | Greek-Boy | exten => s,n,GotoIf($["${CHANNEL}" = "${ARG1}"] && $["${DIALSTATUS}}" = "CHANUNAVAIL"] && $["${EMERGENCY_STATE}" = "0"]$?softhangup:setem) |
02:42.55 | Greek-Boy | exten => s,n(softhangup),SoftHangup(${ARG1}|a) |
02:42.55 | Greek-Boy | exten => s,n,Wait(3) |
02:42.55 | Greek-Boy | exten => s,n(setem),Set(GLOBAL(EMERGENCY_STATE)=1) |
02:42.56 | Greek-Boy | exten => s,n,Monitor(wav,${EMERGENCY_REC_CALLFILENAME},m) |
02:42.58 | Greek-Boy | exten => s,n(dial),Dial(${ARG2}) |
02:43.00 | Greek-Boy | exten => s,n,Hangup() |
02:43.02 | Greek-Boy | exten => s,n,Set(GLOBAL(EMERGENCY_STATE)=0) |
02:43.04 | Greek-Boy | oops |
02:43.06 | Greek-Boy | i was going to paste a pastepin url |
02:43.14 | Greek-Boy | http://pastebin.com/d407638e2 |
02:43.19 | jameswf-home | gah |
02:43.24 | Greek-Boy | sorry :( |
02:43.45 | Greek-Boy | my ctrl+c key shortcut didn't help me on that one |
02:44.31 | jmacz | jameswf-home, had already checked that link. We tested the ISDN line with the Telco and it seems OK (passed the physical and data link tests). Besides, the Telco only sends that hangup cause when the call comes from other service provider. |
02:45.39 | jmacz | jameswf-home, say I got my PRI from provider A, if the call comes from provider B or C (specially long distance calls), the call is dropped but only with Asterisk (as I said, the call is not dropped with the old PBX nor the ISDN tester). |
02:46.18 | jmacz | any ideas? (the thing is driving me nuts) |
02:49.46 | lmadsen | Greek-Boy: not that: 1) your context doesn't start with a priority 1, 2) you have a type on the first line ${DIALSTATUS}} <-- 2 ending curly braces, 3) same line, you have a $? at the end.... you only need ?, 4) you should get into the habit of using commas instead of pipes to separate as pipes went away in 1.6 |
02:49.55 | lmadsen | s/not that/note that |
02:51.16 | *** join/#asterisk joobie (n=joobie@joobie.org) |
02:51.18 | joobie | hey guys |
02:51.51 | joobie | after a basic phone that is good quality for SIP.. ive been looking at the "Polycom SoundPoint IP 320 Phone" phones. Anyone got any recommendations, or is this a good phone to be looking at? |
02:51.59 | joobie | there's also the 330 phone.. not sure which is better |
02:52.05 | joobie | price is $50 more for the 330 |
02:52.15 | BeeBuu | how can i dial someone's phone invite he/she to meet? |
02:53.07 | lmadsen | BeeBuu: stay away from the he/she's.... bad news |
02:53.27 | BeeBuu | lmadsen: what's wrong? |
02:53.36 | lmadsen | I don't understand your question |
02:53.57 | lmadsen | and I'm going to bed... g'night! |
02:54.25 | BeeBuu | funny guy... |
02:55.17 | BeeBuu | ~book |
02:55.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
02:56.53 | joobie | guys - just about to buy a polycom phone |
02:57.01 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
02:57.12 | joobie | wondering if i want to use the internet to dial out to the sip provider.. all i need is the phone yea and a box that has ethernet? |
02:57.18 | joobie | one for the wan and one for the LAN |
02:57.18 | joobie | ? |
02:58.58 | Greek-Boy | lmadsen thanks for your help |
02:59.05 | jameswf-home | just 1 nic |
02:59.09 | Greek-Boy | lmadsen either than those problems will it work? |
03:00.17 | joobie | thanks james |
03:00.21 | joobie | also is that the best basic phone |
03:00.23 | joobie | the polycom? |
03:00.30 | joobie | i just want something that is clear.. and simple.. and works well |
03:00.46 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
03:01.32 | jameswf-home | polycom is most expensive |
03:01.42 | jameswf-home | basic I would look at aastra |
03:02.37 | joobie | ahh k |
03:02.40 | joobie | also james |
03:02.50 | joobie | hope you dont mind me bouncing these Q's off you:P |
03:02.59 | joobie | i have about 20 phones i want to hook up for cheap outbound calls |
03:03.19 | jameswf-home | analog? |
03:03.25 | joobie | do i need asterisk at all? i mean.. i can see some providers allow you to hook striaght up to them, without asterisk.. just so long as u have a sip phone |
03:03.28 | joobie | digital |
03:03.29 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
03:04.11 | jameswf-home | depends on how much control you want |
03:04.18 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-207-15.bltmmd.east.verizon.net) |
03:04.41 | joobie | what are the control mechanisms asterisk can give you ontop james? |
03:05.07 | joobie | so you're saying you can do voip phone -> asterisk -> provider .. all using SIP as the transport protcol.. in a way using asterisk as a sip gateway? |
03:05.34 | jameswf-home | asterisk gives you full controll say you only need 10 lines for 20 phones etc... |
03:05.57 | *** join/#asterisk chendy (n=chendy@58.61.40.239) |
03:06.00 | SteveTotaro | hey, with a little luck and some help with Tzafrir, I got a Junghanns quad port BRI card and a quad port Sangoma FXO board to live happily together on a US BRI no less |
03:06.11 | joobie | but then with SIP providers.. if 10 lines are in use we'll get billed the same as if 20 were yea? |
03:06.26 | jameswf-home | right |
03:06.30 | joobie | i see |
03:06.51 | joobie | but i guess from asterisk we can restrict cant we? |
03:06.59 | SteveTotaro | i would suggest you have a dedicated point to point if you are going to rely on that for business |
03:07.01 | jameswf-home | yes |
03:07.03 | joobie | like can we say "u can only dial local numbers" for particular phones |
03:07.23 | SteveTotaro | that is all done in the dialplan |
03:07.29 | joobie | SteveTotaro, how come? I thought having asterisk would be a better solution? |
03:07.39 | joobie | liek more flexible.. |
03:07.56 | SteveTotaro | asterisk is a great solution, i was talking about using all VoIP |
03:08.11 | joobie | ahhh |
03:08.26 | JT | SteveTotaro: do document US BRI workage online |
03:08.30 | SteveTotaro | either get a point to point to your voip provider or get voip from your ISP |
03:08.34 | JT | SteveTotaro: lots of people here were interested in that |
03:08.46 | JT | a number of months back |
03:09.02 | SteveTotaro | i would love to consult, gas is too damn expensive ;) |
03:09.03 | joobie | SteveTotaro, point-to-point as in from the phone to the sip provider? or the phone, to asterisk, to the sip provider? |
03:09.29 | SteveTotaro | going up to $4/gallon this spring!!!! |
03:09.29 | JT | SteveTotaro: don't look at me, where I am we get ETSI BRI :) |
03:09.39 | joobie | wow |
03:09.41 | joobie | that is steep |
03:09.44 | JT | SteveTotaro: fuel in the US is cheap as hell |
03:09.49 | JT | that's cheap |
03:09.58 | jameswf-home | thats only $1 a quart |
03:10.02 | SteveTotaro | i know and it is going to just rise |
03:10.13 | JT | $1 a litre or so |
03:10.25 | JT | fuel is around AUD$1.40/L here |
03:10.31 | SteveTotaro | in Liberia they sold gas in old plastic water bottles, it was colored red |
03:10.36 | JT | which is probably USD$1.20 |
03:10.36 | drmessano | Ed Begley Jr could care less |
03:10.40 | SteveTotaro | no working gas stations |
03:10.59 | joobie | SteveTotaro, point-to-point as in from the phone to the sip provider? or the phone, to asterisk, to the sip provider? |
03:11.04 | SteveTotaro | hey DRM, how is hanging |
03:11.07 | drmessano | He's got a car powered by douchebagedness, and he gets 1000 miles per episode of St Elsewhere he did |
03:11.13 | drmessano | Whaddup Steve |
03:11.44 | SteveTotaro | St Elsewhere, that is a name that defined an era |
03:11.53 | SteveTotaro | hillstreet blues |
03:12.26 | SteveTotaro | anyways, joobie you don't want public internet if you are serious about your business communications |
03:12.38 | drmessano | Ed Begley Jr doesn't use Asterisk because he can't get SIP to run on Solar power |
03:12.43 | drmessano | chan_douche is not finished |
03:12.53 | joobie | SteveTotaro, i am serious.. but i want to keep costs down |
03:13.03 | joobie | SteveTotaro, I was thinking about ADSL2+ |
03:13.03 | SteveTotaro | but the potato battery is money |
03:13.12 | drmessano | HA |
03:13.17 | drmessano | Potato powered PBX |
03:13.22 | SteveTotaro | how many lines are you looking at? |
03:13.23 | J4k3 | drmessano: eh, my gs bt 101's only draw about 1.5 watts (5V) while talking :P |
03:13.27 | jameswf-home | ~chan_erexic |
03:13.31 | SteveTotaro | what kind of usage? budget? |
03:13.37 | jameswf-home | ~chan_rexic |
03:13.43 | jameswf-home | doh |
03:13.46 | J4k3 | ~chan_suki |
03:13.49 | SteveTotaro | toll free? |
03:13.52 | *** join/#asterisk jmacz (n=jmacz@190.25.38.26) |
03:13.57 | joobie | SteveTotaro, cheap as possible.. without going too nasty with performance.. it's a call center.. so heaps of outbound calls.. wnat to keep costs down |
03:14.01 | joobie | that's why i was thinking of ADSL2+ |
03:14.05 | joobie | and the number of lines is 20 |
03:14.19 | SteveTotaro | then just get an LD pri |
03:14.28 | joobie | how many lines is that? |
03:14.37 | joobie | the thing is.. if i go PRI / BRI.. i pay higher call costs |
03:14.38 | SteveTotaro | 23 voice 1 data |
03:14.40 | joobie | that's what i was looking at sip |
03:15.15 | *** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
03:15.19 | SteveTotaro | what is the cost if your call center is down for one hour? |
03:15.29 | SteveTotaro | how about one day? |
03:15.37 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
03:15.54 | Daviey | Supplement outbound calls with SIP, not rely upon |
03:15.55 | SteveTotaro | the one I implemented lost $27k per hour of downtime |
03:16.04 | joobie | wow |
03:16.06 | joobie | that's a lot |
03:16.09 | SteveTotaro | Daviey has a good point |
03:16.19 | joobie | why is sip so bad? |
03:16.31 | SteveTotaro | public internet is so bad |
03:16.43 | J4k3 | joobie: theres a lot of doom/gloom going on, just get multiple itsps to make calls through |
03:16.47 | J4k3 | so if the routing goes bad, switch |
03:16.53 | joobie | itsps? |
03:16.59 | SteveTotaro | ~itsp |
03:16.59 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
03:17.02 | Daviey | s/switch/auto switch/ |
03:17.06 | joobie | ahhh |
03:17.07 | SteveTotaro | it isn't doom and goom |
03:17.13 | joobie | so if you get two |
03:17.16 | joobie | u just swtich to the other |
03:17.17 | joobie | if it's bad |
03:17.19 | J4k3 | yep |
03:17.22 | SteveTotaro | i pay $150/mo for my LD T1 PRI |
03:17.22 | J4k3 | get two internet connections |
03:17.23 | joobie | and maybe i can get two internet links |
03:17.26 | joobie | two different proiders |
03:17.26 | joobie | ya |
03:17.32 | J4k3 | yeah |
03:17.45 | joobie | ok this brings me to my next question |
03:17.51 | joobie | how much bandwidth do we actually need |
03:17.51 | SteveTotaro | might as well have two different call centers |
03:17.52 | joobie | per call |
03:18.05 | J4k3 | SteveTotaro: twice the management staff? isyounuts?! |
03:18.13 | SteveTotaro | in case there is an earthquake or terrorist attack |
03:18.15 | J4k3 | a second internet connection might cost you $250/month |
03:18.32 | joobie | Daviey, i got ya |
03:18.38 | J4k3 | joobie: depends on how bad of call quality your ears can handle. |
03:18.52 | joobie | i dont want bad quality |
03:18.57 | joobie | like.. in terms of optimal perforamnce |
03:19.00 | joobie | how much does each call need? |
03:19.02 | SteveTotaro | no it depends on what your clients will tolerate |
03:19.04 | J4k3 | ~bandwidth |
03:19.05 | jbot | bandwidth is, like, This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus. |
03:19.08 | J4k3 | ~g729 |
03:19.09 | jbot | g729 is, like, an ITU-standard voice codec which operates at 8kbps and offers quality very similar to GSM. G.729 is patent-encumbered; those wishing to use it with Asterisk must buy a license from Digium. |
03:19.11 | J4k3 | hrm |
03:19.13 | SteveTotaro | you obviously have little call center experience |
03:19.17 | jameswf-home | ~wizard |
03:19.18 | jbot | hmm... wizard is enchancement to howto's |
03:19.38 | J4k3 | http://www.inphonex.com/support/voip-codecs.php |
03:19.55 | J4k3 | calculate that, add some padding |
03:19.57 | J4k3 | etc. |
03:20.04 | Daviey | joobie: Are you sure this project isn't a bit too big for you to handle? |
03:20.15 | J4k3 | 20 lines is hard? |
03:20.16 | J4k3 | wtfbbq |
03:20.28 | joobie | SteveTotaro, i do |
03:20.31 | joobie | why? |
03:20.41 | J4k3 | joobie: you'd be plenty safe if you could get at least 1.5mbit upload |
03:21.04 | joobie | J4k3, that should be doable.. how much per individual call tho |
03:21.05 | joobie | approx |
03:21.06 | J4k3 | I think adsl2+ is asynchonous, so be sure to get close to the CO |
03:21.08 | joobie | for a good connection |
03:21.12 | J4k3 | 32k + overhead + safety |
03:21.16 | J4k3 | for g729 |
03:21.29 | J4k3 | figure 40-45k, assuming no other internet activity and/or very good qos |
03:21.35 | J4k3 | on *Both* ends of the link |
03:21.45 | joobie | aahh k |
03:21.51 | joobie | KB/s |
03:21.53 | joobie | or Kbit? |
03:21.55 | SteveTotaro | ok, get an LD PRI for $150/mo |
03:22.04 | SteveTotaro | pay a penny a minute |
03:22.07 | joobie | SteveTotaro, they are much more expensive here |
03:22.13 | SteveTotaro | where is here? |
03:22.15 | J4k3 | same here |
03:22.17 | joobie | AU |
03:22.27 | J4k3 | I'd have to backhaul to a city to some clec to do that |
03:22.29 | SteveTotaro | how much? |
03:22.34 | joobie | i got a 3 line ISDN and it costs 75$/month |
03:22.45 | SteveTotaro | isdn what bri? |
03:22.47 | J4k3 | The original 'Full Rate' GSM speech codec is named RPE-LTP (Regular Pulse Excitation Long-Term Prediction). This codec uses the information from previous samples (this information does not change very quickly) in order to predict the current sample. The speech signal is divided into blocks of 20 ms. These blocks are then passed to the speech codec, which has a rate of 13 kbps, in order to obtain blocks of 260 bits. |
03:22.49 | joobie | i think so |
03:22.51 | joobie | it's 3 line |
03:22.52 | joobie | afaik |
03:22.54 | joobie | for another client |
03:22.55 | J4k3 | ^^ for the person that was complaining about lack of gsm-fr support |
03:22.58 | joobie | plus calls are not cheap on it |
03:23.14 | SteveTotaro | sucks to be you |
03:23.28 | JT | joobie: no 3 line ISDN |
03:23.34 | SteveTotaro | maybe voip is your "best" option if cost is the main concern |
03:23.35 | J4k3 | 3 lines activated on a PRI |
03:23.46 | J4k3 | you don't have to light up all 23/30 B's |
03:23.59 | jameswf-home | I got a pri installed just to vote for american idol.... j/k |
03:24.00 | JT | J4k3: don't know of a single telco here who will light up less than 10 |
03:24.02 | J4k3 | 30 or however many b's there are on an e1 pri |
03:24.21 | J4k3 | JT: that usually depends on how much bsing you do with your telco salespeople |
03:24.31 | J4k3 | JT: with the right charm, you can get fiber pulled to your garage in timbuktu |
03:24.35 | SteveTotaro | i know one that will do it but they hand it off as analog through a channel bank |
03:25.04 | J4k3 | SteveTotaro: unplug channel bank, plug in pci card. |
03:25.05 | SteveTotaro | there are 31 channels in an e1 |
03:25.12 | SteveTotaro | 1 d 30 b |
03:25.23 | JT | sort of right |
03:25.28 | JT | there's 32 timeslots |
03:25.32 | JT | 30 B channels |
03:25.34 | JT | 1 D channel |
03:25.45 | jameswf-home | technicaly 32 but one of the channels is al magical and invisible... |
03:25.49 | SteveTotaro | sounds like what i just said |
03:25.49 | JT | 1 timeslot reserved for multiframe synch and network alarms |
03:25.57 | J4k3 | what a waste |
03:25.59 | JT | not really, there are 32 timeslots |
03:26.08 | SteveTotaro | i never mentioned timeslots |
03:26.12 | JT | TS0 is multiframe synch and network alarms |
03:26.17 | JT | TS16 is the D channel |
03:26.21 | JT | but that's all it it |
03:26.23 | JT | it is |
03:26.24 | JT | timeslots |
03:26.34 | JT | anyway, only 30 from and end user viewpoint |
03:26.41 | JT | s/and/an/ |
03:26.54 | J4k3 | ~jt |
03:26.54 | jbot | Template to compose LaTeX jewel case CD inserts. URL: http://www-stud.enst.fr/~michon/realisations.html |
03:26.59 | Daviey | shall we start getting itno US, Europe and Japan's variance? |
03:27.04 | Daviey | into* |
03:27.11 | JT | and there are no RBS E1s afaik |
03:27.15 | JT | at least not in australia |
03:27.20 | SteveTotaro | nah, africa and us is enough for me |
03:27.22 | jameswf-home | ~e1 |
03:27.22 | jbot | i heard e1 is the basic digital telephony circuit used everywhere except the US, Japan, Taiwan and Hong Kong. T1, or slight variants of it, are used in those places. E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelised, to provide 31 time slots of voice or data, each of 64kbps. Sync and alarm information occupies the remainder of ... |
03:27.50 | SteveTotaro | hrm 31 timeslots, jbot is all knowing |
03:27.52 | jameswf-home | that sucked |
03:28.19 | SteveTotaro | Sync and alarm information occupies the remainder of ... |
03:28.22 | SteveTotaro | guess not |
03:28.23 | jameswf-home | ~book is all knowing |
03:28.24 | jbot | ...but book is already something else... |
03:28.31 | jameswf-home | bah |
03:28.58 | SteveTotaro | well my ccie training taught me differently |
03:29.24 | SteveTotaro | it is the cisco way or the highway |
03:29.27 | SteveTotaro | ;) |
03:29.31 | Daviey | Cisco :( |
03:29.33 | SteveTotaro | ~cisco |
03:29.33 | jbot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks! |
03:29.34 | JT | i trust telecomms engineering training more ;) |
03:29.43 | jameswf-home | ~porn |
03:29.43 | jbot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
03:29.58 | Daviey | :O |
03:30.31 | jameswf-home | ~random |
03:30.36 | SteveTotaro | Daviey, back to your point about supplementing TDM with an ITSP |
03:31.38 | SteveTotaro | what mechanisms are there to determine when to shut off the voip |
03:31.38 | SteveTotaro | ? |
03:31.38 | Daviey | when the trunk fails :( |
03:31.38 | JT | joobie: are you in the sticks? |
03:31.38 | SteveTotaro | other than that, i am thinking along the lines of poor audio quality |
03:31.49 | Daviey | hmm, can't think of anything in * |
03:32.24 | SteveTotaro | that would be a MAJOR plus |
03:32.25 | Daviey | i suppose you could measure packet loss, use * RT and modify it outside of * |
03:33.12 | Daviey | Firstly, what problems make a call quality bad: |
03:33.19 | Daviey | * long ping times |
03:33.24 | Daviey | * packet loss |
03:33.28 | Daviey | * crappy ITSP |
03:33.29 | SteveTotaro | no, latency is not an issue |
03:33.37 | Daviey | * echo |
03:33.49 | SteveTotaro | it is but does not hurt voice quality |
03:34.09 | Daviey | hmm, UDP :) |
03:34.18 | SteveTotaro | yes udp |
03:34.25 | Daviey | packet loss DOES hurt udp |
03:34.38 | SteveTotaro | i was speaking of latency |
03:34.48 | SteveTotaro | but yes, packet loss hurts udp |
03:35.39 | Daviey | sorry. |
03:35.53 | SteveTotaro | i would say anything that could make a customer not want to stay on the phone or prohibit business would qualify |
03:36.10 | Daviey | hmm, crappy on hold music then |
03:36.12 | Daviey | IVR's |
03:36.22 | SteveTotaro | well that is expected these days |
03:36.29 | Daviey | offshore call centres |
03:36.33 | SteveTotaro | just keep pressing 0 |
03:36.40 | Daviey | heh |
03:36.58 | SteveTotaro | i setup five offshore call centers |
03:37.52 | SteveTotaro | this is one i like to show off but it was not very big http://translate.google.com/translate?hl=en&sl=fr&u=http://www.osiris.sn/article1636.html&sa=X&oi=translate&resnum=3&ct=result&prev=/search%3Fq%3Dbeth%2Bpayne%2Bsenegal%2Bcomputer%2Bfrontiers%26hl%3Den%26client%3Dfirefox-a%26rls%3Dorg.mozilla:en-US:official%26hs%3DgKJ |
03:38.09 | anonymouz666 | SteveTotaro: using hardphone/softphone or ATA? |
03:38.27 | SteveTotaro | this was a funny situation |
03:38.55 | SteveTotaro | to close the deal i had to use a system that had a name, so I used a 3com PBX |
03:39.18 | SteveTotaro | and a "authentication" server with a quad port E1 card |
03:40.50 | SteveTotaro | CSC and the State Dept were not buy "OpenSource Solution" so wording was very important |
03:41.34 | anonymouz666 | that happens all the time |
03:42.07 | SteveTotaro | i was a one man show with 13 bosses, first time in africa |
03:42.22 | SteveTotaro | E1 cost $3k/mo |
03:42.54 | SteveTotaro | Then when Sonatel (the monopoly) found out what it was for, they wanted more money |
03:43.12 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
03:46.52 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
03:47.12 | *** part/#asterisk beek (n=klinebl@static-71-240-222-16.alt.east.verizon.net) |
03:50.05 | joobie | back |
03:50.07 | joobie | sorry phone |
03:50.15 | joobie | JT, sorta.. about 30kms from the city |
03:50.24 | SteveTotaro | i hate phones |
03:50.42 | JT | joobie: which city? |
03:50.49 | jameswf-home | most eople who work with phones for a living hate phones |
03:50.58 | jameswf-home | people even |
03:51.51 | SteveTotaro | i RARELY answer |
03:52.11 | joobie | JT, melb |
03:52.14 | SteveTotaro | good about returning calls though |
03:52.17 | joobie | are you from AU too JT? |
03:52.20 | JT | joobie: sydney |
03:52.24 | joobie | cool :) |
03:52.28 | joobie | g'day mate |
03:52.29 | joobie | :P |
03:52.37 | JT | joobie: you should be able to get decent PRI pricing |
03:52.37 | SteveTotaro | but email is such a better way to communicate |
03:52.39 | JT | hi :) |
03:52.52 | joobie | JT with who?.. is PRI just 3 lines? |
03:52.57 | joobie | because i need 20 lines outbound... |
03:53.10 | JT | joobie: 10-30 |
03:53.37 | jameswf-home | pri is 23 |
03:53.42 | jameswf-home | max |
03:53.46 | JT | joobie: not in australia |
03:53.47 | SteveTotaro | e1 pri is 30 |
03:54.02 | JT | jameswf-home: even |
03:54.03 | joobie | JT.. but line rental on that is like 25$ a line ya? |
03:54.10 | jameswf-home | you can get a partial pri |
03:54.11 | JT | joobie: nope |
03:54.16 | joobie | how much JT? |
03:54.22 | JT | jameswf-home: like i said, 10-30 was correct for australia |
03:54.26 | joobie | also call costs are standard too.. they wont compare to voip call costs.... |
03:54.46 | JT | joobie: don't be so sure |
03:54.50 | SteveTotaro | (10:52:06 PM) jameswf-home: pri is 23 |
03:55.00 | SteveTotaro | (10:52:11 PM) jameswf-home: max |
03:55.06 | JT | joobie: quality and realiability is much better |
03:55.10 | jameswf-home | is there an echo? |
03:55.15 | JT | joobie: and you can get good call rates |
03:55.17 | joobie | JT, what are the costs for like a 20 line pri? |
03:55.24 | JT | jameswf-home: 23 is for a T1 PRI |
03:55.34 | JT | jameswf-home: we do not use T1 PRI in Australia |
03:55.38 | SteveTotaro | you just said pri is 23 max |
03:56.15 | SteveTotaro | they use a much better e1 |
03:56.23 | SteveTotaro | i wish we did as well |
03:56.32 | SteveTotaro | but BRI has to go |
03:56.40 | jameswf-home | ma bell is the future they dont change |
03:56.52 | jameswf-home | we are lucky we have AC power |
03:57.05 | SteveTotaro | i prefer DC power |
03:57.13 | jameswf-home | U.S. is often bass ackwards |
03:57.24 | JT | joobie: umm, $250/mo maybe |
03:57.30 | SteveTotaro | GO METRIC!!! |
03:57.31 | JT | maybe $300 |
03:57.33 | JT | depends on telco |
03:57.35 | joobie | JT with who |
03:57.40 | JT | joobie: Primus |
03:57.43 | joobie | hmm |
03:57.46 | joobie | interesting |
03:57.50 | JT | Optus will be clost to $400 |
03:57.51 | SteveTotaro | minute costs? |
03:57.52 | JT | close |
03:58.01 | JT | SteveTotaro: to call what? |
03:58.02 | SteveTotaro | he is outbound |
03:58.20 | SteveTotaro | not sure, joobie, who you telemarketing to? |
03:58.28 | JT | SteveTotaro: the costs for calls would vary depending on where you call |
03:58.33 | joobie | SteveTotaro, local and international |
03:58.57 | SteveTotaro | so get an itsp for your international |
03:59.15 | joobie | well |
03:59.17 | joobie | that was my next q |
03:59.21 | joobie | if i go the route of sip |
03:59.28 | joobie | are there any decent AU based ones? |
03:59.31 | SteveTotaro | obviously rates vary but i got a flat rate of a penny a minute anywhere in the US |
03:59.40 | JT | local calls are 10c+gst untimed with primus |
04:00.25 | SteveTotaro | i am unfamiliar with the terms "10c+gst untimed" |
04:00.35 | joobie | Ongoing/monthly costs |
04:00.36 | joobie | ISDN 10 lines (PRI) = $ 345 (based on Telstra |
04:00.38 | joobie | that is from whirlpool |
04:00.39 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
04:00.58 | joobie | AAPT's ABC product may be suitable. It can provide ISDN10/20/30 and internet access upto 2M over the same tail greatly reducing costs (no install and $20/month/line).You dont need to stick with ISDN10/20/30 either you can choose any number of lines and only pay for what you need (ISDN14 for example) |
04:01.07 | joobie | another whirlpool post |
04:01.14 | joobie | so that would be |
04:01.17 | JT | joobie: only if someone is moronic enough to choose telstra, or unlucky enough to have them as only choice |
04:01.24 | joobie | 400$ / month just for rental |
04:01.26 | JT | joobie: forget about whirlpool |
04:01.32 | JT | joobie: and quote it yourself |
04:01.42 | JT | primus is the cheapest if you can get it |
04:01.46 | JT | except maybe verizon |
04:01.53 | JT | verizon has steep install charges |
04:02.01 | JT | and very hard to talk to sales |
04:02.12 | SteveTotaro | verizon in Australia, cool |
04:02.23 | J4k3 | verizon, haha |
04:02.25 | J4k3 | sucks2bu |
04:02.29 | JT | yeah they bought MCI Worldcom and UUnet iirc |
04:02.41 | SteveTotaro | at least you have options unlike some countries |
04:03.03 | J4k3 | yeah |
04:03.04 | JT | SteveTotaro: gst is 10% |
04:03.09 | JT | goods and services tax |
04:03.32 | joobie | JT.. what about SIP providers, as an alternative |
04:03.38 | JT | it's payable at invoice time but is input tax creditable |
04:03.42 | JT | joobie: they mostly suck |
04:03.42 | joobie | know any good ones.. i mean ill look down the PRI path.. but want to compare against native sip |
04:03.47 | jameswf-home | diahria or frozen vomit.... sometimes options arent all they are cracked up to be |
04:03.49 | JT | but perhaps Isphone, iVox |
04:03.50 | joobie | mostly? there is hope? |
04:04.52 | joobie | JT, are they decen tin melb too tho? or just sydn |
04:05.10 | JT | joobie: they're big enough for it not to matter in such major metro areas |
04:05.31 | JT | symbio networks might also be an option |
04:05.51 | SteveTotaro | i think frozen vomit is a much better choice |
04:06.06 | SteveTotaro | ever see "joe dirt" |
04:06.31 | SteveTotaro | at least you don't need a VSAT |
04:06.53 | JT | joobie: but on the "forget about it" list: engin, mynetfone, faktortel, koala telecom |
04:07.15 | JT | engin is not that bad, they're just not that good |
04:07.22 | SteveTotaro | JT, i think you should qualify the forget about it list |
04:07.39 | joobie | hehe |
04:07.39 | JT | mynetfone will only do 1 phone call per account - stupid |
04:07.45 | joobie | JT.. out of all those - which do u think is the best |
04:07.45 | JT | koala telecom - clowns |
04:07.52 | joobie | out of theones u suggested |
04:07.58 | JT | faktortel - clowns whose sydney dids were down for over 2 weeks |
04:08.12 | JT | engin - random dropouts and sometimes there's echo |
04:08.12 | drmessano | Did someone say Clowns? |
04:08.13 | SteveTotaro | faktortel just sounds dumb |
04:08.18 | J4k3 | ass clowns |
04:08.24 | drmessano | ~happyclownpbx |
04:08.30 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, and it pwns |
04:08.30 | JT | other than that, they're ok, unlimited concurrent calls |
04:08.30 | drmessano | :( |
04:08.54 | SteveTotaro | for real about happyclownpbx? |
04:08.55 | JT | joobie: btw, my favourite ITSP for personal use: Pennytel |
04:09.01 | SteveTotaro | never heard of it |
04:09.04 | JT | i dunno if i'd use them for business |
04:09.15 | JT | cheapest rates, quality is usually good |
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04:09.21 | JT | sometimes there are hiccups though |
04:09.56 | jameswf-home | I hear its based on web 3.0 |
04:10.20 | joobie | ahh |
04:10.24 | joobie | Jt.. but for business use |
04:10.28 | SteveTotaro | what is that age old saying |
04:10.29 | joobie | whcih one do u think is best? |
04:10.46 | SteveTotaro | price, quality, support, pick two (thats not it though) |
04:11.13 | SteveTotaro | actually maybe it is |
04:12.03 | J4k3 | well, its also vs volume |
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04:12.10 | J4k3 | you can get great prices if you're pushing 100k minutes/month, etc. |
04:12.15 | jameswf-home | I thought it was you cant polish a turd but if you can convince enough people its a diamond you will get a million downloads |
04:12.31 | J4k3 | ie - decent sized 24x7 international call center |
04:13.02 | drmessano | <jameswf-home> I thought it was you cant polish a turd but if you can convince enough people its a diamond you will get a million downloads <---- Sorry, but that's trademarked by trixbox |
04:13.29 | SteveTotaro | you can print ink on paper and convince people it is worth something (the dollar) |
04:13.32 | drmessano | I'm gonna need you to disease and decyst immediately |
04:14.31 | J4k3 | SteveTotaro: I can hand you a piece of plastic, you swipe it, and *hand it back to me* |
04:14.36 | J4k3 | for lots more than a dollar :) |
04:14.47 | SteveTotaro | or less |
04:15.06 | SteveTotaro | heck just give me the numbers on #ccnumbers |
04:15.34 | jameswf-home | #ccnumbers has 0 users |
04:15.51 | SteveTotaro | you dont know what irc server |
04:15.56 | SteveTotaro | very few do |
04:16.04 | J4k3 | thanks to channel logging and google, it'll have 17 secret service agents in it tomorrow afternoon |
04:16.34 | SteveTotaro | do you think they care? |
04:16.42 | J4k3 | yeah |
04:16.46 | J4k3 | they don't like bullshit-paper competition |
04:16.52 | SteveTotaro | it's part of the interest rate and fees paid by the merchants |
04:17.39 | SteveTotaro | chargebacks just cost merchants |
04:18.12 | J4k3 | which increases costs which increases prices |
04:18.37 | SteveTotaro | and all banking wants a paperless economy |
04:18.52 | J4k3 | haha |
04:18.58 | SteveTotaro | that's fine but secret service doesn't give a damn |
04:19.04 | J4k3 | so instead of bullshit dollars that people occasionally pay tax on |
04:19.10 | SteveTotaro | that is why the US dropped the $1k bill |
04:19.16 | J4k3 | they'll barter, tax free* |
04:19.25 | J4k3 | (* - not legally, but who can track?) |
04:19.27 | SteveTotaro | there is a huge barter movement |
04:19.50 | SteveTotaro | i barter as much as possible |
04:20.01 | SteveTotaro | admin a PBX for colo space |
04:20.02 | jameswf-home | too bad cant barter gas |
04:20.32 | J4k3 | I need an electric car |
04:20.39 | J4k3 | power here is considerably cheaper than gas for now. |
04:20.45 | SteveTotaro | if you own a farm you can get tax free gas, much cheaper, but "illegal" to use in regular street vehicles |
04:21.36 | J4k3 | usually its marked with dye |
04:21.47 | J4k3 | some high output UV LEDs in your fuel tank can fix that quickly |
04:22.33 | ManxPower | I wonder if you could get the fuel to generate the power to run the electric car... |
04:23.04 | SteveTotaro | BGE (Baltimore Gas and Electric) just deregulated, prices have jumped 50% immediately |
04:23.16 | J4k3 | some 3v LEDs beats the hell out of the in-tank electric fuel pump running 14-15v |
04:23.19 | JT | i found a manufacturer who make luxeon style high output UV LEDs |
04:23.24 | J4k3 | yay for electromechanical devices immersed in fuel |
04:23.34 | JT | hell, they make almost any wavelength LED in a high output multi watt version |
04:23.38 | JT | from infrared to UV |
04:23.47 | SteveTotaro | i breaking the tail light but not the filament and putting that in the gas tank |
04:24.29 | SteveTotaro | turn on your headlights or hit the breaks |
04:24.49 | J4k3 | not enough oxygen to do anything spectacular |
04:24.59 | jameswf-home | J4k3: dream killer |
04:25.18 | SteveTotaro | with the right additive... |
04:25.33 | SteveTotaro | i don't give all my special ops secrets away |
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04:27.38 | SteveTotaro | who need o2 when you have N2o |
04:27.54 | jameswf-home | ~~ |
04:27.55 | jbot | Every moment in which I'm called upon is torture. |
04:27.58 | jameswf-home | ~~~~~ |
04:27.59 | jbot | grrrr |
04:30.08 | TJNII | ~~~ |
04:30.09 | jbot | I HATE YOU, tjnii!!! |
04:30.23 | Nivex | ~die |
04:30.24 | jbot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
04:30.41 | Daviey | ~people_are_bored |
04:31.42 | jameswf-home | ~taco |
04:31.43 | jbot | TACO TACO TACO! |
04:31.53 | SteveTotaro | ~verizon |
04:31.53 | jbot | Verizon is utter garbage. Do yourself a favor and stay away from that company. |
04:32.01 | SteveTotaro | ~qwest |
04:32.02 | jbot | i guess qwest is a company with secksie backbones but lame peering (www.qwest.net). or a company that randomly scrambles routes and pisses off network engineers worldwide |
04:32.06 | jameswf-home | ~vonage |
04:32.07 | jbot | from memory, vonage is a bunch of monkeys |
04:32.36 | joobie | hey JT |
04:32.37 | jameswf-home | ~canada |
04:32.37 | jbot | We're really sorry we beat you at hockey in the olympics, it's just that we're much much much much MUCH better than you. |
04:32.39 | SteveTotaro | ~jameswf |
04:32.40 | jbot | jameswf loves unsolicited technical support |
04:32.46 | joobie | i just spoke to iprimus |
04:33.13 | jameswf-home | ~fonality |
04:33.13 | jbot | Fonality is hiring, "no Asterisk knowledge needed" or, "I just installed asterisk now what?", and will be the first guys against the wall when the revolution comes! |
04:33.17 | SteveTotaro | ~jameswf-home |
04:33.28 | joobie | 0m contract - $2,000install.. 12m contract - $1,000 install.. 24m contract - $0 install ............ $250/month for ISDN20 .. all costs exclude GST |
04:33.44 | SteveTotaro | $2k install!!! |
04:33.50 | joobie | in terms of call costs he's going to send that through on the email |
04:34.01 | joobie | SteveTotaro, that's AUD.. |
04:34.15 | J4k3 | disposable aussiedollars. |
04:34.27 | jameswf-home | ~jameswf-home |
04:34.27 | jbot | when -home is added it means he is on his own time dont call his boss |
04:34.28 | SteveTotaro | so what does that equate to U$D, like $4k? |
04:35.02 | Daviey | ~SteveTotaro |
04:35.03 | jbot | methinks stevetotaro is an IRC nub |
04:35.20 | jameswf-home | ~unixdog |
04:35.21 | jbot | <unixdog> Everyone use BSD gah linux sux progress is overrated use my project gah |
04:35.21 | SteveTotaro | see, all knowing |
04:35.35 | joobie | SteveTotaro, it's around $1,850 USD |
04:36.04 | Daviey | SteveTotaro: $2K install is bargain! |
04:36.08 | SteveTotaro | that is a tidy bit |
04:36.21 | SteveTotaro | my install was waived |
04:36.40 | J4k3 | unixdog speaks the truth, linux *does* indeed suck |
04:36.51 | Daviey | One of my clients paid >$15K |
04:36.51 | SteveTotaro | it is an LD circuit though so I pay for inbound and outbound |
04:37.19 | jameswf-home | ~linux |
04:37.37 | jbot | linux is, like, the cure for cancer, AIDS and slavery to corporations |
04:37.37 | SteveTotaro | penny a minute all tollfree |
04:38.00 | Daviey | ~ubuntu |
04:38.06 | SteveTotaro | ~fedora |
04:38.06 | jbot | i guess fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
04:38.10 | J4k3 | ~nug |
04:38.10 | jbot | i guess nug is a verb for when your girlfriend graciously feeds you chicken nuggets while you're driving. a lot of fun. any girl who does this for you is a keeper. |
04:38.20 | J4k3 | hahaha |
04:38.22 | J4k3 | wor dup |
04:38.42 | Daviey | fedora :( |
04:38.49 | SteveTotaro | ~fedora |
04:38.49 | jbot | rumour has it, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
04:38.53 | Daviey | fedora :( |
04:38.55 | J4k3 | ~sexualchocolate |
04:38.55 | jbot | I told you that boy could sing |
04:39.00 | SteveTotaro | who put that in there? |
04:39.49 | SteveTotaro | daviey, do you see your name or is it mine? |
04:39.55 | SteveTotaro | ~fedora |
04:39.56 | jbot | from memory, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
04:40.19 | TJNII | ~inflatable dildos |
04:40.19 | jbot | Oh yea, stretch that hole! |
04:40.28 | Daviey | < jbot> rumour has it, fedora is Daviey is <action> tells Daviey that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
04:40.30 | SteveTotaro | gross |
04:40.39 | jameswf-home | lmao |
04:40.46 | SteveTotaro | from memory, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
04:40.51 | Daviey | SteveTotaro is irc nub :) |
04:41.01 | SteveTotaro | true enough |
04:41.07 | J4k3 | ~yourmom |
04:41.07 | jbot | i guess yourmom is a man |
04:41.09 | JT | SteveTotaro: install is waived if you get a 24m contract ;) |
04:41.11 | SteveTotaro | Qwell put that in |
04:41.19 | SteveTotaro | ~qwell |
04:41.19 | jbot | i guess qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
04:41.33 | Daviey | JT: gah, you need at least 5 years to get it waived in the UK |
04:41.53 | JT | joobie: i was right on the money for monthly cost there it seems pretty standard |
04:41.59 | Daviey | well for a high costing one |
04:42.00 | SteveTotaro | 1 year waived, no monthly option |
04:42.01 | JT | Daviey: wow, that's an incredibly long contract |
04:42.03 | JT | for pri... |
04:42.59 | SteveTotaro | took qwest about 50 calendar days to get it installed and provisioned correctly |
04:43.14 | Daviey | I had a customer that wanted fibre net access and PRI to 3 sites from the telco.. cost them nerly $200K |
04:43.19 | SteveTotaro | g'night all |
04:43.30 | Daviey | SteveTotaro: whats the time there? |
04:43.37 | SteveTotaro | going to do some app_rpt stuff tomorrow, should be cool stuff |
04:43.47 | jameswf-home | It took qwest 5 techs and 2 weeks to get me a second pots line... only got fixed cause I threatened to go up the pole my self |
04:43.51 | SteveTotaro | 11:43 says my atomic clock |
04:44.15 | Daviey | Hmm, i should probably go to bed.. |
04:44.21 | Daviey | Mon Mar 3 04:44:20 GMT 2008 |
04:44.23 | Daviey | nn |
04:44.44 | SteveTotaro | mostly site survey stuff tomorrow but need to be creative |
04:45.12 | J4k3 | I hate marketing |
04:45.15 | SteveTotaro | and send out some bills |
04:45.31 | jameswf-home | its only 21:45 here |
04:45.31 | JT | SteveTotaro: app_rpt is neat |
04:45.35 | SteveTotaro | you should outsource your marketing |
04:45.46 | J4k3 | SteveTotaro: no money to do that |
04:45.53 | SteveTotaro | yes, i have three high end repeaters to play with |
04:46.30 | jameswf-home | have to spend $$$ to make $$$ |
04:46.34 | SteveTotaro | sometimes the money you don't think you have can be made by freeing up time spent on things you hate and doing things you like |
04:47.25 | SteveTotaro | especially if you are not a marketing guy, could be wasted time |
04:49.25 | joobie | JT ya u were |
04:50.10 | JT | yes |
04:52.01 | joobie | call rates tho |
04:52.06 | joobie | that might be a different story |
04:52.15 | JT | joobie: they all depend on spend |
04:52.19 | JT | the more traffic you have |
04:52.22 | JT | the cheaper they are |
04:52.37 | joobie | hey guys |
04:52.41 | joobie | ahh |
04:52.43 | joobie | well |
04:52.50 | joobie | what about analogue vs digital handsets |
04:53.03 | JT | that's a no brainer... SIP :) |
04:53.12 | joobie | why?:P |
04:53.15 | joobie | the cost is 150$ / handset |
04:53.18 | joobie | if i go analogue |
04:53.21 | joobie | i can get them much cheaper |
04:53.29 | joobie | and then put an analoge card in asterix box |
04:53.40 | JT | because analogue handsets have almost no features |
04:53.45 | JT | and analogue ports are not free |
04:53.51 | joobie | how much are they |
04:54.03 | joobie | like i can get an analogue for for say 50$ |
04:54.07 | joobie | so i save 100$ per handset |
04:54.19 | SteveTotaro | get a tenor AX 24port sip to FXS |
04:54.27 | JT | but... they're crap |
04:54.32 | joobie | heh |
04:54.33 | SteveTotaro | callcenter phones get beat on |
04:54.59 | SteveTotaro | just go with the standard plantronics analog headsets |
04:55.16 | joobie | i was thinken polycom 330 |
04:55.18 | joobie | err 320 |
04:55.21 | SteveTotaro | and a tenor ax (at least that is what I have had great luck with) |
04:55.23 | *** part/#asterisk duri (n=mduregon@c-76-105-157-51.hsd1.or.comcast.net) |
04:55.41 | SteveTotaro | i keep the expensive stuff in the NOC |
04:55.46 | joobie | http://voip-warehouse.com.au/polycom-soundpoint-ip-320-phone-p-4516.html |
04:55.48 | joobie | check out that price |
04:55.53 | joobie | 159$ |
04:56.02 | SteveTotaro | and put thin clients and crappy analog headsets on the floor |
04:56.46 | JT | analogue phones lack call handling ability |
04:56.56 | joobie | fark |
04:56.58 | joobie | they are steep tho |
04:57.08 | SteveTotaro | for an outbound call center?? |
04:57.12 | joobie | SteveTotaro, |
04:57.16 | SteveTotaro | what functionality do you need? |
04:57.16 | joobie | 2900$ for that |
04:57.21 | joobie | may as well get the polycom handsets |
04:57.22 | joobie | it'll be cheaper |
04:57.23 | joobie | and digital |
04:57.35 | joobie | just route it straight to a standard etehrnet nic |
04:57.40 | SteveTotaro | it will be cheaper until they are abused and broken |
04:58.03 | JT | not all callcentres are in the bronx ;) |
04:58.09 | joobie | ahahah |
04:58.13 | SteveTotaro | i have never seen a call center agent give a sh*t about the equipment |
05:01.50 | joobie | btw |
05:02.01 | joobie | wat about asterisk at home vs the full product |
05:02.14 | joobie | can i get awawy with the gui of @home? |
05:02.21 | joobie | for 20 line setup |
05:02.24 | JT | asterisk at home hasn't been updated for years |
05:02.30 | joobie | ahh |
05:02.54 | joobie | i hear the full asterisk is a PITA to setup |
05:03.07 | JT | ~book |
05:03.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
05:03.42 | joobie | nice.. thanks jt |
05:04.17 | joobie | u kno i think google books has\ most ofthe oreilly books for free |
05:04.24 | joobie | its odd |
05:04.33 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
05:05.05 | *** part/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com) |
05:06.28 | joobie | see that is just weird |
05:06.28 | joobie | oreilly charge to view that book |
05:06.28 | joobie | normally |
05:06.28 | joobie | online |
05:06.28 | joobie | but u can get it free |
05:06.30 | joobie | duno wats up with that.. they changed their model or sumthen |
05:08.24 | MatBoy | wow, the asterisk.org site is slow most of the time |
05:08.32 | MatBoy | actually the digium site |
05:09.27 | JT | joobie: the asterisk books from o'reilly have always been freely downloadable |
05:10.04 | MatBoy | JT, I think no-one ever would have bought it when it was a payed version |
05:10.18 | JT | "payed"? |
05:10.21 | joobie | u sure JT? |
05:10.21 | jameswf-home | hmmmm http://travel.state.gov/passport/ppt_card/ppt_card_3926.html |
05:10.23 | MatBoy | paid |
05:10.24 | MatBoy | :) |
05:10.25 | joobie | i thought they charged |
05:10.26 | JT | joobie: yes |
05:10.28 | joobie | they used to have demo books |
05:10.30 | joobie | like a few chapters |
05:10.33 | joobie | the rest was paid |
05:10.38 | JT | joobie: for this book. |
05:10.53 | joobie | ahh certain books are free? |
05:10.59 | joobie | not all? |
05:11.01 | MatBoy | jooby asteriskdocs.org |
05:11.06 | JT | i know this one is available for free |
05:11.29 | MatBoy | there you can download it |
05:12.07 | joobie | ya got it bro |
05:12.07 | joobie | thanks |
05:12.29 | MatBoy | ok, cu you in the bronx :P |
05:12.57 | joobie | werd my bruda from anutha mutha |
05:13.01 | MatBoy | It's Bell day today !! |
05:13.11 | joobie | bell day? |
05:13.13 | MatBoy | yep |
05:13.23 | MatBoy | http://en.wikipedia.org/wiki/Alexander_Graham_Bell |
05:13.35 | MatBoy | would be nic eto add into the topic for fun :) |
05:13.38 | jameswf-home | ma bell day |
05:13.49 | MatBoy | jameswf-home, hey ! |
05:13.56 | MatBoy | Asterisk made me sleep better and wake up better |
05:13.57 | MatBoy | :) |
05:14.09 | jameswf-home | Asterisk can cure cancer |
05:14.15 | jameswf-home | ~asterisk |
05:14.15 | jbot | somebody said asterisk was the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org |
05:14.25 | jameswf-home | thats not it |
05:14.30 | MatBoy | ow |
05:14.56 | MatBoy | no really, I have a great feeling about it |
05:15.03 | MatBoy | this is the OSS solution that I needed |
05:16.31 | MatBoy | jameswf-home, but what did you actually mean with it ? |
05:17.14 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
05:18.42 | *** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com) |
05:20.22 | MatBoy | I'm actually figuring out what is the best to have national numbers with the country area that are in the system direcly called to sipfriends and what is not known in the system is routed to the outsideworld |
05:23.41 | MatBoy | I think a default prefix |
05:29.28 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
05:29.44 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
05:36.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
05:38.32 | *** part/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com) |
05:40.40 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.88.227) |
05:41.55 | *** join/#asterisk AJayMN (n=myemail@71-82-218-158.dhcp.mdsn.wi.charter.com) |
05:42.48 | AJayMN | If Im using g729.. the phone will use 1 license.. but if my provider makes me connect to them with g729 with that call is that another g729 count? or is that considered a pass-thru and only registers as 1? |
05:42.51 | jameswf-home | ~sleep |
05:42.51 | jbot | i guess sleep is overrated, and a poor substitute for caffeine. |
05:43.16 | MatBoy | jameswf-home, hehe, I almost didn't slept for 2 days |
05:43.24 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
05:43.53 | newmember | Can I run two instances of a SIP phone on the same computer? |
05:43.54 | jameswf-home | some Jack-Ares on the plane got me sick.... |
05:44.12 | MatBoy | Jack-Ares ? |
05:44.19 | jameswf-home | arse |
05:44.25 | MatBoy | ow hehe |
05:44.50 | MatBoy | newmember, yes, but it won't woirk... maybe you can bind some tools to seperate IP's on your PC ? |
05:44.57 | MatBoy | dunno if those clients exist |
05:45.12 | Der-Tim | hi there |
05:45.21 | MatBoy | at least, I thought it was not possible |
05:45.22 | Der-Tim | good morning from germany |
05:45.27 | newmember | MatBoy: interesting ID |
05:45.35 | newmember | MatBoy: interesting idea |
05:46.31 | MatBoy | newmember, yeah, but I didn't see any clients yet that can be bound to an IP, but I never needed it.. it's actually very nice for testing, so let me know if you want when you found something. I now do it using my VM and Host or a laptop next to me |
05:50.20 | AJayMN | any of you guys using g729? |
05:53.05 | *** join/#asterisk Wayhigh (i=noid@www.kevinlynn.com) |
05:54.14 | drmessano | AJayMN |
05:54.39 | drmessano | If your device supports G729 and your ITSP is using G729, that is passthru |
05:54.59 | *** join/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com) |
05:56.42 | AJayMN | so it would only hit the count if they were in a conference room on the local system? |
06:00.30 | *** join/#asterisk salzh (n=salzh@124.77.124.252) |
06:01.48 | drmessano | If there is transcoding, its using a license |
06:01.52 | drmessano | or licenses |
06:04.22 | drmessano | You're welcome |
06:04.26 | drmessano | Jerk |
06:04.39 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
06:08.56 | BeeBuu | drmessano: are you still there? |
06:10.04 | drmessano | yes |
06:10.33 | BeeBuu | how can i call someone and invite to meet? |
06:10.57 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
06:12.06 | drmessano | You can call them and transfer to meet |
06:12.17 | drmessano | or have them call you and transfer them |
06:12.28 | drmessano | You still trying to do this multi server conference? |
06:12.36 | BeeBuu | which command can transfer to meet? |
06:12.48 | BeeBuu | drmessano: yes,i am. |
06:12.58 | drmessano | Have you googled like I advised you? |
06:13.22 | BeeBuu | yes, just get ChannelRedirect() |
06:13.29 | BeeBuu | but that's beyong me |
06:13.36 | drmessano | Nope |
06:13.37 | drmessano | http://www.trixbox.org/forums/trixbox-forums/help/multi-site-conference |
06:13.41 | drmessano | Read that |
06:13.57 | BeeBuu | thanks,drmessano,you are so nice.. |
06:14.19 | BeeBuu | let me check it... |
06:15.12 | drmessano | That's not a perfect solution, but it's a damn good try at it.. I'm sure someone can polish it up a lot more and make it bulletproof |
06:23.22 | craigk | does anybody know if it is 'faster' or 'better' to use the asterisk built in berkley db, or AGI and an external database ? I guess I am looking for an opinion on speed v reliability v features comparing berkley db to AGI and mysql/sqlite |
06:27.04 | *** join/#asterisk Abu-Abudrahman (n=chatzill@84.36.147.94) |
06:29.29 | Nugget | bdb wins in speed and in reliability. |
06:29.57 | Nugget | mysql is a big lose on the complexity and reliability fronts, and will *always* be a more poorly tested configuration. |
06:30.51 | *** join/#asterisk steliosk (n=Stelios@85.75.211.185) |
06:32.04 | *** join/#asterisk chendy (n=chendy@202.122.97.200) |
06:39.09 | Abu-Abudrahman | am trying to install asterisk the last versions using svn but when i run the ./configure i got that error "configure: error: C preprocessor "/lib/cpp" fails sanity check" OS is Solaris 10 |
06:39.22 | Abu-Abudrahman | any suggestions ? |
06:39.34 | Nugget | why are you using the development code from svn? |
06:39.55 | Abu-Abudrahman | i was following instruction on digium fourms |
06:40.14 | Abu-Abudrahman | n the author wrote the howto usin svn |
06:40.40 | Nugget | you're sure you have a c++ compiler installed on the machine |
06:40.40 | Nugget | ? |
06:40.52 | Abu-Abudrahman | yes am sure |
06:42.00 | Abu-Abudrahman | bash-3.00# pkg-get -i gcc3g++ |
06:42.02 | Abu-Abudrahman | WARNING: gpg not found |
06:42.03 | Abu-Abudrahman | No worries... you already have version 3.4.5 of gcc3g++ |
06:42.23 | BeeBuu | i want dial out and auto play something to callee? |
06:50.18 | BeeBuu | ~ |
06:50.23 | BeeBuu | ~~ |
06:50.23 | jbot | Every moment in which I'm called upon is torture. |
06:50.47 | drmessano | ~~~~~~~~ |
06:50.47 | jbot | You know, this got old a long time ago. |
06:50.51 | drmessano | ~~~~~~~~~ |
06:50.51 | jbot | I'm ignoring you now. |
06:50.55 | drmessano | ~~~~~~~~~~ |
06:51.00 | drmessano | bah |
06:51.04 | drmessano | ~~~~~~~~~~~ |
06:51.04 | jbot | No, really, STOP! |
06:51.08 | drmessano | ~~~~~~~~~~~~ |
06:51.08 | jbot | ~~~~~~~~~~~ are YOU READY?????????? ~~~~~~~~~~~~~~~~~~ |
06:51.12 | drmessano | ~~~~~~~~~~~~~ |
06:51.16 | drmessano | ~~~~~~~~~~~~~~ |
06:51.19 | drmessano | Hmm |
06:51.26 | drmessano | ~~~~~~~~~~~~~ |
06:51.28 | drmessano | ~~~~~~~~~~~~ |
06:51.28 | jbot | ~~~~~~~~~~~ are YOU READY?????????? ~~~~~~~~~~~~~~~~~~ |
06:51.33 | drmessano | ~~~~~~~~~~~~~~~~ |
06:51.48 | drmessano | At some point it's supposed to shoot me or something |
06:51.55 | drmessano | ~~~~~~~~~~~~~ |
06:52.05 | drmessano | ~~~~~~~~~~~~~~ |
06:52.07 | *** part/#asterisk hads (n=hads@reef80.anchor.net.au) |
06:52.13 | drmessano | ~~~~~~~~~~~~~~~~~~~~ |
06:52.47 | drmessano | dicks? |
06:56.08 | *** join/#asterisk ArashHemmat (n=ArashHem@91.184.88.227) |
06:57.55 | J4k3 | ~no |
06:57.55 | jbot | YES |
06:58.06 | J4k3 | ~norway |
06:58.06 | jbot | rumour has it, norway is a country in Scandinavia. A beer costs EUR 1,80 there. The capital of Norway is Oslo. And yes norway isn't a member of the EU, or a great skiing nation, or actually it is a great skiing nation, or make sure you know where your towel is before you go |
06:58.15 | J4k3 | ~kenya |
06:58.15 | jbot | hmm... kenya is Where can you find Lions? Only http://mastaile.mine.nu/kenya1.mov !, or http://www.weebls-stuff.com/toons/kenya/ |
06:58.26 | J4k3 | jbot: forget norway! |
06:58.26 | jbot | i didn't have anything called 'norway!' to forget, J4k3 |
06:58.46 | drmessano | ~dubai |
06:58.55 | J4k3 | dubai is for smoking |
06:59.28 | drmessano | ~dubai |
06:59.28 | jbot | Halliburton! |
06:59.32 | J4k3 | haha |
06:59.34 | drmessano | better |
06:59.35 | J4k3 | ~bush |
06:59.35 | jbot | somebody said bush was chick plumbing or the current president and potential dictator, or the guy that made stupidity fashionable. |
06:59.46 | J4k3 | ~cheney |
06:59.49 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
07:00.06 | J4k3 | jbot: cheney is <reply> dick |
07:00.06 | jbot | J4k3: please, watch your language. |
07:00.13 | drmessano | ha |
07:00.15 | J4k3 | wtf |
07:00.27 | J4k3 | dick is a name, not a 'bad word' |
07:00.28 | J4k3 | wtfbbq. |
07:01.12 | drmessano | ~cheney |
07:01.12 | jbot | I bet you thought I was going to say "diiiiiick" |
07:01.21 | drmessano | handled |
07:02.16 | J4k3 | haha |
07:02.53 | J4k3 | I need to stick this evdo card in my laptop and hack an antenna onto it. |
07:03.03 | *** join/#asterisk exerdigit (n=eric@adsl-68-72-220-3.dsl.akrnoh.ameritech.net) |
07:03.36 | *** join/#asterisk siya (n=djerk@194.60.207.239) |
07:03.50 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
07:06.15 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
07:06.29 | drmessano | All I have is tin foil, and I am using that for my hat |
07:16.24 | *** join/#asterisk lnx (n=lnx@183-83-66.ip.adsl.hu) |
07:16.27 | lnx | hi all |
07:24.01 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
07:25.16 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
07:36.49 | BeeBuu | how can i run multi command with asterisk -rx? |
07:37.05 | drmessano | what do you mean? |
07:37.26 | BeeBuu | Separate with ;? |
07:38.05 | BeeBuu | asterisk -rx "command 1;command 2..." |
07:38.12 | BeeBuu | can i do that? |
07:38.23 | drmessano | No idea |
07:38.31 | drmessano | What are you trying to do? |
07:38.44 | BeeBuu | auto run some command.. |
07:38.57 | BeeBuu | with console |
07:39.03 | drmessano | duh really |
07:39.11 | drmessano | I never would have guessed |
07:40.10 | drmessano | Just seems like a silly way to go about whatever it is you're trying to do |
07:40.43 | drmessano | But without knowing more than "trying to do stuff", i'll just leave it as "uh, dunno" |
07:41.38 | *** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
07:46.09 | tzafrir | BeeBuu, no. this doesn't work |
07:46.15 | lnx | hmh i have try to write out a return code of function send_text via AGI perl script. Text has been arrived. http://pastebin.com/d33413141 |
07:46.29 | lnx | but out file is empty |
07:46.34 | BeeBuu | tzafrir:how can i run multi command? |
07:46.57 | tzafrir | BeeBuu, only decent way I know: send commands directly to the unix-domain socket |
07:47.00 | lnx | BeeBuu: what kind of command wish to run? |
07:47.23 | BeeBuu | dial to someone and play some sounds. |
07:47.43 | BeeBuu | lnx: any thougths? |
07:48.24 | lnx | i have medi it via call file cause my server has not sound card :) |
07:48.36 | lnx | /medi/made |
07:48.48 | tzafrir | the base of the script I have: |
07:48.58 | tzafrir | while read line; do |
07:49.09 | tzafrir | echo -n "$line" |
07:49.19 | BeeBuu | tzafrir: what's that? |
07:49.20 | tzafrir | sleep 0.001 # separate between commands. |
07:49.20 | tzafrir | done |
07:49.28 | tzafrir | and that piped to: |
07:49.37 | lnx | BeeBuu: shell script :) |
07:49.45 | tzafrir | socat - /var/run/asterisk/asterisk.ctl |
07:50.02 | tzafrir | this is only for sending commands |
07:50.31 | BeeBuu | would you A-Z? |
07:50.46 | BeeBuu | would you teach me with A to Z? |
07:51.05 | drmessano | BeeBuu, you may want to take a look at the book |
07:51.14 | drmessano | It can answer a lot of these things for you |
07:51.22 | BeeBuu | where's the book? |
07:51.26 | BeeBuu | ~book? |
07:51.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
07:51.27 | drmessano | ~book |
07:51.27 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
07:51.47 | J4k3 | ~sike |
07:51.47 | jbot | [Sike!] The preceeding statement was intented to be humourous, and may have involved false statements used in a humorous manner, or an exaggeration of a real life situation with humourous intent, but in no way should be taken as pure fact or not considered questionable at best. |
07:52.03 | *** join/#asterisk PepOSX (n=angeldav@190.72.147.233) |
07:52.30 | lnx | te-asterisk-book.com is a nice one too |
07:53.03 | BeeBuu | lnx: where iss it? |
07:53.40 | lnx | on the Moon BeeBuu |
07:53.48 | BeeBuu | :-D |
07:53.56 | tzafrir | I believe recent nc also supports unix-domain sockets, so you may also use it. socat is packaged in e.g. debian |
07:54.21 | J4k3 | its in the air, dawg |
07:54.57 | tzafrir | ~j4k3 |
07:54.57 | jbot | i heard j4k3 is a dream killa |
07:55.01 | drmessano | J4k3: Did you ever play the game Castles? |
07:55.08 | BeeBuu | tzafrir: i conected to 5038 now |
07:55.27 | tzafrir | BeeBuu, that's another option. Requires setting up a user in advance |
07:55.38 | drmessano | In order to even start it, you needed to answer a question from the book... |
07:55.48 | drmessano | That would be a good arrangement for #asterisk lol |
07:55.50 | lnx | BeeBuu: u must read some docs @ www.voip-info.org |
07:56.04 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
07:56.27 | BeeBuu | lnx: i do,but there are so many... |
07:56.32 | lnx | lol |
07:56.35 | drmessano | lnx: Isn't it better just to ask every single minute question in here? |
07:57.00 | lnx | Isn't :) |
07:57.15 | drmessano | Surely you jest :) |
07:58.19 | lnx | BTW drmessano can u tell me please why return code of send_text does not present @ logfile http://pastebin.com/d33413141 :)) |
07:59.38 | drmessano | No, but I bet tzafrir can |
08:00.31 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
08:01.17 | Asterisk-nob | hi |
08:01.27 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
08:04.29 | Asterisk-nob | I've a question about the pound key mechanism, like press 1 followed by the pound key, so for this we simply create exten => 1#,1,dial(sip/ext) or it has some otherway to do? |
08:06.30 | BeeBuu | trafrir: i can |
08:06.43 | BeeBuu | not send command to AMI |
08:06.56 | tzafrir | lnx, have you checked that the open was successful? |
08:07.19 | BeeBuu | tzafrir: nc 127.0.0.1 5038 |
08:07.46 | BeeBuu | and send command,but doesn't work... |
08:07.47 | tzafrir | BeeBuu, sure. That's to the manager interface. |
08:08.16 | tzafrir | But the protocol there is slightly more complex. For starters you need a manager username and password |
08:08.19 | *** part/#asterisk MoreAllLess (n=jackjust@cpe-76-169-131-140.socal.res.rr.com) |
08:08.19 | tzafrir | and then: |
08:08.44 | tzafrir | something of the sort of: (/me drafts) |
08:08.50 | BeeBuu | must login? |
08:10.37 | *** join/#asterisk jeanmiii_i (n=besnard@LMontsouris-152-61-20-55.w80-13.abo.wanadoo.fr) |
08:10.46 | jeanmiii_i | Hi |
08:11.02 | *** join/#asterisk suahmed (n=Administ@69.88.13.17) |
08:12.17 | lnx | tzafrir: yes |
08:13.00 | lnx | tzafrir: outfile has been summoned :) |
08:13.24 | jeanmiii_i | One a call has been initialized, is it possible to change the media info (IP and port for rtp) using a reinvie or update ? |
08:14.56 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
08:15.10 | jeanmiii_i | I am trying to find out if it is possible to do a man-in-the-middle attack by updating the media of two phones by asking them to no longer send/receive rtp to/from eachother but to go through another IP/port that would then forward to the phones |
08:15.14 | lnx | tzafrir: i have try to give a variable to per script from extensions.conf , exten => 10,3,AGI(testscript.pl,${DIALSTATUS}) |
08:15.28 | jeanmiii_i | I am not sure if I am making myself very clear .... |
08:15.56 | lnx | tzafrir: then, no result |
08:17.01 | lnx | i have no more idea at all to get variables |
08:22.09 | obnauticus | what is caller id callback? |
08:22.13 | obnauticus | I don't understand the term |
08:30.55 | tzafrir | BeeBuu, http://pastebin.ca/925785 |
08:30.58 | JT | obnauticus: when you call asterisk, it detects your callerid as being one to use callback on, and disconnects the call, then it calls you back |
08:31.03 | tzafrir | If you feel like taking this further |
08:31.14 | tzafrir | you can replace the sed line with dos2unix |
08:31.45 | tzafrir | Sorry, busy elsewhere now |
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08:41.52 | *** part/#asterisk IgorG (n=IgorG@195.162.49.23) |
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09:02.17 | lnx | tzafrir: hmmh it seems the script makes the file but doesn't write into, |
09:02.40 | tzafrir | try a different print, then? |
09:02.56 | tzafrir | print L "whatever\n"; |
09:03.05 | tzafrir | perhaps you need the \n? |
09:03.12 | lnx | tzafrir: if i move open() down; the file does not become |
09:03.29 | lnx | only if open is in the 1st line |
09:03.41 | lnx | k testing : |
09:03.42 | lnx | ) |
09:04.54 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
09:05.04 | lnx | tzafrir: nothing, but -- AGI Script testscript.pl completed, returning 0 |
09:05.21 | lnx | it drives me crazy :S |
09:05.44 | BeeBuu | tzafrir:thanks |
09:07.04 | tzafrir | lnx, it returns 0 because this is what you returned |
09:07.50 | BeeBuu | tzaafrir: sleep 0.1 is enough? |
09:08.24 | lnx | tzafrir: it seems print to file by-pass |
09:09.33 | *** join/#asterisk CaRb0n^ (n=playa@203.81.237.252) |
09:09.48 | tzafrir | BeeBuu, I have no idea |
09:09.56 | tzafrir | Pure guess work |
09:10.14 | BeeBuu | what's sleep for? wait for result? |
09:10.16 | tzafrir | Try watching it in action (hint: tee) |
09:13.24 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
09:14.06 | [hC] | huh |
09:14.08 | [hC] | outcall is pretty nice |
09:14.11 | [hC] | for an outlook integrated plugin |
09:14.21 | lnx | tzafrir: $AGI->send_text('Hello World'); is succesful, what do you think why print does not work? |
09:14.39 | lnx | [hC]: nice :) |
09:15.58 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
09:20.24 | *** join/#asterisk akira2014 (n=chatzill@172.Red-88-8-198.dynamicIP.rima-tde.net) |
09:21.34 | akira2014 | hello |
09:21.37 | tzafrir | lnx, sorry, busy |
09:23.57 | *** join/#asterisk sergey (n=sergey@91.189.233.66) |
09:24.37 | jeanmiii_i | a phone is calling an extension on my aserisk PBX, I have written a script that sends an INVITE to the phone (using the callID of the estasblished call, so which should be considered as a reinvite) |
09:24.51 | jeanmiii_i | the phone is considering the INVITE as a new call |
09:25.10 | jeanmiii_i | instead of a reinvite (even though I am using the same from/to/call-id) |
09:25.21 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
09:25.37 | jeanmiii_i | do you any clue about what I might be doing wrong |
09:25.54 | agallo | there is a way to stop FORTINET to drop multiple outgoing INVITEs sent by asterisk outside? it believe its some kind of spoofing attempt; i mean how to delay invites each others? seems no settings about this in sip.conf |
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09:32.41 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
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09:33.23 | CaRb0n^ | <CaRb0n^> even if i dial 7777 |
09:33.23 | CaRb0n^ | <CaRb0n^> it plays the ivr but does not accept any didgits 9keys0 after that |
09:33.35 | *** join/#asterisk Datax (n=Datax@glou.nurvnet.org) |
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09:49.07 | lnx | what is the reason of that asterisk variable has no value? my $status = $AGI->get_variable('DIALSTATUS'); ? Local SIP dialed via call file, dial is successful... |
09:49.10 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
09:59.25 | Der-Tim | hi there |
10:00.14 | Der-Tim | i'm using a dialstring like this Dial(IAX2/username:password@host/${EXTEN}) but i get an error on the remote site: Registration Refused |
10:00.51 | tzafrir | Der-Tim, that is a dial, not a registration |
10:00.55 | tzafrir | Look elsewhere |
10:01.20 | tzafrir | Specifically, at the 'register =>' strings in iax.conf |
10:02.38 | Der-Tim | tzafrir: i don't use the register string, as an author of a german asterisk book wrote, that this is not needed... |
10:05.20 | *** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@octroi.enst-bretagne.fr) |
10:06.48 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
10:08.29 | BeeBuu | can i run any AGI program with CLI? |
10:09.57 | akira2014 | tzafrir did you remember me ? |
10:10.20 | akira2014 | i still in troubles with zaptel, after rebuilding all from sources |
10:10.40 | tzafrir | akira2014, what troubles? |
10:10.56 | tzafrir | random disconnections? |
10:11.12 | akira2014 | no, i cant pick up calls from pstn |
10:11.30 | tzafrir | it causes a hangup? |
10:11.31 | akira2014 | wen a call caomes nothing happens |
10:11.39 | akira2014 | no |
10:11.47 | *** join/#asterisk hijacked (n=argh@cerebus.clandestineresearch.com) |
10:12.34 | akira2014 | asterisk sees the zap channel, but nothing happens wen a call comes from landline |
10:12.50 | tzafrir | http://lists.digium.com/pipermail/asterisk-users/2008-March/206851.html |
10:14.45 | tzafrir | maybe downgrade to 1.4.7.1 (but in this case, be sure to rebuild asterisk) |
10:15.13 | tzafrir | hmm... "nothing happens" is something different from the problem described there |
10:15.44 | akira2014 | the problem related in this link is different |
10:16.22 | lnx | tzafrir: do you know why my ${DIALSTATUS} is empty? |
10:16.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
10:16.58 | akira2014 | my problem is exactly nothing appens, no messages from zaptel in log, no asterisk console messages.... nothing no where :( |
10:17.37 | tzafrir | lnx, sorry, wasn't following it |
10:17.59 | tzafrir | All I can suggest is to use extra tracing |
10:18.12 | akira2014 | strace ? |
10:18.29 | tzafrir | akira2014, any debug messages when you enable that debug? |
10:18.39 | akira2014 | no |
10:18.49 | akira2014 | verbose is 100 and debug is 100 |
10:21.13 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
10:21.22 | lnx | tzafrir: i have put verbose after answer exten => 10,2,Answer() exten => 10,3,AGI(verbose "MUUUU ${DIALSTATUS}",1) and the output is "MUUUU " -only |
10:22.34 | *** join/#asterisk g0mb0 (n=root@external.micom.mng.net) |
10:23.08 | Der-Tim | what can i do if i see such an error message: Auto-congesting call due to slow response ? |
10:23.38 | Der-Tim | i'm facing a problem with an static ip asterisk host without nat and a dynamic ip host behind nat... |
10:25.47 | *** join/#asterisk budol (i=bezpu@202.124.138.70) |
10:26.21 | jeanmiii_i | I have put a .call file in /var/spool/asterisk/outgoing to make an outgoing call but the file remains in the directory without any call being triggered (and no error message whatsoever) |
10:26.50 | jeanmiii_i | do I have to enable something somewhere in order to have the .call files triggering a call ? |
10:27.17 | sergee | ~seen danpwi |
10:27.30 | jbot | sergee: i haven't seen 'danpwi' |
10:28.41 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
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10:31.13 | budol | ~asterisk |
10:31.13 | jbot | methinks asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org |
10:31.39 | budol | ~zaptel |
10:31.39 | jbot | [zaptel] zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
10:31.59 | budol | ~bri |
10:31.59 | jbot | BRI == Basic Rate Interface, usually consisting of (1) 64kbps bearer (B) channel, and (1) 16kbps signalling (D) channel |
10:32.24 | budol | !pri |
10:32.29 | budol | ~pri |
10:32.29 | jbot | pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
10:37.15 | *** join/#asterisk xezz (n=asdasd@trust-it.gr) |
10:37.46 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
10:38.00 | JT | what the hell |
10:38.05 | JT | jbot: forget bri |
10:38.05 | jbot | JT: i forgot bri |
10:38.10 | xezz | hello, i've installed mod_ssl but http access still exists with https, any idea on how to disable http ? |
10:39.13 | JT | jbot: BRI is Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D) |
10:39.14 | jbot | JT: okay |
10:39.48 | g0mb0 | sergee? |
10:39.56 | g0mb0 | http://bugs.digium.com/view.php?id=11993 |
10:44.17 | Der-Tim | mmh, got a new problem by now... :-( |
10:44.35 | Der-Tim | Call rejected. no authority found... |
10:44.51 | Der-Tim | but there is an context "from-internal" on the remote system |
10:53.27 | beasty_ | is it possible to let sip user make an outgoing call by a IAX channel |
10:53.57 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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10:59.39 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
10:59.39 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform (asterisk.org) -=- Asterisk 1.6.0-beta4 (2008/02/21), Asterisk 1.4.18 (2008/02/07), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- FYI: no sharks in this channel |
11:00.38 | sergee | g0mb0: that was you, right? |
11:00.48 | g0mb0 | yes |
11:01.35 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-120ff3d37b7eee24) |
11:01.44 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
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11:09.38 | sergee | g0mb0: oops, i meant danpwi from http://bugs.digium.com/view.php?id=9299, 11993 is second in my list after 9299 |
11:09.57 | g0mb0 | ok |
11:10.01 | g0mb0 | gotta go, thanks |
11:10.44 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-89e6adf3782fc7f6) |
11:11.24 | nixguy | could anyone recomend a nice guy where users can change forwarding of phone numbers listen to messges etc? |
11:11.28 | nixguy | gui |
11:11.30 | nixguy | not guy :) |
11:15.14 | tzafrir | destar? |
11:15.20 | tzafrir | hmm... not listen |
11:22.21 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-99-78.d4.club-internet.fr) |
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11:23.36 | xezz | <PROTECTED> |
11:24.10 | angryuser | hello everybody, i have strange traffic amounts, i have installed pfsense with traffic shaper, and when i call it show's 88kb/s!? why so much? |
11:26.17 | jblack | angryuser: perhaps 40Kb a sec in, 40Kb a sec out. |
11:26.43 | jblack | Though offhand I can't think of a 40Kb codec. |
11:26.48 | angryuser | i got queue in and out, in 88 out 92 |
11:26.58 | jblack | There's 20Kb codecs, 56Kb codecs.... |
11:27.16 | beasty_ | anyone ever got this ? |
11:27.18 | beasty_ | WARNING[21381]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/jdecoste-081d4aa0 compatible with IAX2/329909001611-5 |
11:28.04 | angryuser | <jblack> maximum that i am able to use is G711 64kbit/s that is 8 ko |
11:28.17 | angryuser | <jblack> i dont have any special codec installed |
11:28.37 | angryuser | kinde strange |
11:28.39 | jblack | I didn't say those things. |
11:29.38 | jblack | Didn't you say you're using 88kb a sec, as in 88 kilobit (~ 10 kilobytes) |
11:30.13 | angryuser | <jblack> no it was 88 ko |
11:30.20 | jblack | I didn't say that either. |
11:30.23 | angryuser | like 10 times more |
11:30.52 | jblack | angryuser: "<jblack> no it was 88 ko" is quoting me as having said "no it was 88 ko" |
11:30.57 | JT | angryuser: 64kbit/s dows not include sip and rtp overhead |
11:30.59 | JT | what is ko? |
11:31.27 | angryuser | <JT> ah sorryn damn french it's kb ;) |
11:31.33 | JT | kB or kb? |
11:32.13 | angryuser | whatever it's like 512kbit/sec for one call (all traffic) |
11:32.50 | jblack | That doesn't sound like one call to me. Perhaps you have unrelated traffic in that count. |
11:33.15 | JT | kb = kilobits |
11:33.19 | JT | kB = kilobytes |
11:33.37 | angryuser | nope i am sure of that, i got like near 0 before call, and 88kB during |
11:33.58 | angryuser | maybe pfsense i lying ? |
11:36.49 | angryuser | i will play with codec's let's see |
11:38.16 | angryuser | codec's permitted alaw ulaw libc gsm, not a traffic consuming |
11:38.23 | JT | i think it's really kbit/s |
11:38.51 | angryuser | 49.30Kb/s -----status of queue |
11:38.52 | *** join/#asterisk atop (n=user@oaktyres.force9.co.uk) |
11:39.07 | angryuser | <JT> kbit's you think ? |
11:39.38 | JT | kb/s is kilbits per second |
11:39.50 | JT | kilobits |
11:40.09 | angryuser | i will ask someone to load a big file |
11:43.22 | angryuser | <JT> yes it was kbit/s! stupid me ;) |
11:45.02 | atop | If I recompile Asterisk with dont_optimize and malloc_debug flags set, can the resulting code be ran for a few days without problem, or will it run like a pig? |
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11:49.23 | *** join/#asterisk duckz (n=duckz@81.180.102.217) |
11:49.41 | beasty_ | anyone knows if i can let my SIP/ users call out using a IAX line ? |
11:51.25 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
11:52.57 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
11:57.57 | akira2014 | when a call comes from a zap channel and i make a sip/XXX why my phone rings forever even if the other side has hangup? thk's |
12:03.29 | tzafrir | akira2014, where are you at? |
12:03.47 | tzafrir | does your provider give any sort of disconnect supervision? |
12:04.01 | akira2014 | i'm in spain |
12:04.06 | tzafrir | if not, use busydetect :-( |
12:04.08 | akira2014 | my provider is telefonica |
12:04.33 | tzafrir | hmm... I'm not sure if it uses polarity |
12:05.04 | akira2014 | i will search in google about polarity on my profider |
12:05.43 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
12:07.49 | akira2014 | tzafrir, yes they use, i've forget to add it on zapata |
12:09.03 | tzafrir | <PROTECTED> |
12:09.52 | akira2014 | ok, now it works |
12:10.09 | akira2014 | thk's tzafrir |
12:13.41 | agallo | i've 16 account registered onto the same sip server, is there a way to give a little delay between each outgoing "REGISTER" attempts ? |
12:14.13 | jblack | not that I know of. |
12:14.42 | mosty | agallo, do you have a problem with the registrations? |
12:15.03 | agallo | mosty, yes but i think its related to some Fortinet Fortigate filtering out traffic |
12:15.14 | agallo | its seems does not like to spam many register at the same time |
12:15.41 | mosty | well that's easy to test, just disabled the filter temporarily |
12:15.52 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
12:16.37 | agallo | mosty, well its not a problem of me, its the DID provider that sucks! lol :) |
12:17.34 | mosty | contact the DID provider's tech support |
12:19.00 | agallo | mosty, they say its my router fault; i tested with other DID providers, other routers, other xDSL line and its not my fault; i'm getting 1 OK at the 1st REGISTER and no reply for the other 15. Their fortinet is filtering out stuff or there is some routing/spoofing problem with their packets |
12:20.58 | mosty | still sounds like you need to talk to them |
12:29.25 | jblack | Here's what you can do. |
12:29.36 | jblack | change the qualify time for them so that they're all different. |
12:30.12 | jblack | You'll still spam yourself out for the first round on all of them, but the later reregisters will catch back up on the 2nd round. |
12:32.03 | agallo | jblack, qualify handle the OPTIONS message not the register, afaik |
12:32.17 | jblack | Personally, I think the problem is not that they have abuse protection, but that the abuse protection is a little too stringent. |
12:32.23 | jblack | You got something better to try? |
12:33.07 | agallo | jblack, yes since they say its router fault i'm trying with 2 different routers and 2 different xDSL lines... also trying if i've some strange stuff in /proc/sys/dev/ ... but i never had problems with other providers.... |
12:33.27 | jblack | my thinking is it's possible that with qualify, when the timeout comes up, it'll notice that registration hasn't happen, and it'll reregister. |
12:34.03 | jblack | agallo: So your something better is to go with a different provider.. That's reasonable. |
12:35.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
12:37.29 | *** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
12:37.56 | agallo | jblack, unluckly i cannot switch it since it takes months to switch the telephone numbers too :-( |
12:38.21 | jblack | That doesn't sound like a better idea to me. |
12:38.29 | agallo | i'm very upset, this time its not asterisk fault, i cannot fix it myself :-P |
12:39.04 | jblack | I'm not upset. |
12:42.17 | *** join/#asterisk GBR_ (n=gbr@200.103.96.98) |
12:43.45 | coppice | "its out of my hands" sounds so much better than "I have to spend hours fixing stuff" |
12:44.31 | jblack | Didn't he say months, not hours? |
12:45.00 | coppice | that's even worse, isn't it? |
12:45.57 | coppice | I just hate work |
12:45.59 | agallo | its 2 month minimum to switch telephone number between DID providers :) |
12:46.08 | agallo | coppice, true, lets abolish it |
12:46.17 | jblack | It's as if you weren't being sarcastic. |
12:46.40 | Der-Tim | re |
12:46.51 | coppice | I'm not. I honest and sincere |
12:47.02 | jblack | agallo: So, try the qualify thing. Perhaps you'll be able to get all 15 registered over a period of 15-20 minutes. Then, you can just worry about not restarting * |
12:47.16 | Der-Tim | sorry for asking, but what does "no authority" mean? a remote asterisk is rejecting a call and i don't know why |
12:47.32 | jblack | That might buy you the time you need to port the numbers to a handful of providers. |
12:48.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
12:48.45 | jblack | der-tim: That sounds like a bad user/pass. I'd check the sip/iax regististrations to make sure that the servers are registering to each other with the right accounts and passes. |
12:49.11 | jblack | heh. Bill Richardson has a beard. |
12:49.12 | Der-Tim | jblack: there's nothing like registration... just username / secret in the iax config |
12:49.34 | agallo | jblack, tryed but there is no OPTIONS going out since they're not registered (indeed i never get a packet back from the server) |
12:49.47 | jblack | der-tim: Try registering (you can register with iax). See if that casts some light on the problem. |
12:50.43 | Der-Tim | jblack: well, i tried it with the user credentials in the dial string... |
12:50.57 | jblack | heh. |
12:51.21 | Der-Tim | jblack: if i'm using registration, am i in need of a [usernam] type=user section? |
12:52.13 | jblack | Not to do the registration itself. But if you're lacking an authentication stanza for a machine that's saying "you're not authenticated"... Well, to be polite.. shouldn't that be telling you something? |
12:54.48 | jblack | Presuming that machine B is trying to call machine A, and machine A is telling B that "youre not authenticated".. well, that comes down to two possibilities. |
12:55.21 | jblack | Either 1) Machine A doesn't have a guest account. or 2) Machine B isn't using a account that A recognizes |
12:56.14 | jblack | 2) Can be broken up into a half dozen possibilities... such as "Not trying to authenticate at all".. "Bad username." "Bad password".. "wrong hostname setup"... etc etc |
12:57.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
13:00.00 | Der-Tim | jblack: nothing of those possibilities... the username / password is the same on both machines... for sure... all i get are two different messages, which don't make sense atm |
13:00.13 | coppice | anyone know what happened to vovida.org? |
13:00.31 | Der-Tim | the first is "no authority" |
13:01.11 | Der-Tim | and the other is "no registration for peer 'username'" |
13:01.35 | Der-Tim | well, i think, i should take a look later... ;-) |
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13:21.55 | flujan | hi all, I need a tool to convert wav files to g729. |
13:22.32 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
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13:24.57 | beasty_ | is it hard to setup a musiconhold with mp3 ? |
13:27.52 | PepOSX | beasty_, nop |
13:27.57 | PepOSX | use playback |
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13:31.11 | jblack | Heh. Do trumpets go off when you get home? |
13:31.19 | stansmith | and the lights flash |
13:32.04 | jblack | "Children, daddy is home! Meet him at the front door and bow to him!" |
13:32.28 | stansmith | maybe some day... i still live at home with mommy |
13:32.54 | jblack | Oh, so those flashing lights are the alarm system.... |
13:33.20 | jblack | Didn't your parents teach you "gotocollege.getajob.moveout" ? |
13:33.32 | stansmith | done with college, at my job right now, ...... |
13:33.47 | jblack | 2/3 of the way there. |
13:34.09 | stansmith | i think my mom is scared to let her lil baby go, since im the youngest and what not |
13:34.23 | stansmith | chicks totally dig it by the way |
13:34.24 | stansmith | anyways.. |
13:34.36 | jblack | Nah. Mom is waiting for you to gtfo so that she can go on an ocean cruise. Trust me. |
13:34.47 | stansmith | haha |
13:34.52 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.131.143) |
13:35.00 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.71) |
13:35.09 | jblack | The flashing lights should have clued you into that. ;) |
13:35.10 | [TK]D-Fender | beasty_: Install "asterisk-addons" for MP3 suppotr |
13:35.35 | jblack | [TK]D-Fender: Hey, you want an invite to hulu? The site is a place to watch TV shows. |
13:35.48 | [TK]D-Fender | jblack: Thanks for the offer, but no need. |
13:35.59 | jblack | okey-dokey |
13:36.47 | PepOSX | LOL @ okey-dokey |
13:36.52 | *** join/#asterisk jarod14 (n=jarod14@ns1.viatelecom.com) |
13:37.54 | [TK]D-Fender | PepOSX: And Playback has nothing to do with MoH |
13:38.07 | beasty_ | [TK]D-Fender: now i get this |
13:38.09 | beasty_ | WARNING[30303]: res_musiconhold.c:947 local_ast_moh_start: No class: default |
13:38.32 | [TK]D-Fender | beasty_: Go set up your "musiconhold.conf" |
13:38.34 | PepOSX | MoH? |
13:38.37 | beasty_ | did that |
13:39.08 | [TK]D-Fender | beasty_: Well it seems to clearly think your class [default] does not exist |
13:39.55 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
13:40.41 | *** part/#asterisk suahmed (n=Administ@69.88.13.17) |
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13:41.35 | beasty_ | [TK]D-Fender: http://rafb.net/p/nrdcv071.html |
13:41.51 | real0ne | any one here use iaxcomm? |
13:43.40 | [TK]D-Fender | beasty_: "module reload res_musiconhold.so" |
13:45.43 | beasty_ | mm |
13:48.19 | ManxPower | I hate mornings |
13:50.55 | stansmith | someone has a case of the mon-daze! |
13:52.17 | ManxPower | I hate auto mechanics too. |
13:52.51 | *** join/#asterisk hi365 (n=hi365@77.125.78.34) |
13:53.25 | Havokmon | My last mechanics experience " You need a new computer" |
13:53.46 | Havokmon | Months later, afte rI put in a new computer, and Check Engine light still comes on I discover.... |
13:53.55 | agallo | jblack, *couch* after my chief phoned their chief now "magically" works :) |
13:53.58 | Havokmon | The o2 sensor they installed was unplugged :// |
13:54.09 | stansmith | lol my co-worker is snoring so loud |
13:59.01 | jblack | agallo: Yay for the old boy's network |
13:59.23 | ManxPower | I got a new clutch, I needed a new clutch anyway, but it did not fix the problem I was having ("bucking" when going some speeds) |
13:59.25 | *** join/#asterisk mattman99 (n=chatzill@ppp121-44-207-170.lns3.mel4.internode.on.net) |
14:02.00 | beasty_ | how do i convert a .mp3 to .wav for asterisk ? |
14:02.34 | stansmith | beasty_: google "convert mp3 to wav" |
14:02.36 | ManxPower | beasty_: however you would do it if you didn't have Asterisk. |
14:02.45 | mattman99 | linux or winows? |
14:02.51 | beasty_ | linux |
14:03.09 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
14:03.17 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
14:03.53 | [TK]D-Fender | beasty_: just use asterisk-addons |
14:04.11 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
14:04.12 | mattman99 | mpg123 will do it |
14:04.38 | [TK]D-Fender | beasty_: and I see your error in your MoH config. You put "mode=file", this should be "mode=files". |
14:05.49 | ManxPower | I thought it was mode=muffins |
14:06.44 | beasty_ | idd |
14:07.00 | beasty_ | [TK]D-Fender: it's my boss his 'ubuntu' apt-get install |
14:07.10 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
14:07.18 | beasty_ | and feisty doesn't have a asterisk-addon package since it's br0ken |
14:07.32 | ManxPower | beasty_: you don't want to install from packages anyway |
14:07.34 | ManxPower | not for Asterisk |
14:07.44 | ManxPower | not for Zaptel |
14:08.00 | [TK]D-Fender | beasty_: Then I guess this just reinforces that you should have simply compiled from source like the rest of us |
14:08.06 | jblack | beasty_: Theres good packages in hardy heron, which should be out soon. Excluding zaptel drivers, that is. |
14:08.40 | beasty_ | ManxPower: well i had it compiled |
14:08.41 | jblack | I use packages, and understand the reasons why you may want to use them too. |
14:08.58 | beasty_ | my boss removed it ... and installed from packages |
14:09.02 | stansmith | zaptel is easy to compile, the only dependency is glibc |
14:09.45 | jblack | zaptel is also dependant upon the kernel headers, for which there have recently been some impedance mismatches. |
14:09.57 | stansmith | true |
14:10.37 | ManxPower | beasty_: Asterisk is one of the VERY few applications that I compile from source |
14:10.53 | mattman99 | i agree manx |
14:10.58 | stansmith | young chris sounds just like jay-z its disgusting |
14:11.00 | beasty_ | true |
14:11.05 | beasty_ | can you tell that to my boss |
14:11.07 | ManxPower | The others are SpamAssassin, ClamAV, and a commercial web interface to IMAP |
14:11.15 | beasty_ | idd |
14:11.19 | ManxPower | Notice all of those change fast |
14:11.49 | jblack | I'm on your boss'es side. I don't think production work should depend upon the recently buildable crack. |
14:12.46 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
14:12.49 | ManxPower | jblack: Oh, I don't CARE if you use package or build from source. What I care about is people coming here expecting us to help fix something that is broken or changed in their package .vs. Asterisk |
14:13.36 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
14:13.37 | *** mode/#asterisk [+o lmadsen] by ChanServ |
14:13.50 | ManxPower | "Asterisk isn't reading the files in /etc/asterisk. I'm using package X from distro Y." "Where is Asterisk compiled to look for it's config files?" "I have no idea, I didn't build it." |
14:13.52 | jblack | manxpower: sure, I can see how that's inconvienant. |
14:14.13 | ManxPower | jblack: it also takes away support resources from people that need them. |
14:14.45 | stansmith | i mean, the book outlines the dependencies, so there really isnt an excuse for not being able to compile it from source |
14:15.12 | ManxPower | *grumble* I supposed I should take the mechanic that worked on my truck on a "test drive". Need to make sure I have the buckets and cement first. |
14:15.20 | jblack | Heh. Production systems are where one can derive support income. Supporting CVS from 45 minutes ago doesn't feed your kids. |
14:15.36 | ManxPower | jblack: I would never run CVS. |
14:15.47 | ManxPower | I want a stable, working system, that is the same across all servers. |
14:15.55 | *** join/#asterisk fskrotzki_ (n=fskrotzk@host198.textwise.com) |
14:16.09 | ManxPower | CVS is French for "have more time than money", you know. |
14:18.17 | jblack | whatever works for you |
14:18.33 | stansmith | svn > cvs ? |
14:18.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
14:19.40 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
14:19.45 | Katty | o hai! i upgraded your pbx. |
14:19.50 | jblack | For some value of "greater than" |
14:20.00 | Katty | x is greater than or equal to 4. |
14:20.10 | Katty | and y=5 |
14:20.13 | Katty | plot it. |
14:20.45 | Katty | that'd be like half a parabola. |
14:22.54 | [TK]D-Fender | Katty: Not without an exponent, otherwise you'll only get a straight line.... |
14:23.03 | stansmith | LOL |
14:23.06 | stansmith | 0wn3d |
14:23.08 | [TK]D-Fender | Katty: (Mew) |
14:23.13 | Katty | you are SO right. |
14:23.33 | Katty | way to be on your toes this morning, mister mathematical ballet performer. |
14:23.49 | jblack | No he's not. |
14:24.00 | Katty | oh sure he is. |
14:24.11 | [TK]D-Fender | Its like they say, there's only three kinds of people out there ; those that can do math, and those that can't. |
14:24.11 | jblack | He's on the balls of his feet, because there's more ways to give a curve than an exponent. |
14:24.58 | Katty | anymore talk like that on a monday morning, and i'll make sure your exponentially removed. |
14:25.13 | Katty | have pity on my poor sleepy brain! |
14:25.18 | jblack | Your contribution here is logarithmic, at best. :) |
14:25.50 | tzanger | haha no cable yet Katty? |
14:25.58 | Katty | jblack: i resent that. |
14:26.00 | jblack | oh, that came out mean. I'm sorry. It was meant to be merely funny. |
14:26.31 | stansmith | question - is there some sort of switch statement that can be used in extensions.conf or must one resort to a lot of GotoIf()'s? |
14:26.32 | Katty | jblack: i'm, by far, more than just some little f(x) = c log x!! |
14:26.43 | Katty | tzanger: no :/ |
14:26.59 | Katty | jblack: and you're wrong. i'm a female. i'm in no way a constant :P |
14:27.14 | tzanger | amen, sister! |
14:27.43 | jblack | So, you're a high order polynomial? Up down, up down, everywhere a tight curve? |
14:27.52 | jblack | uh. gah. sorry again |
14:28.04 | Katty | no references this early in the morning. |
14:28.10 | jblack | I can't stay out of trouble this morning |
14:28.33 | tzanger | nothing wrong with tight curves |
14:28.39 | jblack | I didn't mean to imply innuendo. I'm still on my first pot of coffee. I swear |
14:28.40 | stansmith | O!! |
14:28.40 | tzanger | even girls like 'em |
14:28.49 | Katty | i think i'm going to have to cite the story of the little equation that could |
14:29.02 | Katty | the little equation that ate EVERYTHING |
14:29.17 | Katty | and that little equation COULD EAT YOU! |
14:29.21 | jblack | I was referring to the old, sexist stereotypical rollercoaster of female emotions |
14:29.26 | jblack | Yes Ma'am |
14:29.33 | tzanger | there is just way too much innuendo in this channel this morning. I love it |
14:29.40 | jblack | I didn't mean it@! |
14:29.43 | tzanger | there isn't anything old or sexist about that |
14:29.44 | Katty | jblack: yeah, but there are reasons for that. |
14:29.52 | *** join/#asterisk BrokenNoze (n=root@host81-149-254-218.in-addr.btopenworld.com) |
14:29.52 | tzanger | it's a given fact |
14:29.52 | Katty | jblack: and don't tell me males don't have mood swings |
14:29.52 | jblack | Yes ma'am |
14:29.55 | Katty | jblack: cause i know better :P |
14:30.01 | tzanger | oh we do |
14:30.03 | tzanger | for sure |
14:30.05 | Katty | jblack: i live with a very moody man right now |
14:30.17 | Katty | poor guy has to do a tower climb today :< |
14:30.19 | jblack | Sure... they go from quiet... to angry.. to quiet... to brooding.. back to quiet. |
14:30.21 | *** join/#asterisk docelm0 (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
14:30.21 | BrokenNoze | Hi All, anyone know if there's a way to bridge two active channels? |
14:30.29 | tzanger | jblack: haha |
14:30.39 | Katty | jblack: cave syndrome |
14:30.44 | jblack | Ugh. |
14:31.03 | [TK]D-Fender | I am NOT a passive-agressive powder-keg just waiting to explode, and I'll *KILL* anyone who even looks at me funny at the suggestion of it! |
14:31.33 | ManxPower | BrokenNoze: almost every call in asterisk is bridged between two channels. Perhaps a less generic question with a bit of background would be more productive? |
14:31.44 | Katty | morning Manx (= |
14:31.45 | jblack | Here, it's just me and a 14 year old girl. So, don't you go talking to me about female emotions. I'm ON THE FRONT LINES! |
14:31.46 | yang | I am wondering about video phone that use h.263 & h.264 does this have a working support in asterisk ? |
14:31.57 | Katty | jblack: i am /so/ incredibly sorry. |
14:32.08 | BrokenNoze | ManxPower: OK, then attended tranfer via the Manager API... |
14:32.13 | Katty | jblack: you have my most sincere condolences. |
14:32.14 | jblack | Not as sorry as I am. :) |
14:32.25 | ManxPower | BrokenNoze: now you at least are asking the right questions |
14:32.39 | BrokenNoze | just that question never seems to get me anywhere so thought i'd change it a little ;-) |
14:32.41 | Katty | jblack: i can't even imagine how you can put up with a 14 year old.. |
14:32.50 | tzanger | jblack: mine start young. I've got a 4yo and an 8yo lil bundle of emotion, and a 33 year old leader in that territory |
14:33.10 | [TK]D-Fender | jblack: Just because you see the fall-out with a front-row seat doesn't mean you necessarily have a clue whats really going on :) |
14:33.16 | tzanger | [TK]D-Fender: hahahahaha |
14:33.26 | BrokenNoze | ManxPower: if you can answer that question with a positive I'll owe you bigtime! |
14:33.30 | jblack | I feel like a elephant that can't turn around fast enough to deal with a pesky hornet. |
14:33.36 | [TK]D-Fender | jblack: So grab some popcand enjoy the show! |
14:33.52 | Katty | jblack: yeah, and if you tried... |
14:34.01 | Katty | jblack: it'd be hell on earth, with a few exponents. |
14:34.07 | [TK]D-Fender | s/popcand/popcorn and/ |
14:34.23 | Katty | i remember being 14, just barely. |
14:34.29 | *** join/#asterisk kraypius (i=user@c-67-175-202-53.hsd1.il.comcast.net) |
14:34.50 | jblack | I've found things work best if I just trudge through life, and ignore the mild stings. She doesn't mean them anyways. |
14:35.02 | Katty | jblack: no, no she doesn't... |
14:35.07 | ManxPower | send her off to boarding school. problem solved |
14:35.11 | kraypius | <PROTECTED> |
14:35.16 | Katty | ManxPower: horrors! |
14:35.20 | Katty | ManxPower: you may never have children. |
14:35.25 | ManxPower | And try to help others avoid the mistake of having children! |
14:35.26 | jblack | Every once in awhile, I'll knock down her metaphorical hornet's nest to remind her that while I lumber, I can affect things she likes. |
14:35.42 | Havokmon | I have a 14 year old girl.. Brothers in arms :) |
14:35.50 | ManxPower | Katty: My children have four paws. |
14:35.55 | Katty | ManxPower: cheers. |
14:35.58 | BrokenNoze | kraypius: X-lite |
14:36.03 | jblack | You sexual deviant you. |
14:36.11 | Katty | what did we say about references! |
14:36.21 | x86 | :P |
14:36.26 | stansmith | ? |
14:36.28 | ManxPower | jblack: you have no idea. |
14:36.33 | jblack | ewwwww. |
14:36.40 | *** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
14:36.50 | ManxPower | Anyway, I'm off to torture an auto mechanic. Wish me luck. |
14:37.08 | BrokenNoze | OK, new question. How do i bridge two active calls in asterisk? |
14:37.16 | jblack | We may not hear from him for days if he's on the losing end of that battle. ;) |
14:37.28 | jblack | brokennoze: There's a bridge app |
14:37.39 | [TK]D-Fender | BrokenNoze: There is a redirect AMI call you can use. |
14:37.47 | jblack | Except it's called something else |
14:37.57 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
14:38.00 | BrokenNoze | It doesn't bridge two ACTIVE channels though does it? |
14:38.41 | BrokenNoze | i use redirect to redirect one active channel to another point in the dial plan, not to another actual open channel |
14:38.42 | [TK]D-Fender | BrokenNoze: Yes |
14:38.54 | jblack | Perhaps chanspy |
14:39.02 | x86 | BrokenNoze: chanspy will do it |
14:39.12 | x86 | or you could park/pickup |
14:39.34 | BrokenNoze | I'm basically directing a call to a hold extension |
14:39.52 | jblack | Oh, Try the transfer application |
14:40.10 | BrokenNoze | transfer is blind though isn't it? |
14:41.19 | anonymouz666 | putnopvut: do you care if I ask you directly an app_queue user question? |
14:41.21 | jblack | well, there's the TRANSFERSTATUS variable.... |
14:41.23 | *** part/#asterisk PepOSX (n=angeldav@190.72.147.233) |
14:41.34 | jblack | With typical j n+101 behaviour |
14:41.38 | BrokenNoze | trying to simulate an attended transfer by holding one channel, connecting another to the remote party, dropping the first party and connecting the held channel to the dialled endpoint |
14:41.49 | Katty | anonymouz666: morning (= |
14:41.53 | anonymouz666 | hello kaldemar |
14:41.55 | anonymouz666 | ops |
14:41.56 | nixguy | is there a variable for the incoming number? |
14:41.57 | anonymouz666 | Katty! |
14:41.58 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
14:42.07 | jblack | BrokenNoze: Hmm. Did you check to see if your phone already supports it with flash? |
14:42.17 | nixguy | $EXTEN is outgoing, so what is it for incoming? |
14:42.25 | jblack | flash - new number - "Hello there. have a call for ya" - hang up |
14:42.27 | putnopvut | anonymouz666: What's your question? |
14:42.34 | stansmith | omg omg omg |
14:42.39 | BrokenNoze | jblack: no good, i need to do it via Manager as we support multiple SIP hardphones |
14:43.06 | anonymouz666 | putnopvut: Once the caller is waiting in a queue, is it possible to reinject the PRIO? |
14:43.18 | anonymouz666 | before I know it is possible. |
14:43.23 | [TK]D-Fender | BrokenNoze: Use "Redirect" and point to a dialplan context that will prompt the call like you described |
14:43.27 | anonymouz666 | just using QUEUE_PRIO |
14:43.37 | putnopvut | anonymouz666: what do you mean by "reinject?" |
14:43.41 | *** join/#asterisk lirakis (i=lirakis@pr0tected.us) |
14:43.42 | anonymouz666 | change the prio. |
14:44.01 | anonymouz666 | it is a important customer, he should be the first |
14:44.19 | anonymouz666 | but he is already waiting... |
14:44.25 | BrokenNoze | nixguy: CALLERID(num) or CALLERID(name) |
14:44.29 | b11d | . |
14:44.37 | putnopvut | anonymouz666: Well you can set the QUEUE_PRIO variable before he joins the queue, but once he's waiting you can't change his priority. |
14:44.43 | stansmith | is there a more "elegant" way to capture user key press for up to 3 digits delimited by "#" then this? ---> http://www.pastebin.ca/926030 |
14:44.54 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
14:45.19 | [TK]D-Fender | stansmith: "core show application read" |
14:45.34 | putnopvut | anonymouz666: there are hackish ways you can go about moving a person up in the queue... |
14:45.35 | [TK]D-Fender | stansmith: Yes, this should have been 1 line of dialplan. |
14:45.40 | BrokenNoze | Fender: that works for the first two parts, but when i actually have the two active calls in progress... |
14:45.41 | nixguy | BrokenNoze: k thnx |
14:45.43 | stansmith | @#$@! |
14:46.01 | Katty | nixguy: are you australian? |
14:46.18 | putnopvut | anonymouz666: but no way of just manipulating the QUEUE_PRIO variable to move a caller already in the queue up. |
14:46.26 | [TK]D-Fender | BrokenNoze: What are you going to do with your destination's CURRENT call? You're saying "Person B, here's a call NOW, deal with it" |
14:46.28 | anonymouz666 | putnopvut: ChannelRedirect on the Zap channel and then make hit the QUEUE_PRIO before join the queue |
14:46.44 | putnopvut | anonymouz666: yes something like that. |
14:46.54 | anonymouz666 | putnopvut: you think that could be a interesting feature? |
14:47.21 | putnopvut | anonymouz666: I don't know really. It seems like something you should take care of prior to joining the queue to me. |
14:47.29 | *** join/#asterisk docelm0 (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net) |
14:47.32 | Katty | docelm0: mew. |
14:48.05 | anonymouz666 | alright, thanks for your time. |
14:48.11 | BrokenNoze | Fender: no Person A is calling person B to tell him that person C is holding. when person A hangs up, person B and person C need to be connected |
14:48.12 | putnopvut | anonymouz666: no problem. |
14:48.43 | [TK]D-Fender | stansmith: and your PB showed I could punch in any multiple of 3-digits I wanted. |
14:48.49 | x86 | BrokenNoze: that sounds like an attended transfer to me |
14:48.51 | BrokenNoze | if person A can't get hold of person B because he's already on a call, he'll just grab person C back and apologies for not being able to redirect |
14:48.55 | *** join/#asterisk twisted (n=root@pdpc/supporter/active/twisted) |
14:48.56 | *** mode/#asterisk [+o twisted] by ChanServ |
14:48.59 | Katty | hi twisted! |
14:49.03 | BrokenNoze | x86: that is EXACTLY what it is |
14:49.09 | x86 | BrokenNoze: yeah that's easy :) |
14:49.18 | x86 | BrokenNoze: you using analog channels? |
14:49.43 | BrokenNoze | no |
14:49.52 | BrokenNoze | SIP |
14:50.30 | BrokenNoze | x86: lay it on me! I've been after it for months!! |
14:50.50 | stansmith | [TK]D-Fender: thats the idea, a caller can have an account with a suffix up to 3 digits |
14:50.55 | *** join/#asterisk PepOSX (n=angeldav@201.243.76.220) |
14:51.16 | [TK]D-Fender | stansmith: But your context allows me to enter 12 if I wanted. |
14:51.29 | stansmith | yea thats fine |
14:51.34 | [TK]D-Fender | stansmith: Yuo only needed 1 single line of dialplan with "Read" to do this properly. |
14:51.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
14:51.58 | *** mode/#asterisk [+o russellb] by ChanServ |
14:52.43 | Katty | hi russel! |
14:52.54 | stansmith | [TK]D-Fender: actually, its somewhat tricky with Swift(), but i believe this is my work around in case anyone is interested ---> http://www.pastebin.ca/926040 |
14:53.16 | stansmith | i havent been able to determine whether ${SWIFT_DTMF} is persistant or only holds the last digit |
14:53.33 | stansmith | *hopefully* it only holds the last digit, ill have to check the source |
14:54.22 | BrokenNoze | x86: so what do i do? |
14:54.26 | *** join/#asterisk RoyK_ (n=roy@fw.fortel.no) |
14:54.37 | Katty | roy (= |
14:55.18 | [TK]D-Fender | stansmith: It should only take 1 line fo dialplan for your read.. thats a lot of filler for nothing... |
14:56.05 | stansmith | [TK]D-Fender: yea but if the caller presses something while Swift() is executing, swift will try and go to that extension (via swift.conf --> "goto_exten=yes"), so i need to catch that key press |
14:56.31 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-164-21.dsl.teksavvy.com) |
14:56.31 | stansmith | if your talking about the noop(), i put that in cause i didnt want to renumber everything, n priority ftl |
14:56.36 | lnx | i have put some exten => 10,7,NoOp(exten_Dialstatus ${DIALSTATUS}) in my dialplan and ${DIALSTATUS} is always ampty, but the call is succesful. Can u tell me why? please |
14:56.54 | JenniferAkemi | good morning everyone |
14:57.01 | [TK]D-Fender | stansmith: Ah, it aborts on DTMF? Ok, you will need an IVR approach then |
14:57.26 | russellb | Katty: greetings :) |
14:57.30 | [TK]D-Fender | lnx: pastebin your entire context and a sample call's CLI output at verbose 10 |
14:57.32 | [TK]D-Fender | ~pb |
14:57.33 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
14:57.33 | stansmith | [TK]D-Fender: yea, swift will go to the extension if it exists, but i think this is a work around, i need to test it |
14:57.35 | [TK]D-Fender | ^^^^^^^^^^^ |
14:57.41 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
14:57.54 | [TK]D-Fender | stansmith: You should read "_X", not "_XXX", and collect each digit. |
14:58.08 | [TK]D-Fender | stansmith: And keep a length count so they don't enter too mcuh. |
14:58.46 | *** join/#asterisk LjL (n=ljl@ubuntu/member/ljl) |
15:00.37 | Katty | hi ccvp (= |
15:01.07 | ccvp | hello Angela |
15:02.11 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
15:03.49 | lnx | [TK]D-Fender: http://pastebin.com/m7cf4d2ab |
15:05.04 | *** join/#asterisk CVirus (n=GoD@196.205.192.185) |
15:05.06 | [TK]D-Fender | lnx: You never call "Dial" anywhere is there, of course ${DIALSTATUS} is going to be empty. |
15:05.32 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
15:06.43 | lnx | [TK]D-Fender: asterisk does not use Dial() via outgoing/ ? |
15:06.50 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584628.dsl.bell.ca) |
15:07.47 | stansmith | that is tits, i think my solution works |
15:08.22 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-54-40.dsl.tul2ok.sbcglobal.net) |
15:08.23 | stansmith | i think im gonna submit that to the asterisk cookbook wiki |
15:08.26 | *** join/#asterisk acron17 (n=joe@p5089F6C0.dip.t-dialin.net) |
15:08.34 | acron17 | hi there |
15:08.42 | stansmith | ~hi acron17 |
15:08.43 | jbot | Many greetings, acron17, most strange traveller, to this IRCdom of plenty. |
15:09.14 | acron17 | i know that asterisk can bi configured to use info-messages for dtmf |
15:09.26 | acron17 | that works great with me voip-app |
15:09.31 | stansmith | acron17: IVR? |
15:10.16 | Katty | to play warcraft or not play warcraft... |
15:10.20 | acron17 | just how the dtmf tones are transported |
15:10.26 | cmantito | silly Katty. the answer is ALWAYS play. |
15:10.28 | cmantito | :P |
15:10.33 | stansmith | cs > wow |
15:10.37 | cmantito | nahh |
15:10.42 | ccvp | acron17 |
15:10.44 | ccvp | s/bi/be |
15:10.46 | ccvp | imo :) |
15:10.47 | stansmith | lol |
15:10.48 | Katty | hrmm |
15:10.55 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
15:10.55 | Katty | to play the hunter, the mage, or the pally.. hrmm |
15:11.00 | stansmith | acron17: are you asking a question or stating a fact? |
15:11.01 | acron17 | is there way a voip-app may tell asterisk which kind of DMTF mode to use? |
15:11.01 | ccvp | can u get level 80's |
15:11.03 | ccvp | in wow yet? |
15:11.08 | cmantito | not yet |
15:11.08 | Katty | not yet |
15:11.08 | cmantito | soon |
15:11.18 | ccvp | i wonder how hard it'll be |
15:11.26 | ccvp | big experience curve, ie: 2-3months RL time, or PL in few days heh |
15:11.27 | cmantito | Katty: what realm/region/faction are you, out of curiosity? |
15:11.39 | Katty | cmantito: horde, llane |
15:11.44 | ccvp | i never got into WoW |
15:11.48 | ccvp | im waiting for blizzards next free game |
15:11.49 | ccvp | Diablo 3 |
15:11.49 | cmantito | Katty: For the horde! |
15:11.55 | Katty | cmantito: (= |
15:12.09 | ccvp | Diablo 2 > WoW |
15:12.11 | Katty | i'm not that serious... |
15:12.20 | Katty | but i have a 70 mage and a 70 hunter, and my pally is uh... 13 |
15:12.24 | stansmith | LOL |
15:12.26 | cmantito | hehe |
15:12.31 | cmantito | I have a lvl 50 shammy |
15:12.34 | ccvp | i played wow once like 1.5 years ago |
15:12.35 | drfreeze | Hello |
15:12.39 | ccvp | and had some class that had a blue ghost |
15:12.42 | ccvp | as my pet, it was a tank |
15:12.44 | [TK]D-Fender | lnx: that var only gets set when App_dial is called |
15:12.54 | cmantito | I was serious for a little while, but I've found something else that holds my fantasy interests |
15:12.58 | cmantito | more dorky than wow |
15:13.02 | drfreeze | I've had some problems this morning with the not being able to hear incoming calls |
15:13.05 | Katty | a warlock gets a blue 'ghost' tank |
15:13.16 | [TK]D-Fender | lnx: and the fact you are IN the dialplan at all means the call is in progress and has ALREADY been answered |
15:13.25 | drfreeze | One just occurred where I could hear the incoming call, then they went silent |
15:13.40 | cmantito | I've taken up LARPing :3 |
15:13.56 | acron17 | is there way a voip-app may tell asterisk which kind of DMTF mode to use? |
15:14.03 | drfreeze | On a system that is working normally, what could be a potential problem that is causing this? |
15:14.04 | lnx | [TK]D-Fender: i understand thankl you |
15:14.08 | joshaidan | Is there an * postgresql app that lets your run queries in extension.conf similar to MySQL() |
15:14.20 | stansmith | joshaidan: yes |
15:14.27 | stansmith | res_postgresql i think |
15:14.28 | [TK]D-Fender | joshaidan: use ODBC |
15:14.32 | stansmith | ^ |
15:15.21 | stansmith | O'REILLY has a book called "cookbooks in a nutshell"? |
15:15.21 | acron17 | no answers or no hints? |
15:15.42 | *** join/#asterisk pithen (n=pithen@mail.graphlogic.com) |
15:15.45 | tzanger | stansmith: heh |
15:16.24 | angryuser | offtopic someone with good knowledge of openvpn here? what is this ? write UDPv4: Operation not permitted (code=1) (openvpn log) unable establish connection |
15:16.44 | Katty | sounds firewall related |
15:16.51 | stansmith | sounds permission related |
15:16.58 | Katty | could be that too |
15:17.09 | pithen | Hey all.. I have an * system with a Digium (4xFXO) card.. for the most part everything works fine, but occassionally the system will not answer calls from PSTN, and when dialing an outgoing call I just get loud static.. Reboot the system and all is fine again. Any thoughts? Card going bad? |
15:17.13 | stansmith | youll get a message similiar to that when you try to do something root as a non-root user |
15:17.18 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
15:17.30 | Katty | pithen: i've had plenty of those problems |
15:17.32 | stansmith | pithen: how do u dial out with 4 fxo? |
15:17.39 | Katty | pithen: i think it's just the cards really |
15:17.46 | JenniferAkemi | larping seems like it would be better exercise than wow at least |
15:17.52 | pithen | stansmith, did i swtich my terminology..fxs? |
15:17.57 | ThatKidKel | When calling Dial(SIP/xxxx) is it not possible to get the SIP failure reason? Other than CONGESTION or BUSY? Ie. 404 |
15:18.01 | stansmith | pithen: i get confused too haha |
15:18.08 | acron17 | russellb: you said in asterisk-dev, that there is a possibility to do that,,,, |
15:18.12 | Katty | pithen: we had to reboot our server from random hiccups once a month or so |
15:18.20 | pithen | Katty, this is almost every day nowe |
15:18.29 | Katty | hrmm. that is a bit excessive. |
15:18.31 | lnx | [TK]D-Fender: how can i call an agi script without a callfile please |
15:18.38 | Katty | do you have another card you can throw in there to test? |
15:18.48 | drfreeze | Anyone know why this doesn't reboot a polycom phone: sip notify polycom-reboot ext |
15:18.52 | stansmith | lnx: exten => s,n,AGI([script name]) |
15:18.59 | [TK]D-Fender | lnx: What do you need your script to do anyways? |
15:19.21 | pithen | Katty, yeah the boss is ready to toss out the whole system.. I don't have anything else on hand, but was hoping there was some way to test the card before I go out and blow another $x00 |
15:19.27 | [TK]D-Fender | drfreeze: because you need to set your provisioning configs to accept the notice. |
15:19.29 | stansmith | lnx: place the script in /var/lib/asterisk/agi-bin/ (default location) |
15:19.47 | Katty | pithen: i can sympathize with that. for awhile, the boss was ready to toss ours too |
15:19.51 | stansmith | chmod +x ! |
15:20.09 | stansmith | chmod +x digiumcard should fix it |
15:20.10 | lnx | [TK]D-Fender: call a number and analise ${DIALSTATUS} automatically |
15:20.14 | Katty | pithen: i'm sure there's a way to enable debug for the card, but i couldn't tell you what to really look for. |
15:20.19 | lnx | stansmith: ty |
15:20.33 | pithen | Katty, ill look into that, thank you |
15:20.42 | drfreeze | [TK]D-Fender: ahh. How do I do that? |
15:20.46 | [TK]D-Fender | lnx: Think about what kind of channel you are placing your call against. |
15:20.56 | [TK]D-Fender | drfreeze: Go check your admin guide. |
15:21.16 | stansmith | lmadsen: is it still possible to create an account for asteriskcookbook.com ? |
15:21.31 | lnx | stansmith: exten => s,n,AGI([script name]) when invokes this line? |
15:21.33 | lmadsen | stansmith: yes, you have to follow the instructions on the site though |
15:21.41 | lmadsen | stansmith: which basically means you have to email the guy who will set it up |
15:21.44 | lnx | [TK]D-Fender: maybe i must create one |
15:22.03 | stansmith | ah...create account wasnt creating account |
15:22.18 | stansmith | lnx: i dont understand your question |
15:22.26 | [TK]D-Fender | lnx: Just look at the channel you are originating and think about what kind of channel would allow you to deal with it not being answered.... |
15:23.00 | pithen | Anybody have experience with Rhino? Quality hardware? Im hoping to get a new system altogether with a 24 channel fxs bank and a fxo card |
15:23.18 | Katty | pithen: we had a rhino server once. |
15:23.25 | lnx | [TK]D-Fender: i have to call *94 example |
15:23.30 | [TK]D-Fender | pithen: Not worth it. |
15:23.33 | pithen | Katty, i notice thats past tense :) |
15:23.39 | *** join/#asterisk Fusoya (i=quality@togi.homeunix.org) |
15:23.44 | Katty | pithen: tho, we were more curious about the Pretty Software |
15:23.49 | [TK]D-Fender | pithen: Just get a SIP gateway |
15:23.55 | Katty | pithen: eh, i guess it worked okay with rhino equipment |
15:24.04 | Katty | pithen: but, for the price, ... |
15:24.08 | Katty | pithen: not really worth it |
15:24.11 | [TK]D-Fender | lnx: What is this talk about dialing "*94" have to do with your call-file> |
15:24.13 | [TK]D-Fender | ? |
15:24.26 | coppice | aren't rhinos an endangered species? :-\ |
15:24.36 | Katty | coppice: i don't think so. |
15:24.38 | pithen | my current setup involves about a dozen Sipura units..its hell to maintain |
15:24.46 | [TK]D-Fender | pithen: as CB's go, Rhino is decent, but its not a solution I advise unless you need it or ahve the extra T1 ports to spare. Even then.... |
15:24.58 | coppice | [checks with the WWF] wrong! |
15:25.08 | Katty | not the first time ;) |
15:25.08 | [TK]D-Fender | pithen: use a bigger SIP gateway like AudioCodes MP-124 or Mediatrix 1124 |
15:25.24 | pithen | [TK]D-Fender, will check those out |
15:26.09 | coppice | choose a big name like mediatrix. they aren't any less buggy, but everyone has worked arounf their bugs so they are compatible :-) |
15:26.11 | lnx | [TK]D-Fender: it was a testing :) now i know that i must use Dial() |
15:27.21 | Fusoya | Is there a way to fix the "ring requested on channel already in use" zaptel bug that wouldn't require me to restart the server? |
15:27.56 | ccvp | in topic, what exactly is switchvox |
15:28.04 | ccvp | an expensive turkey provider for businsesses? |
15:28.05 | ccvp | seems pricey |
15:28.07 | Fusoya | (ast1.2.13 zap1.2.11) |
15:28.09 | ccvp | turn key |
15:28.35 | stansmith | bruce@oreilly.com no worky? |
15:29.19 | lnx | [TK]D-Fender: my goal is : make timing calls and analise ${DIALSTATUS} of that calls. I hope it is acchieve in enxtensions.conf |
15:30.11 | lnx | [TK]D-Fender: and *94 is auto answering :) |
15:32.49 | lnx | [TK]D-Fender: it would be a testing mechanism to examine calls are going well |
15:33.12 | Havokmon | Regarding configs, is an upgrade from * 1.2 to 1.4 pretty much seamless? I've seen some differences in some configs - like parking.... |
15:33.32 | Fusoya | Nobody knows a way to un-hang a zap channel without a full restart in Asterisk 1.2? |
15:34.32 | [TK]D-Fender | Havokmon: Go read upgrade.txt and all the dozens of articles about whats been changed |
15:35.04 | [TK]D-Fender | Fusoya: "soft hangup Zap/1-1", etc |
15:35.08 | ccvp | d-fender |
15:35.21 | ccvp | is [TK] a guild/clan tag for you in an online game? :) |
15:35.23 | Fusoya | [TK]D-Fender: Ooooh, let me try that one. |
15:35.26 | ccvp | Tha Killaz? :) |
15:35.41 | [TK]D-Fender | ccvp: Many years ago, yes. I keep it for sentimental reasons. |
15:35.45 | ccvp | heh |
15:35.53 | ccvp | did you play qw/dm? |
15:35.56 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
15:35.58 | [TK]D-Fender | ccvp: and no, not "Tha Killaz" |
15:36.06 | stansmith | ~caturday |
15:37.06 | ccvp | bleh wtf, its 75 degrees yesterday and today, and a front is rolling through around 2am |
15:37.12 | ccvp | and gonna drainop 50 degrees in 1 hour, massive tornados yet ag |
15:37.14 | ccvp | again |
15:37.21 | stansmith | ccvp: location? |
15:37.24 | ccvp | north, AL |
15:37.27 | ccvp | just below TN |
15:37.41 | stansmith | its a nice 50 something here in columbus..hope that front doesnt reach here |
15:37.47 | ccvp | look at TX now |
15:37.49 | ccvp | on weather.com |
15:37.52 | ccvp | thick purple line of storms |
15:37.54 | ccvp | purple > red |
15:38.26 | ccvp | look at that front in TX |
15:38.28 | ccvp | its gigantic |
15:38.52 | ccvp | http://www.weather.com/maps/news/severewinterforecast/floater1_large_animated.html?from=hp_main_maps |
15:39.34 | stansmith | at least its not gonna snow |
15:39.49 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
15:39.51 | ccvp | that area in TX where there is a blizzard now |
15:39.55 | ccvp | had 83f 2 days ago |
15:40.06 | ccvp | global warming ftw :) |
15:40.13 | stansmith | you can say that again! |
15:40.23 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
15:40.26 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
15:42.56 | Fusoya | [TK]D-Fender: Doesn't seem to be able to hang up this call |
15:43.37 | pithen | So heres a dummy question.. these sip gateways you've pointed me to all have RJ21x connectors.. is that something I can just go to HomeDepot and pick up a junction block or something? |
15:45.07 | x86 | RJ21X? that's an Amphenol 25-pair connector |
15:45.40 | x86 | no, Home Depot does not sell them |
15:46.20 | pithen | meh..im going to have to rip apart the whole network closet for this project ;) |
15:46.24 | x86 | need to go to some wiring specialty place... here we go to Graybar, which is a wholesaler |
15:46.40 | Katty | if you're doing Background(menu) and you want to wait 15 seconds before repeating it... |
15:46.46 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
15:46.50 | x86 | you can probably hit up some local wiring vendor for RJ21X cables |
15:46.53 | Katty | do you use Wait(15)? For some reason, when i use wait, it won't accept my input |
15:47.14 | [TK]D-Fender | pithen: I haven't seen a mass gateway with 24x RJ11 before. |
15:47.17 | x86 | WaitExten is what you want Katty |
15:47.19 | De_Mon | Katty use WaitExten |
15:47.21 | Katty | thanks. |
15:47.37 | [TK]D-Fender | pithen: Closest I've seen is the SPA-8000 w/ 8 FXS |
15:47.57 | [TK]D-Fender | pithen: When when you think of it is better than 4x 2-port gateways |
15:47.57 | Fusoya | LOL, I love how the bug for this (6147) just got closed for no reason |
15:48.17 | x86 | Fusoya: happens quite frequently it seems ;) |
15:48.29 | [TK]D-Fender | Katty: "Set(TIMEOUT(response)=15" |
15:49.37 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.198) |
15:49.55 | pithen | [TK]D-Fender, thats what was appealing about the rhino bank, it had rj11's.. i suppose if im going to do this though i autta do it right |
15:50.11 | [TK]D-Fender | pithen: Which Rhino has RJ11's? |
15:50.15 | [TK]D-Fender | pithen: News to me... |
15:50.38 | coppice | i think their analogue cards do |
15:50.39 | tzafrir | [TK]D-Fender, huh? |
15:51.11 | De_Mon | Fusoya http://bugs.digium.com/view.php?id=6147? |
15:51.21 | De_Mon | no that can't be what you're talking about |
15:51.34 | tzafrir | we provide RJ11-s . or an optional bundled interface. It's much simpler to test the unit when you have an RJ11 output for each port |
15:52.01 | Fusoya | De_Mon: Yeah it is. |
15:52.19 | De_Mon | Fusoya the last comment was in 2006, and says exactly why it was closed |
15:53.02 | Fusoya | De_Mon: "I don't believe this bug is going anywhere" isn't a reason in my book to close a ticket. :) |
15:53.06 | Fusoya | But that's me |
15:53.49 | Fusoya | I can't find any more recent ones on this issue, but I see reports elsewhere of people having it very recently, with recent versions |
15:54.11 | De_Mon | read the whole comment and if you still feel that way, we'll just have to agree to disagree |
15:54.41 | Fusoya | We'll have to do that then, because I've read it. |
15:54.44 | De_Mon | bunching a bunch of problems into 1 ticket is an excelent reason to close that ticket and make them open separate issues. |
15:55.00 | Fusoya | There was no reason to believe the issues were seperate. |
15:55.03 | Fusoya | Separate, rather. |
15:55.22 | Fusoya | All produce the same symptom... how do you track a bug if not by symptom when the cause is unknown? |
15:55.42 | pithen | hmm..maybe im mistaken "Analog Output: 24 Loop start lines via 25 pair amphenol connector " i thought i saw a picture with 24 ports on it |
15:56.01 | Fusoya | Is it better for every user to open a ticket that says "hey I don't know why but I'm getting the same error as the people who posted the last 18 tickets" |
15:58.12 | [TK]D-Fender | pithen: So you'll need a break-out box |
16:00.17 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
16:01.04 | BCS-Satori | I noticed that my app followme was not loaded in asterisk but the module was there, so i did load app_followme.so which worked from the console. Will this stay there forever or when I reboot will it disappear or should i just add it under modules? |
16:01.24 | stansmith | BCS-Satori: pb modules.conf |
16:01.32 | stansmith | er..check modules.conf |
16:02.08 | BCS-Satori | stansmith: its not listed there before or after the load command |
16:02.14 | JenniferAkemi | BCS-Satori: you must have autoload=yes or have it loaded explicity |
16:02.21 | stansmith | ^ |
16:02.50 | BCS-Satori | autoload is set to yes, so i guess i need to laod it explicity then |
16:03.04 | JenniferAkemi | i think you should really only need one or the other |
16:03.32 | JenniferAkemi | you could always try rebooting and see if it's loaded :) |
16:03.53 | JenniferAkemi | i'm right now playing with modules.conf with autoload=no |
16:04.47 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-d01891759a901cbf) |
16:04.48 | *** join/#asterisk kraypius (i=user@c-67-175-202-53.hsd1.il.comcast.net) |
16:04.50 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
16:06.04 | kraypius | <PROTECTED> |
16:06.22 | [TK]D-Fender | kraypius: ~freepbx |
16:06.26 | [TK]D-Fender | ~freepbx |
16:06.26 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
16:06.42 | stansmith | lol |
16:07.32 | JenniferAkemi | do i need adsi for callerid on call waiting? |
16:07.46 | stansmith | how would a rookie programmer implicity set TIMEOUT()'s for each context in extensions.conf? |
16:07.51 | stansmith | hypothetically speaking that is |
16:08.22 | [TK]D-Fender | JenniferAkemi: No. |
16:08.33 | JenniferAkemi | cool thanks [TK]D-Fender |
16:08.36 | [TK]D-Fender | stansmith: There is no "implicit". You have to set them |
16:08.58 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
16:08.58 | *** mode/#asterisk [+o anthm] by ChanServ |
16:10.15 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
16:10.35 | Katty | hi anthm (= |
16:10.47 | anthm | hi =D |
16:11.15 | *** join/#asterisk VaNNi (n=VaNNi@216.70.165.200) |
16:11.33 | stansmith | which of these 2 context is recommended in production? http://www.pastebin.ca/926142 |
16:12.02 | stansmith | pretty much having While() vs. "t" extension |
16:13.18 | pithen | Okay.. final question to all before I get out of your hair: as far as FXO cards go, I need a 4 port card; Digium the best in terms of reliability/price, or should I consider other manufacturers (which?)? |
16:13.45 | stansmith | !dialogic |
16:13.54 | stansmith | (w/ asterisk that is) |
16:14.12 | [TK]D-Fender | stansmith: 1st, without "waitexten", but rather "autofallthrough=no", and also with an actual patter that will match. Next it'd be nice if your timeout counter worked. |
16:14.38 | [TK]D-Fender | pithen: I use Sangoma A200d's exclusively for that |
16:14.48 | *** join/#asterisk sudhir492 (n=sudhir@adsl-18-47-35.mco.bellsouth.net) |
16:14.53 | x86 | wow... all of my branches made a combined total of 47,392 calls LAST WEEK ALONE ;) |
16:14.58 | stansmith | i have REPEAT defined in [globals] |
16:15.15 | sudhir492 | Is there an FXO/FXS card for USB? |
16:15.24 | x86 | [TK]D-Fender: A20002D-x :) |
16:15.28 | [TK]D-Fender | stansmith: Wasn't talking about the global which I took for granted you'd at least done properly... |
16:15.50 | coppice | sudhir492: dunno. ask tzafrir :-) |
16:15.52 | [TK]D-Fender | stansmith: You have ANOTHER error. Stare at it till you see it, or your eyes bleed. |
16:15.58 | x86 | sudhir492: Xorcom makes USB channel banks, talk to tzafrir |
16:16.10 | tzafrir | sudhir492, hi |
16:16.58 | sudhir492 | hi tzafrir |
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16:19.13 | *** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-de5349e0ec31b467) |
16:19.46 | x86 | tzafrir: what kind of density does Xorcom offer with those things? |
16:19.56 | x86 | tzafrir: can yall do 48 ports in 1U yet? |
16:20.23 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
16:20.32 | ThatKidKel | CDR Question--Is it possible to have CDRs written to the database AND in cases of DB failure to the Master.csv? |
16:20.43 | stansmith | ThatKidKel: yes |
16:20.48 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
16:20.53 | ThatKidKel | can you point me to documentation? |
16:20.58 | tzafrir | x86, 32 ports . But you can connect many of them. |
16:21.04 | BCS-Satori | Is there a way to detect a hangup and execute the next line of code in a exten series? For example, i am setting up a monitor that will send the message they record to email, but if the user hangs up the phone it will never email. the only way it currently emails is when they press #. Is there a way to decect the hangup and move to next step? |
16:21.06 | stansmith | ThatKidKel: its really up to you, i do it via an AGI script |
16:21.23 | ThatKidKel | ah... |
16:21.55 | pithen | [TK]D-Fender, excellent..thank you so much |
16:22.13 | stansmith | ThatKidKel: i think it gets written to Master.csv regardless |
16:22.35 | ThatKidKel | ok. |
16:22.58 | ThatKidKel | i've noticed something werid in my Master.csv.. If its a NO ANSWER, i get two records.. |
16:23.08 | ThatKidKel | One with the destination of "s" and the other with the real destination |
16:23.35 | stansmith | s = destination to go here when there is nothing explicitly defined |
16:24.11 | stansmith | i use "s" as an entry point to the context |
16:24.14 | JenniferAkemi | maybe the phone is calling back |
16:25.38 | stansmith | [TK]D-Fender: were you trying to tell me to do "exten => _1,n,..." rather than "exten => 1,n,..." ? |
16:25.52 | stansmith | when you said pattern matching |
16:26.07 | [TK]D-Fender | stansmith: No. |
16:26.24 | [TK]D-Fender | stansmith: You had nothing DIALABLE in there. |
16:26.30 | *** join/#asterisk sergey_masushko (n=sergey@66.243.68.219) |
16:27.07 | stansmith | [TK]D-Fender: Swift() jumps to the extension...so when i press 1, it will go the the "1" extension and execute that |
16:27.18 | JenniferAkemi | there isn't even a exten => 1,n.... in your thing |
16:27.23 | [TK]D-Fender | stansmith: http://www.pastebin.ca/926142 <- Where do you see anything you can actually DIAL in there? You have no dial patterns. |
16:27.37 | stansmith | im not trying to dial though..its an IVR |
16:27.41 | stansmith | strictly an IVR |
16:27.47 | [TK]D-Fender | stansmith: You are NOT getting it.. |
16:27.53 | stansmith | i know im not :-( |
16:28.11 | [TK]D-Fender | stansmith: You can't dail anything in there. Everything comes up "i" and sends you in circles. |
16:28.53 | stansmith | oooooooooooo |
16:29.10 | stansmith | haha, the "..." meant there are extensions in there that i didnt PB cause i thought they were irrelevant |
16:29.12 | [TK]D-Fender | dial* |
16:29.20 | stansmith | they were just taking up space |
16:29.46 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
16:29.54 | [TK]D-Fender | stansmith: You show me something that laks some important pieces and I will assume you didn't think to actually make it DO anything. |
16:30.38 | JenniferAkemi | [TK]D-Fender: do you speak french too? |
16:30.46 | [TK]D-Fender | JenniferAkemi: Naturally. |
16:30.56 | JenniferAkemi | [TK]D-Fender: nice :) |
16:31.11 | [TK]D-Fender | s/laks/lacks |
16:31.15 | stansmith | my question was more directed towards "should i use a while loop or the "t" extension for the main extension in that context" |
16:31.15 | *** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com) |
16:31.20 | stansmith | sorry for the mix up [TK]D-Fender |
16:31.26 | JenniferAkemi | man i wish i didn't do install samples |
16:31.36 | JenniferAkemi | i wonder if thats why all these conf files are here. |
16:31.39 | [TK]D-Fender | stansmith: You ask which is better for production and I'll check if the ENTIRE thing works or not :) |
16:31.42 | stansmith | JenniferAkemi: it is |
16:31.59 | JenniferAkemi | i wish it would just make them blah.conf.sample |
16:32.10 | stansmith | haha, i was like "well he prob doesnt care to see "exten => 1,1,Goto(login,s,1)" |
16:32.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
16:32.38 | [TK]D-Fender | stansmith: Yes I want to see if you've done anything else silly... |
16:33.23 | [TK]D-Fender | stansmith: and what I was pointing out earlier is that your 2 context samples showed a bug between them. |
16:33.34 | [TK]D-Fender | stansmith: exten => s,5,Set(COUNT=$[${COUNT} + 1]) <- good. |
16:33.44 | [TK]D-Fender | stansmith: exten => t,1,Set(COUNT=${COUNT}+1) <- bad. |
16:33.55 | [TK]D-Fender | stansmith: Please pay attention to your use of * evaluations. |
16:34.14 | [TK]D-Fender | stansmith: Which is why I told you your rety loop was busted |
16:34.17 | [TK]D-Fender | retry* |
16:34.42 | stansmith | ahhhh - yea, the second example was my first iteration developing the IVR, i didnt really realize how $[] was supposed to be used |
16:35.03 | stansmith | by second iteration, i knew better *cough* read the f-ing manual *cough* |
16:35.13 | [TK]D-Fender | stansmith: 57th times the charm.... |
16:35.37 | stansmith | did i mention im fresh out of college ? |
16:35.44 | stansmith | good thing is, im highly trainable |
16:35.52 | stansmith | like a young puppy |
16:35.55 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
16:36.25 | ZPertee | does anyone have any experience with integrating asterisk with older pbx? |
16:36.42 | JenniferAkemi | what kind of experience |
16:36.50 | stansmith | what kind of integration? |
16:37.00 | JenniferAkemi | i'm connecting asterisk to my older stuff |
16:37.12 | JenniferAkemi | but, it's just like being a pri provider |
16:37.24 | JenniferAkemi | it being the older switch |
16:37.58 | stansmith | De_Mon: thanks, i just saw it |
16:38.04 | ZPertee | I have an avaya partner acs system and I want to put asterisk as an analog extension on it. however, asterisk won't answer my call |
16:38.06 | stansmith | ur message that is |
16:38.13 | JenniferAkemi | what is a "skinny channel" in asterisk? |
16:38.17 | *** part/#asterisk sergey_masushko (n=sergey@66.243.68.219) |
16:38.31 | drmessano | --> )( |
16:38.32 | ZPertee | I will also sit asterisk in front of avaya as well |
16:38.50 | [TK]D-Fender | JenniferAkemi: SCCP protocol (Cisco Phones) |
16:39.01 | JenniferAkemi | [TK]D-Fender: thank you |
16:39.04 | ZPertee | what signalling would I use on my Digium fxo card? |
16:39.16 | stansmith | fxs_ks is recommended |
16:39.18 | drmessano | SCCP, one letter off from CCCP.. no coincidence |
16:39.28 | stansmith | ZPertee: wait dont listen to me |
16:40.09 | ZPertee | ok |
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16:41.52 | *** join/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
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16:43.13 | stansmith | ZPertee: maybe i am right |
16:43.19 | stansmith | are you referring to zapata.conf? |
16:43.26 | [TK]D-Fender | ZPertee: Typcally fxs_ks |
16:43.34 | stansmith | ZPertee: yea i was right |
16:45.31 | tzafrir | drmessano, but also one letter from scp |
16:45.34 | ZPertee | ok. so if asterisk doesn't answer then either the dial plan is wrong or the fxo card? any idea how I configure my fxo cards in asterisk now? supposed to be automatic but I can't figure out how to see if the card is even working or not |
16:46.36 | stansmith | ZPertee: in zapata.conf, u need to tell it where to enter the dialplan at |
16:46.46 | tzafrir | ZPertee, asterisk doesn't answer? |
16:46.51 | stansmith | for me, it is "context=init" because i have a context [init] |
16:47.15 | tzafrir | why does it sound familiar |
16:47.18 | tzafrir | ? |
16:47.30 | stansmith | o lol |
16:47.43 | stansmith | ZPertee: in your context, is the first thing Answer()? |
16:47.55 | ZPertee | yes and I have it pointed there |
16:47.58 | stansmith | o.. |
16:48.08 | ZPertee | can I test the card from the asterisk console? |
16:48.15 | ZPertee | or get any info? |
16:48.28 | stansmith | ZPertee: are you running zaptel service? |
16:48.41 | stansmith | i.e - is ur driver loaded? |
16:48.56 | tzafrir | cat /proc/zaptel/* |
16:49.01 | stansmith | (i like `lsmod | grep zap`) |
16:49.11 | ZPertee | thanx...newb |
16:49.30 | ZPertee | yep its running |
16:49.33 | *** join/#asterisk mihinomenest (n=argh@66.255.220.17) |
16:49.52 | ZPertee | i'll play with it some more, prob something stupid...usually is |
16:50.16 | stansmith | ZPertee: if you edited zapata.conf or extensions.conf you should restart asterisk |
16:50.35 | ZPertee | ok |
16:50.36 | stansmith | dialplan reload will reload extensions.conf while asterisk is running, but i dont know the command to reload zapata.conf |
16:50.43 | stansmith | "dialplan reload" |
16:51.33 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
16:51.39 | ZPertee | whats difference between "reload" and "dialplan reload"? anything |
16:51.50 | stansmith | reload reloads everything perhaps? |
16:52.03 | stansmith | i know there is a command to reload everything, that may be it, i have never used it |
16:52.34 | ZPertee | seems to take longer... so its possible |
16:52.42 | Qwell | help reload |
16:53.21 | ZPertee | <PROTECTED> |
16:53.21 | ZPertee | <PROTECTED> |
16:53.27 | tzafrir | stansmith, for most things: 'reload' or 'modules reload chan_zap.so' |
16:53.46 | stansmith | ah..i tried "modules reload zapata" and it error'd |
16:55.11 | ZPertee | would it hurt anything if I hava an ata connected to * with extension 1. and then I dial ext.1 and and have a telephone line plugged from ata to * fxo? |
16:55.27 | ZPertee | dial from softphone |
16:55.40 | ZPertee | trying to test system |
16:57.07 | stansmith | i test via softphone |
16:57.12 | stansmith | make sure u edit sip.conf accordingly |
17:00.25 | stansmith | does anyone here use asterisk to port knock their gateway/firewall? i.e - is that idea original or old news? |
17:01.05 | *** join/#asterisk ddunavant (n=David@pool-71-191-24-36.washdc.east.verizon.net) |
17:01.30 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
17:02.23 | Havokmon | You mean allow incoming connections (RDP/ssh/etc) from the same IP a user registered their phone from? Interesting thought... |
17:02.50 | *** join/#asterisk af_ (n=getsmart@88-149-241-244.dynamic.ngi.it) |
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17:06.03 | ZPertee | If my fxo card on my * box is configured correctly should I at least be able to see an incoming call in the asterisk console? |
17:06.15 | stansmith | i mean to have asterisk on same box as your gateway, then call it when you are not home, dial extensions which are really just the ports you want to knock on, and if its the right sequence then run a simple bash script to alter iptables |
17:06.36 | stansmith | im waiting on my w2 so i can buy a card at home and try it |
17:06.47 | Qwell | stansmith: you haven't gotten it yet? |
17:06.54 | stansmith | i mean, the actual cash |
17:06.57 | Qwell | oh |
17:07.02 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
17:07.21 | stansmith | ZPertee: yes |
17:07.41 | ZPertee | stansmith if your firewall can be controlled by the command line couldn't use just use the System() application |
17:07.45 | stansmith | you might see something like "starting simple switch on ... " |
17:08.08 | stansmith | yea, more then 1 way to skin a cat..typical with *nix |
17:08.24 | ZPertee | you gotta love it though |
17:10.47 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
17:12.02 | grandpapadot | Hi all. In this outbound dialing extension: exten => _91NXXNXXXXXX,1,dial(Zap/g1/${EXTEN}) If g1 has no free lines, how do I get congesion playing to the callers? I tried just adding a priority right below this one with congestion but it's not working. |
17:12.38 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
17:13.14 | ManxPower | What's that online website/store that has to do with mozilla that sells VoIP stuff. I just can't remember the name of it. |
17:14.12 | ManxPower | voxilla was the one I was thinking, just found it |
17:14.14 | stansmith | voipzilla? |
17:14.24 | stansmith | thats actually a good name... |
17:17.41 | grandpapadot | If a zaptel group has no free channels shouldn't Dial() exit and the call flow to the next priority? |
17:19.28 | [TK]D-Fender | grandpapadot: it does, so you've done something wrong... |
17:19.35 | [TK]D-Fender | grandpapadot: pastebin is your friend. |
17:19.42 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
17:20.38 | coppice | except when it remembers your embarassing mistakes |
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17:30.50 | *** part/#asterisk FlatFoot (n=bigflatf@80.88.192.83) |
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17:37.38 | generalhan | can someone tell me what this 'fullcontact' line is all about in the sip DB |
17:37.53 | grandpapadot | [TK]D-Fender: Looks like n+101 is where the calls were going ... |
17:37.59 | generalhan | according to how the table is setup it cannot be NULL, but i cant figure out what i need there |
17:38.27 | [TK]D-Fender | grandpapadot: You should not be doing priority jumping since 1.2 |
17:38.44 | x86 | priority jumping is such a hack |
17:39.08 | x86 | grandpapadot: create a macro that checks dialstatus and makes decisions based on that |
17:39.23 | Wayhigh | got a refund from x100p.com. |
17:39.45 | *** join/#asterisk Skarmeth (n=Skarmeth@200.253.26.150) |
17:43.55 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
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17:45.39 | tzafrir | Wayhigh, interesting |
17:45.46 | tzafrir | They actually respond |
17:47.00 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
17:48.27 | Wayhigh | tza: yeah but it took me a lot.. they're going through some HR problems apparently |
17:48.45 | Katty | mmm, lunch. |
17:48.48 | Katty | i haz it! |
17:48.58 | *** join/#asterisk atis_work (n=atis_wor@81.198.164.2) |
17:48.58 | stansmith | lolkat? |
17:49.08 | Katty | stansmith: precisely. |
17:49.13 | stansmith | ha |
17:49.30 | Katty | stansmith: this is why asterisk syntax sometimes does not parse. |
17:49.33 | Katty | stansmith: it does not speak kat. |
17:49.59 | stansmith | haha wow |
17:50.32 | *** join/#asterisk greekguy8888 (n=alex@c-76-118-204-95.hsd1.ma.comcast.net) |
17:51.23 | greekguy8888 | anyone in here? |
17:51.23 | [TK]D-Fender | *b00m* |
17:51.23 | stansmith | omg netsplit! |
17:51.23 | *** join/#asterisk angryuser (i=nononon@df01t2-212-194-99-78.d4.club-internet.fr) [NETSPLIT VICTIM] |
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17:51.25 | Wayhigh | i've got $43 to spend on voip gear.. what can I get? |
17:51.25 | russellb | that was hot |
17:51.25 | Wayhigh | :P |
17:51.41 | mvanbaak | ;) |
17:51.46 | mvanbaak | howz you ? |
17:51.52 | russellb | quite good. |
17:51.55 | mvanbaak | good |
17:51.57 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM] |
17:52.00 | Katty | [TK]D-Fender: "no boom today, boom tomorrow. there's always a boom tomorrow.... What? somebody's gotta have some damn perspective around here. Boom! Sooner or later. BOOM!" |
17:52.03 | Katty | [TK]D-Fender: name the quote. |
17:52.15 | mvanbaak | can * do video conferencing ? |
17:52.34 | stansmith | i collect straws |
17:52.42 | Katty | cpm: you sir, must be russian. |
17:52.47 | greekguy8888 | lawl |
17:52.57 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-57-203.lns10.syd7.internode.on.net) |
17:53.04 | [TK]D-Fender | Katty: B5 FTW |
17:53.14 | Katty | [TK]D-Fender: horay! my reference didn't go unnoticed! |
17:53.21 | mvanbaak | I have to setup this 'proof-of-concept' system with video conferencing |
17:53.27 | mvanbaak | I want to use * where possible |
17:53.41 | stansmith | ~B5 |
17:53.46 | stansmith | !starwars |
17:53.46 | Katty | jbot: babylon 5? |
17:53.52 | greekguy8888 | does anyone know if there is a workaround for transferring calls deliverd by queue via the sip transfer button on a phone? (currently only releases agent channel if u use asterisk native transfer) |
17:53.53 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-157-126.bltmmd.east.verizon.net) |
17:53.53 | Katty | jbot: B5? |
17:54.29 | Katty | someone teach jbot that B5 and Babylon 5 = http://en.wikipedia.org/wiki/Babylon_5 |
17:55.22 | Wayhigh | jbot: B5 is also at http://en.wikipedia.org/wiki/Babylon_5 |
17:55.23 | jbot | Wayhigh: okay |
17:55.41 | Wayhigh | jbot: Babylon 5 is also at http://en.wikipedia.org/wiki/Babylon_5 |
17:55.41 | jbot | Wayhigh: okay |
17:55.43 | sudhir492 | Does someone here have a recommendation for an ATA other than Linksys? |
17:55.47 | Wayhigh | ~wayhigh |
17:55.48 | jbot | Asterisk mouse WAZ in his 1U, eatinz his thermo ribbons.. HE R MOUSEKILLA |
17:56.00 | greekguy8888 | the granstream ata's are only other option i think |
17:56.29 | stansmith | wow this chick is snoring so loud! |
17:56.31 | tzafrir | sipura :-) |
17:56.32 | stansmith | should i wake her up? |
17:56.45 | greekguy8888 | give her an elbow lol |
17:56.48 | stansmith | lol |
17:56.50 | Wayhigh | sudhir: depends what you're looking for.. I'm having good luck with my x100p.com S100-FX |
17:57.09 | Qwell | ~cheap |
17:57.09 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
17:57.11 | Wayhigh | stansmith: depends on if he's a gnaw your arm off kinda gal |
17:57.37 | stansmith | haha...i kinda need both my arms today |
17:57.47 | greekguy8888 | nice |
17:58.37 | tzafrir | Well, IAX ATAs are probably better targetted for traveling and such |
17:59.27 | *** join/#asterisk djs (n=djs@unaffiliated/djs26) |
18:00.58 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
18:02.30 | *** join/#asterisk eGecko2 (i=egecko@cpe-76-176-233-72.san.res.rr.com) |
18:03.12 | stansmith | lol, ecoterrorism? |
18:04.39 | stansmith | LOL are you even allowed to do that in IRC? |
18:04.43 | x86 | oh noes now i'm teh ecoterrorismnist! |
18:04.48 | stansmith | LOL |
18:04.58 | Qwell | stansmith: I don't think there's a specific rule against it.. |
18:05.02 | x86 | stansmith: "allowed" and "IRC" should not be used in the same sentance ;) |
18:05.11 | stansmith | haha |
18:05.38 | generalhan | can anyone tell me what this warning is all about ? app_voicemail.c:2262 inboxcount: Failed to obtain database object for 'asterisk'! i just loaded a new 1.4 install with realtime and this is just constantly scrolling |
18:07.18 | greekguy8888 | looks like your realtime vm is not connecting |
18:07.39 | greekguy8888 | <trying this again> does anyone know if there is a workaround for transferring calls deliverd by queue via the sip transfer button on a phone? (currently only releases agent channel if u use asterisk native transfer) |
18:07.44 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
18:07.48 | generalhan | greekguy8888: any ideas as to why that might happen ? |
18:07.57 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
18:08.33 | greekguy8888 | u did not setup your realtime correctly, could be a number of thinngs... the fact it can't find an object says either its not connecting to the db, OR... you don't have a vm box setup for what is being accessed |
18:09.11 | greekguy8888 | probably the first thing if its consantyl scrolling |
18:09.34 | generalhan | i see. well lets see if i can figure out why it wouldnt be connecting |
18:11.15 | *** join/#asterisk angryuser[A] (i=nononon@df01t2-213-44-144-97.d4.club-internet.fr) |
18:12.03 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
18:14.24 | x86 | generalhan: execute this from CLI: "realtime mysql status" |
18:14.27 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
18:14.37 | x86 | replace mysql with whatever database driver you're using |
18:14.54 | x86 | generalhan: for CDR, do this "cdr mysql status" |
18:15.17 | generalhan | x86: it says connected |
18:15.29 | x86 | then your database is not setup properly |
18:15.31 | generalhan | but you know what ... im not getting that warning anymore, and i think i know why |
18:15.33 | x86 | did you load the schema? |
18:15.41 | x86 | ;) |
18:15.53 | stansmith | schemas ftw |
18:16.07 | generalhan | i had copied over my configs to the new server ... so there was a sip.conf entry for phones that arent in the DB, so i think it was trying to load the VM info for those phones that wasnt setup |
18:16.20 | generalhan | so when i removed the sip.conf references the warning went away |
18:17.00 | generalhan | but ... should i still be able to see the phones that are setup in the DB by using sip show peers in the CLI ? |
18:17.07 | generalhan | cause it produces nothing |
18:17.16 | *** join/#asterisk dexpdx (n=dexpdx@66-162-134-242.static.twtelecom.net) |
18:18.22 | *** join/#asterisk Tommy3 (n=Tommy2@66.0.46.210) |
18:19.22 | dexpdx | Anyone run across a scenario where cdr_odbc connect's just fine bust doesn't store records in that datbase |
18:19.44 | dexpdx | with cdr-csv working fine, isql connecting & inserting just fine |
18:20.01 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
18:20.25 | hmmhesays | odbc killing the connection after so many hours? |
18:20.32 | dexpdx | nope |
18:20.47 | dexpdx | stair off a fresh restart no inserts |
18:20.55 | dexpdx | s/stair/strait/ |
18:21.51 | *** join/#asterisk jlb (n=jlb@75.148.162.90) |
18:23.18 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
18:24.18 | JayTee52 | What is the best or preferred voice recoginition add on for Asterisk for creating a Dial by name voice menu? |
18:24.29 | *** join/#asterisk flujan_ (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
18:25.44 | dexpdx | Here is my CLI output: shows no errors: http://pastebin.ca/926420 |
18:38.10 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
18:39.50 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
18:41.29 | generalhan | can anyone tell me what ${DATETIME} was replaced by in 1.4 ?? |
18:41.50 | stansmith | generalhan: http://www.voip-info.org/wiki-Asterisk+variables |
18:41.57 | generalhan | ty |
18:44.00 | ccvp | what would be a good piece of software to deploy to a website, that would be community driven, where people could whistle blow about a particular topic, and submit images, and videos as evidence of something going on, that would have a web 2.0 feel to it? |
18:44.43 | stansmith | what is a wiki? |
18:44.56 | ccvp | for $200 please |
18:45.00 | ccvp | Thanks Alex Trebek |
18:45.18 | Qwell | ccvp: youtube |
18:45.31 | ccvp | well something i can integrate w/ adsense |
18:45.32 | ccvp | heh |
18:45.38 | ccvp | personal gain, not no profit |
18:45.51 | stansmith | adsense can integrate with anything html, no? |
18:45.52 | Qwell | monetary gain from a whistle blower site? |
18:46.00 | Qwell | yeeeeeeaaahhhhh.... good luck with that |
18:46.02 | ccvp | the more people that come , the more CPM |
18:46.15 | ccvp | adsense is a nifty tool qwell , when used w/ adsense |
18:46.24 | ccvp | payments based on ad impression, and more payment when their actually clicked |
18:46.30 | cpm | the more cpm what? |
18:46.46 | ccvp | impression counts |
18:46.52 | dexpdx | anyone here familiar with cdr_odbc? |
18:47.01 | dexpdx | i.e. troubleshooting it |
18:47.06 | ccvp | google adsense also has a calculation that pays pays based on how many people see an add, but its smaller |
18:47.11 | ccvp | compared to if someone actually clicks an ad |
18:47.13 | stansmith | ccvp: impressions get paid out by every thousand visitors though, no? |
18:47.17 | ccvp | yep |
18:47.30 | ccvp | when i tested a site of mine with TOR |
18:47.37 | ccvp | so my ip isnt linked to me, and clicked an add twice |
18:47.42 | ccvp | I got $2.80 cents in 4 minutes |
18:47.47 | stansmith | thats illegal |
18:47.49 | ccvp | but I wont abuse, risk getting my adsense account terminated |
18:48.09 | ccvp | :) |
18:48.19 | stansmith | ;) |
18:48.37 | stansmith | 1.40 a click? i thought it was pennies a click |
18:48.43 | ccvp | some ads pay more |
18:48.50 | ccvp | im not an expert on adsense |
18:49.04 | ccvp | my MIS 595 class has 2 weeks on adsense next semester |
18:49.07 | stansmith | whoa whoa whoa |
18:49.30 | ccvp | im surprised that hardly |
18:49.33 | ccvp | anyone knows about adsense |
18:49.37 | ccvp | how it works etc,,,,, |
18:49.48 | ccvp | i know us tech geeks despise ads, but theres a market , billions of people use the internet |
18:49.51 | stansmith | ccvp: here check me out, im about to make u a lot of money |
18:50.28 | stansmith | you set up an asterisk box, and then anytime someone calls you via voip, you use their IP to fetch the URL of a page that has an ad on it |
18:50.31 | stansmith | wammy |
18:50.37 | stansmith | like stealing candy from a baby |
18:50.47 | ccvp | you cant target google ads |
18:50.51 | ccvp | its hidden in javascript, heh |
18:51.27 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
18:51.38 | stansmith | i was thinking more along the lines of the impressions |
18:51.47 | ccvp | fractional |
18:51.55 | stansmith | slow progress > none |
18:52.00 | ccvp | better bet would be to malware 10's of thousands of people |
18:52.04 | ccvp | and have drones do the click fraud heh |
18:52.05 | stansmith | lol |
18:52.42 | stansmith | what domain were u gonna put the adsense on? |
18:52.55 | ccvp | cannot say heh |
18:52.57 | stansmith | shorter, more basic domains = higher return |
18:53.05 | stansmith | voipzilla.com? |
18:53.07 | ccvp | yep, but alot of them are already taken |
18:53.08 | *** join/#asterisk vap0rtranz (n=jpittman@75.110.17.157) |
18:53.19 | ccvp | back in the day in like |
18:53.22 | ccvp | 1994,1995, some guy made like |
18:53.28 | ccvp | 280 million before patent squatting was law |
18:53.35 | ccvp | reserved 500+ website domains ofm ajor corporations |
18:53.41 | ccvp | before they had web presences |
18:54.18 | stansmith | ha..i was interning at a company that was hired to redo sex.com..we had all the traffic info for that domain..made a million dollars every month or so just redirecting international traffic |
18:54.32 | stansmith | of course, the name itself goes for a couple million |
18:55.06 | stansmith | at least |
18:56.38 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
18:58.11 | vap0rtranz | why does my last peer in users.conf match against all incoming calls? so dialing the next-to-last peer's DID fails with 'rejected ... extension not found.' |
18:58.46 | stansmith | ~externalivr |
18:59.17 | grandpapadot | Anyone here use Packet 8? What phones are they selling under their brand? We have a customer that wants to convert and I was just doing some preliminary research on the phones. |
19:04.58 | *** part/#asterisk jlb (n=jlb@75.148.162.90) |
19:05.04 | dexpdx | grandpapadot: probably what ever they can get their hands on for the smallest overhead |
19:05.15 | *** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca) |
19:07.19 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
19:10.35 | Tommy3 | zzz |
19:11.51 | Tommy3 | any recommendations on where to go for local phone number over voip for my asterisk box. shopping on the web wore me out with poor results. |
19:11.54 | J4k3 | packet8 stuff is ghettofied |
19:12.09 | dexpdx | <PROTECTED> |
19:12.10 | dexpdx | err! |
19:12.13 | J4k3 | they're about onpar with vonage.... amateur hour. |
19:12.19 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
19:12.23 | J4k3 | err, overpriced amateur hour |
19:12.46 | Tommy3 | using skype now, but not sure I can use it with asterisk... |
19:13.04 | *** part/#asterisk ice_croft (n=nolan@85.172.5.106) |
19:13.32 | lmadsen | you can't |
19:14.35 | J4k3 | Tommy3: there are lots of different offers out there, the issue is getting something that works for tou |
19:14.38 | J4k3 | err you |
19:14.55 | Katty | mmmmmapple! |
19:15.05 | J4k3 | gapple |
19:15.10 | Katty | also good |
19:15.16 | Katty | and very juicy! |
19:15.28 | Katty | bit spensive tho :< |
19:15.56 | BrianR___ | At least with number portability you can switch if things don't work out. |
19:16.04 | Katty | i gave up soda tho |
19:16.13 | BrianR___ | I've used both gafachi and vitelity for US48 termination... |
19:16.15 | lmadsen | I try not to drink pop anymore |
19:16.17 | Katty | miss it :/ |
19:16.25 | Katty | lmadsen: yeah. lotta stuff in it |
19:16.27 | lmadsen | my problem is I get addicted to it :) |
19:16.31 | alrs | I was late this morning so I was unable to stop at the grocer and restock my Earl Grey cache |
19:16.31 | Katty | lmadsen: especially the diet version. |
19:16.34 | lmadsen | Katty: gives you a bit of a beer belly too :) |
19:16.37 | BrianR___ | My feeling is that the gafachi calls are slightly higher quality. They also do t.38 fax |
19:16.42 | Katty | lmadsen: i imagine |
19:16.49 | Katty | lmadsen: it made me SO incredibly hungry |
19:16.52 | Katty | lmadsen: i could eat an elephant |
19:16.55 | alrs | BrianR___: I've had really good luck with Gafachi, you? |
19:16.56 | lmadsen | and gut rot |
19:17.03 | Katty | lmadsen: and that's Not Good when you can't go get a snack :< |
19:17.04 | stansmith | elephants are at the top of the food chain |
19:17.08 | lmadsen | indeed |
19:17.17 | Katty | )_= |
19:17.22 | Katty | hates you |
19:17.28 | lmadsen | I made pasta and a veggie sauce for lunch, and that was only 2 hours ago, and I'm hungry again |
19:17.38 | lmadsen | you hate me? that is sad... |
19:17.40 | Katty | what kind of pasta? |
19:17.44 | lmadsen | fuscilli |
19:17.50 | J4k3 | Earl Grey?!?! floor sweepings! |
19:17.51 | Katty | is that made from flour or egg? |
19:17.58 | lmadsen | hrmmm.... good question |
19:17.59 | *** join/#asterisk e` (n=e@38.102.196.202) |
19:17.59 | stansmith | lol |
19:18.03 | Tommy3 | is there a good service that I should look at. I can only find those rediculous monthly rates so far. would like something around $35/year with unlimited calling and a local phone number. would rather not mess with per minute stuff (hate suprizes). Ok, got gapple, gafachi, vitelity. enough for me to go do research. thanks guys! and Katty too! |
19:18.07 | Katty | what kind of veggies did you eat with it? |
19:18.10 | lmadsen | now you have me curious... /me goes to the kitchen to check |
19:18.12 | alrs | J4k3 I've got a bergamot problem |
19:18.27 | Katty | lmadsen: i'm going to wager it wasn't whole wheat pasta. |
19:18.32 | Katty | lmadsen: and you didn't eat enough veggies |
19:18.40 | Katty | lmadsen: and thus your being eaten from the inside OUT |
19:18.40 | J4k3 | Tommy3: vitelity offers various plans, depending on rate center |
19:18.56 | lmadsen | Katty: onion, garlic, orange pepper, tomato, and some hot peppers with butter, chili power, lemon and black pepper, and red chili flakes |
19:19.01 | J4k3 | ie - where I live I can get a $7.95/month unlimited incoming line that offers multiple call paths (I've pushed 3 calls over it, so far) |
19:19.08 | J4k3 | err 3 simultanious calls |
19:19.08 | Katty | lmadsen: that sounds lovely. |
19:19.15 | Katty | lmadsen: but how much did you use per serving total of veggies? 1c? |
19:19.17 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
19:19.21 | lmadsen | Katty: ya... I have whole wheat pasta in the cupboard, but wanted to try the fuscilli |
19:19.28 | J4k3 | but, you go 25 miles east of that rate center, where my girlfriend lives, and the best they offer is $1.50/mo + $1.49/minute |
19:19.37 | Katty | lmadsen: it sounds exotic! |
19:19.45 | lmadsen | Katty: oh.. I usage a whole tomato, a whole orange pepper, whole hot pepper, whole small onion, etc.... |
19:19.51 | ZPertee | what is the cheapest way to get a SIP DID? don't care what the number is as long as it is in the USA |
19:19.54 | BrianR___ | alrs: No major complaints. |
19:19.56 | Katty | lmadsen: ahh.. |
19:19.59 | lmadsen | Katty: so I got like... a good chunk of veggies |
19:20.03 | Katty | mad. |
19:20.09 | BrianR___ | alrs: COmpared to vitelity, I think whatever Gafachi is using for echo can does a better job. |
19:20.15 | Katty | lmadsen: stop running up and down stairs! |
19:20.17 | lmadsen | I like to do half veggies, half pasta |
19:20.24 | Katty | lmadsen: that's about what i do |
19:20.27 | lmadsen | Katty: isn't that like... good for you? :) |
19:20.32 | [TK]D-Fender | ZPertee: www.ipkall.com <- free |
19:20.35 | lmadsen | plus I don't have any stairs in my condo |
19:20.49 | BrianR___ | alrs: Also, there's some problems with REINVITE on Vitelity and they don't seem interested in helping me. When I transfer a call on Vitelity, I get a trombone shaped audio path. |
19:20.51 | Katty | lmadsen: 6oz grains, 5oz protein, 3 C Dairy, 1.5 C Fruit, 2.5-3 C veggies is what i eat a day |
19:21.09 | BrianR___ | alrs: But when I transfer on Gafachi, my asterisk winds up out of the audio path. |
19:21.12 | lmadsen | Katty: ya... I need a better meal plan. I do pretty good, but I don't have a set plan |
19:21.12 | J4k3 | gafachi seems kinda outrageously overpriced til you hit 10k/month for termination |
19:21.20 | Katty | lmadsen: 1C rice = 2 oz grains |
19:21.30 | Katty | lmadsen: go to mypyramid.gov |
19:21.30 | BrianR___ | J4k3: I agree that their termination prices are high.. |
19:21.33 | Katty | lmadsen: they have a handy generator |
19:21.36 | lmadsen | Katty: oh nice! |
19:21.38 | lmadsen | I'll check it out for sure |
19:21.39 | Katty | lmadsen: pretty accurate too |
19:21.46 | lmadsen | I've been meaning to put together a good diet plan |
19:21.48 | lnx | when i use SIP/0001*001 channel why my endpoint is ringing? :) And when exten => s,1,Dial(SIP/ipphone) executed? Where can i find base documentation about channels etc... ? |
19:21.48 | BrianR___ | Katty: I though that site was for social security... |
19:21.56 | lmadsen | Katty: well... not diet -- healthy eating plan |
19:22.01 | Katty | lmadsen: indeed. |
19:22.09 | Katty | lmadsen: we don't use the four letter D word in my house |
19:22.21 | ZPertee | [TK]D-Fender: thanks |
19:22.41 | BrianR___ | I currently use gafachi for my fax stuff, vitelity for my home voip (because they have e911) |
19:22.44 | Katty | lmadsen: i blogged it too, if you wanan see what i did: http://angela.sleekgeek.org/category/health/mypyramidgov/ |
19:22.49 | J4k3 | diet is overrated, everyone needs more sexercise. |
19:22.53 | lnx | i must learn SIP basics |
19:23.04 | stansmith | ~sexercise |
19:23.16 | [TK]D-Fender | lnx: You need to learn * basics... go read.. THE BOOK |
19:23.16 | stansmith | jbot like 0/4 today |
19:23.17 | jbot | ACTION smooches 0/4 today on the lips |
19:23.18 | [TK]D-Fender | ~book |
19:23.19 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
19:23.22 | lmadsen | i use 0% milk, eat cereal like multi-grain cheerios and Just Right, eggs and veggies, whole grain (most of the time) pasta and veggies, long grain and wild rice, etc... |
19:23.36 | lmadsen | then I tend to snack on trail mix and such |
19:23.45 | stansmith | trail mix is the whip |
19:23.48 | Katty | lmadsen: beginning http://angela.sleekgeek.org/category/health/mypyramidgov/page/9/ |
19:23.51 | stansmith | good, and good for u! |
19:23.55 | J4k3 | trail mix = simple carbos deluxe. |
19:23.56 | [TK]D-Fender | lmadsen: Nuke the grain overload.... |
19:24.01 | J4k3 | straight to your ass. |
19:24.08 | [TK]D-Fender | lmadsen: and indeed trail-mix = EVIL |
19:24.13 | lmadsen | ya but it's tasty |
19:24.17 | stansmith | good, and good for u! |
19:24.18 | lmadsen | I'm not trying to be a stick here :) |
19:24.19 | Katty | depends on the trailmix |
19:24.28 | Katty | dried peas have a nice crunch |
19:24.29 | lmadsen | stuff I have is mostly cranberries |
19:24.41 | Katty | i tend to snack on fruit |
19:24.50 | Katty | a ginormous apple takes a good 10 minutes to eat |
19:24.51 | lnx | http://www.asteriskdocs.org/ how nice |
19:24.59 | [TK]D-Fender | Katty: Pretty much all of them are a caloric overload and the serving size accounts for too much of your dietary intake. |
19:25.10 | lnx | ~book |
19:25.11 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
19:25.14 | Katty | [TK]D-Fender: baroo? |
19:25.23 | Katty | [TK]D-Fender: apple is not a caloric overload |
19:25.28 | [TK]D-Fender | Katty: talking about trail-mix here.. |
19:25.32 | Katty | [TK]D-Fender: oh, right. |
19:25.38 | Katty | yes. trailmix can be bad. |
19:25.41 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
19:25.43 | Katty | but! can also be properly accomodated for |
19:26.03 | Katty | you could eat a little less rice and pasta throughout the day to accomodate for some trailmix snackings |
19:26.14 | Katty | or... mayhaps, make it a weekly treat. that won't screw anything up. |
19:26.23 | BCS-Satori | I am looking for an 8 port FXO gateway to use with asterisk. Besides the grandstream is there another that is better or is the grandstream the way to go |
19:26.39 | J4k3 | heres a question: why hasn't somebody come up with an asterisk gui that isn't a trainwreck (freepbx) |
19:26.48 | J4k3 | seems like there'd be millions of dollars for that. |
19:26.49 | lmadsen | I don't think anyone here is ever going to say grandstream is the way to go |
19:26.57 | lmadsen | J4k3: because it's hard |
19:27.06 | J4k3 | grandstream is the way to go.... if you've got $60 to spend and need two handsets |
19:27.06 | [TK]D-Fender | BCS-Satori: Sangoma A200d. |
19:27.15 | Katty | BCS-Satori: Sangoma++ |
19:27.34 | J4k3 | otherwise I'd avoid buying grandstream's stuff |
19:27.38 | J4k3 | gs's ATAs made me cry |
19:27.41 | BCS-Satori | I personally never used grandstream i heard they were cheap |
19:27.43 | J4k3 | they got returned |
19:28.04 | [TK]D-Fender | ~gs |
19:28.05 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
19:28.05 | *** part/#asterisk pkunkra (n=chris@cpe-74-73-10-32.nyc.res.rr.com) |
19:28.22 | SteveTotaro | gs ATAs are decemt |
19:28.24 | J4k3 | oh lord, somebody woke up ol duhfender. |
19:28.25 | SteveTotaro | decent |
19:28.29 | J4k3 | ;) |
19:28.48 | J4k3 | SteveTotaro: they're ok. I didn't wanna get stuck having to support them :) |
19:29.06 | SteveTotaro | i sold them when i had a webstore |
19:29.13 | BCS-Satori | [TK]D-Fender: the Sangoma A200d seems very nice, is there anything thats an external product sorta like SPA400 |
19:29.25 | SteveTotaro | many bricked ATAs during firmware |
19:29.30 | [TK]D-Fender | BCS-Satori: Only decent ones are quite pricy. |
19:29.40 | [TK]D-Fender | BCS-Satori: AudioCodes MP series |
19:29.57 | CanWood | Great... we just bought 4 Grandstream phones and set up and Asterisk box yesterday with the intentions of rolling out a 12 user site |
19:30.02 | SteveTotaro | but in my use, the ATAs have been good for things like analog polycom conference phones |
19:30.05 | [TK]D-Fender | BCS-Satori: Thats the thing... FXS is quite aggressively priced, FXO not so. |
19:30.10 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
19:30.11 | J4k3 | CanWood: which model phones? |
19:30.18 | CanWood | GXP2000 |
19:30.22 | CanWood | yep |
19:30.25 | SteveTotaro | they seem ok |
19:30.34 | J4k3 | gxp2000's don't seem too bad |
19:30.35 | CanWood | and two TDM808B cards |
19:30.48 | BCS-Satori | [TK]D-Fender: Thanks for the information |
19:30.50 | [TK]D-Fender | CanWood: 16 FXO? |
19:30.51 | SteveTotaro | i just took on a support role for a company that has gxp2000 |
19:30.57 | CanWood | 2x8 |
19:31.00 | SteveTotaro | they like them |
19:31.04 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
19:31.09 | CanWood | good to hear |
19:31.11 | J4k3 | I think I'm going to take the plunge and get a nxtvox nxa800p and a couple modules |
19:31.16 | [TK]D-Fender | CanWood: Fractional PRI is not an viable option where you are? |
19:31.18 | J4k3 | I need an FXO and an FXS would be fun to play with |
19:31.39 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
19:31.42 | SteveTotaro | nxtvox is the one in china right? |
19:31.47 | CanWood | Our current system is analog in and the transtion will be smoother if we change one piece only |
19:31.47 | J4k3 | yeah |
19:31.49 | SteveTotaro | they mail from chine |
19:31.52 | SteveTotaro | china |
19:31.54 | J4k3 | so does half of ebay |
19:31.58 | J4k3 | so its ok. |
19:32.10 | Katty | i would like to hire an assassin. |
19:32.20 | Katty | do we have any ninjas? |
19:32.24 | SteveTotaro | i took the plunge from a different place |
19:32.36 | Katty | :< |
19:33.24 | stansmith | ecoterrorism? |
19:34.03 | SteveTotaro | www.getvoicecards.com |
19:34.20 | SteveTotaro | ebay worked out a few bucks cheaper |
19:34.37 | Katty | stansmith: that sounds like some sort of prehistoric dino. |
19:34.52 | *** join/#asterisk Servergod (n=servergo@70.97.159.120) |
19:35.03 | Katty | stansmith: ecoterroroserious. |
19:35.06 | Servergod | Hi all! |
19:35.17 | Katty | surious. |
19:35.25 | Katty | something like that. |
19:35.36 | x86 | I'm using MySQL for CDR, and I'm wondering what (if anything) uses the 'userfield' field, or if that's something I can put whate ever arbitrary value I want in? |
19:35.39 | budol | hi Katty |
19:35.43 | Servergod | phones are displaying caller id number twice and no name. Any ideas? |
19:35.53 | Katty | budol: allo. |
19:35.55 | SteveTotaro | <PROTECTED> |
19:35.55 | SteveTotaro | <PROTECTED> |
19:36.05 | Servergod | ie 702-xxx-xxx as number and 702xxxxxx as name |
19:36.06 | budol | Katty : how are you |
19:36.11 | *** join/#asterisk phillipk (n=phillipk@fw.datafax.net) |
19:36.14 | Katty | budol: digesting apple. |
19:36.23 | Katty | budol: that's all i'm doing today. |
19:36.42 | x86 | any ideas? |
19:36.44 | Katty | budol: and giving stansmith a lot of crap. |
19:36.52 | Katty | budol: no relation to apple digestion. |
19:36.53 | stansmith | :-( |
19:36.58 | stansmith | ew ha |
19:37.10 | SteveTotaro | stansmith sounds like a fedora guy |
19:37.15 | SteveTotaro | ~fedora |
19:37.16 | jbot | rumour has it, fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
19:37.21 | stansmith | wrong! |
19:37.24 | Katty | whenever someone says fedora, i think about eudora |
19:37.25 | stansmith | Arch |
19:37.27 | Katty | that old email client |
19:37.40 | Katty | i think it was eudora anyway ^_- |
19:37.46 | J4k3 | when people say 'fedora' I think of a big hat. |
19:37.47 | Katty | debian++ |
19:38.07 | Katty | what was mandrake an offshoot of? |
19:38.07 | SteveTotaro | i think of a gangster with a tommy gun |
19:38.07 | BCS-Satori | SteveTotaro: any experience with the OpenVox? |
19:38.08 | stansmith | LPCI > RHCP? |
19:38.11 | stansmith | debian i think |
19:38.23 | x86 | stansmith: LPIC* |
19:38.32 | SteveTotaro | nope, but the card is at my office, installing it tomorrow |
19:38.33 | J4k3 | well, linux is more like a whole family-sized bag of cheap wet noodles. |
19:38.36 | x86 | stansmith: Linux Professional Institute Certification |
19:38.38 | stansmith | oops, thanks x86 |
19:38.41 | Katty | ubuntu is kinda nice |
19:38.46 | x86 | stansmith: there are three of them, I have two ;) |
19:38.50 | Katty | all i remember about mandrake was all the Easy Wizards that Didn't Work |
19:39.05 | J4k3 | manrape |
19:39.07 | stansmith | i met one of the authors that published the oreilly book at linuxfest2007...great book |
19:39.09 | x86 | stansmith: two LPIC certs, CompTIA Linux+ cert, and a Brainbench Linux cert |
19:39.33 | stansmith | i want to get the level 1 cert, i think i can do it with studying very little...just havent gotten around to it |
19:39.34 | SteveTotaro | is linux + like A +? |
19:39.36 | x86 | I avoid distro-specific certs like the plague |
19:39.37 | Katty | where is Nugget when you need him! guess i'll fill in for him. |
19:39.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
19:39.41 | Katty | <Nugget> Linux is poo. |
19:39.44 | x86 | SteveTotaro: kinda |
19:39.44 | SteveTotaro | or network +? |
19:39.54 | x86 | SteveTotaro: it's linux+ :) |
19:40.02 | x86 | SteveTotaro: similar test setup |
19:40.10 | Katty | i did A+ for awhile |
19:40.16 | J4k3 | linux kernel is poo, the 23434534535 craptastic wheel-reinvention distros that run on top of it all suck furiously in their own little way |
19:40.21 | budol | Katty : lol |
19:40.21 | Katty | i decided it was redonkulus and i had better things to do :/ |
19:40.22 | SteveTotaro | VxWorks rocks!! |
19:40.32 | Katty | like chase the ferrets around the house |
19:40.32 | stansmith | so pretty much we all agree Arch is the best distro? great! |
19:40.43 | J4k3 | Microsoft, I'm sure, loves linux, since it just clogs up a few hundred thousand clued people into this constant wheel reinvention routine |
19:41.08 | J4k3 | without linux, microsoft would probably actually some day get competition |
19:41.22 | Katty | i must agree completely. |
19:41.27 | SteveTotaro | nah |
19:41.28 | J4k3 | but for now, everyones gonna waste time on garbage like... the uber-bloated linux kernel and the total waste of time xwindows is. |
19:41.40 | stansmith | J4k3 hates freedom...must be a terrorist..get him! |
19:41.56 | J4k3 | stansmith: eh? BSD license is a lot more free than the GPL |
19:42.00 | J4k3 | so don't do into that 'freedom' bullshit |
19:42.01 | stansmith | :-( |
19:42.14 | Servergod | can anyone help with caller id display? |
19:42.18 | SteveTotaro | 64 degrees out, crap, i gotta fire up my Kawasaki!!! |
19:42.20 | Katty | how fitting that freedom and bullshit be in the same sentance. |
19:42.32 | J4k3 | I can put a product on the market running a bsd 4.4-lite based OS and not releases sources and *gasp* not get my pants sued off. |
19:42.38 | Katty | SteveTotaro: eep. |
19:42.42 | tzafrir | Servergod, everyone is busy with more important flamefests |
19:42.48 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
19:42.51 | J4k3 | gpl-violations = why I'd NEVER EVER release a linux-running commercial product. |
19:42.57 | tzafrir | Servergod, or waiting for you to provide details |
19:43.02 | stansmith | lol |
19:43.15 | tzafrir | J4k3, ever heard of BSD violations? |
19:43.28 | Servergod | phones are displaying caller id number twice and no name |
19:43.28 | Tommy3 | is cisco 7940 a good phone for sip conversion? work with asterisk??? |
19:43.31 | Servergod | ie 702-xxx-xxx as number and 702xxxxxx as name |
19:43.33 | J4k3 | tzafrir: I'm sure somebody out there has violated the BSD license, but it'd be quite a bit harder. |
19:43.38 | J4k3 | ie - removing the copyright line |
19:43.48 | tzafrir | J4k3, AT&T lost a trial for BSD license violations |
19:43.50 | Servergod | i can see the name comming in properly from the provider |
19:43.56 | J4k3 | tzafrir: hence why I said bsd 4.4-lite |
19:43.59 | SteveTotaro | i have heard of S&M violations |
19:44.00 | *** join/#asterisk bkw___ (n=brian@adsl-70-234-182-211.dsl.tul2ok.sbcglobal.net) |
19:44.04 | Katty | bkw___: YOU |
19:44.13 | stansmith | J4k3: associating gpl with gnu/linux kind of a fallacy, no? |
19:44.14 | [TK]D-Fender | Tommy3: Where are you located? |
19:44.16 | Katty | bkw___: never say hi anymore :< |
19:44.28 | tzafrir | Most of the current GPL violations are very similar in nature: the distributors never bother to do the basic thing |
19:44.36 | *** part/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
19:44.36 | Katty | what kind of license thing does gentoo have? |
19:44.38 | Tommy3 | huntsville |
19:44.44 | SteveTotaro | servergod, go to www.voip-info.org |
19:44.55 | SteveTotaro | and search for set caller id |
19:44.56 | J4k3 | yes, because complying with the gpl leads to jerkoff kids creating warranty problems - see the wrt54g and why Linksys has done everything possible to avoid linux now. |
19:45.36 | [TK]D-Fender | Tommy3: Then forget Cisco and go Polycom. |
19:45.37 | SteveTotaro | the wrt54g running linux sold more of those units because of those jerkoff kids |
19:45.43 | *** join/#asterisk ManxPower (n=manxpowe@209.16.72.139) |
19:45.43 | J4k3 | I could, for example, base a product off an x86-running system running FreeBSD, carefully avoid GPL code, and never have to release a damned bit of source. |
19:46.01 | J4k3 | SteveTotaro: a few... but warranty hassles cost a lot more than the profit they got off the unit. |
19:46.26 | tzafrir | J4k3, and not use Asterisk, gcc, bash |
19:46.31 | J4k3 | so, in reality, those nerdy kids that returned hardware got in cisco's pocket, deep. |
19:46.35 | SteveTotaro | warranty is void if you load unofficial firmware |
19:46.54 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
19:46.56 | J4k3 | tzafrir: gcc is only required for compiling... compilers are for developers :) |
19:46.58 | stansmith | warranty is voided if u open the case...check the sticker underneath where the blue meets the black |
19:47.06 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
19:47.22 | Tommy3 | Why? problems with cisco phones (cheap on ebay) all I know about them.... |
19:47.24 | J4k3 | SteveTotaro: no, its not, at least in the USA. |
19:47.34 | SteveTotaro | yes it is |
19:47.52 | J4k3 | SteveTotaro: theres a federal act saying you can load any oil you want into your car, and any firmware you want onto your linkydink... its just a matter of if said oil or said firmware *breaks* the warrantied product is when the warranty is over. |
19:48.03 | tzafrir | J4k3, your loss. You avoid the pool of useful programs |
19:48.08 | [TK]D-Fender | Tommy3: Cisco's SIP implementation sucks and maintenance is a PITA |
19:48.20 | J4k3 | and its up to the manufacturer to prove your oil or your firmware broke the unit. |
19:48.28 | *** join/#asterisk pat2man (n=pat2man@ip67-90-247-203.z247-90-67.customer.algx.net) |
19:48.34 | J4k3 | in cisco's case that'd just add more costs, so they had to eat it |
19:48.43 | J4k3 | while they poked R&D to move to make the unit less linux-friendly. |
19:48.44 | SteveTotaro | According to Linksys, flashing firmware from sources other than Linksys does void the WRT54G's warranty. |
19:49.07 | J4k3 | SteveTotaro: yeah, and your dealership would like to lead you to believe changing your own oil will do the same |
19:49.14 | J4k3 | when in reality its quite the opposite. |
19:49.16 | stansmith | but so what? if u flash the firmware, why should linksys be responsible? |
19:49.24 | J4k3 | should they not be responsible? |
19:49.26 | SteveTotaro | they aren't |
19:49.27 | stansmith | no they shouldnt |
19:49.29 | J4k3 | if the hardware breaks, its their problem |
19:49.31 | stansmith | yea |
19:49.33 | J4k3 | if you break the unit with software, its your problem |
19:49.43 | stansmith | if i drive my car off a cliff its not hyundai's fault |
19:49.51 | SteveTotaro | if you break your hardware, it is your problem |
19:49.59 | stansmith | great, so we all came to a conclusion |
19:49.59 | J4k3 | yep, and its not hyundai's fault if you buy pennzoil thats actually maple syrup |
19:50.20 | stansmith | but that begs the question..was it ever penzoil to begin with? |
19:50.21 | J4k3 | but, alas, if you pour out their quaker state and add in mobil1 synthetic, you're not violating the warranty. |
19:50.27 | Tommy3 | Fender: OK, convinced... I was thinking about downloading the firmware and setting up a server to convert it, but you are right, it does take quite a bit of work according to what I have read. |
19:50.43 | SteveTotaro | j4k3, your argument is /dev/null |
19:50.47 | stansmith | LOL |
19:50.50 | J4k3 | SteveTotaro: no, its not. |
19:50.55 | stansmith | SteveTotaro: +2 xp |
19:51.10 | J4k3 | SteveTotaro: and its gone to court before... you've got desktop pc manufacturers not wanting to back hardware failures on PCs that people loaded linux on... its gone to court and the manufacturer lost. |
19:51.32 | SteveTotaro | totally different situation |
19:51.33 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
19:51.45 | [TK]D-Fender | Tommy3: What kind of prices were you finding? |
19:51.53 | J4k3 | not a different situation at al |
19:51.55 | J4k3 | er all |
19:52.00 | SteveTotaro | if i put maple syrup in my car does it void the warranty? |
19:52.00 | J4k3 | a PC and your router are the same idea |
19:52.05 | J4k3 | its a processor, some ram, some storage... |
19:52.16 | tzafrir | Can that WRT54G run Linux anyway? isn't ther a separate L model for that? |
19:52.26 | lmadsen | yes there is |
19:52.26 | stansmith | tzafrir: the older models can |
19:52.27 | J4k3 | yes, but only on parts directly affected by the maple syrup |
19:52.31 | lmadsen | or pre v.5 |
19:52.38 | SteveTotaro | i have several running busybox |
19:52.39 | J4k3 | they couldn't say, not cover the rear door hinge cuz you put maple syrup in the engine. |
19:52.42 | SteveTotaro | even asterisk |
19:52.43 | stansmith | after that, linksys started giving the newer models less vram |
19:52.43 | Tommy3 | Fender: $60 |
19:52.46 | SteveTotaro | openvpn |
19:53.05 | J4k3 | tzafrir: they'll all run linux, its just a matter of having any ram/flash left over :) |
19:53.21 | lmadsen | I have a WRT54GL running DD-WRT and have openvpn connected so all my phones and such can access the VPN |
19:53.28 | Tommy3 | Fender: (probably heavily used or stolen :) |
19:54.03 | [TK]D-Fender | Tommy3: Well..... hate to say it, but yeah, $60 is extremely ahrd to pass up and even I would probably go for it if I'm not concerned with warranty, etc... |
19:54.16 | J4k3 | linksys's big mistake on their product was crappy CFE, so when you did mis-flash the unit fell on its ass |
19:54.30 | J4k3 | toshiba, asus, buffalo... they didn't have that problem cuz their cfe wasn't shit |
19:54.32 | SteveTotaro | wrong again j4k3, some run VxWorks |
19:54.35 | [TK]D-Fender | Tommy3: But if we're talking new for business use, then I'd just say Polycom. |
19:54.40 | SteveTotaro | http://www.linuxdevices.com/news/NS4729641740.html |
19:54.50 | J4k3 | SteveTotaro: yes, and vxworks can be removed and linux put in their place |
19:55.14 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
19:55.23 | J4k3 | SteveTotaro: don't even need to jtag the unit... 100% network based bootloader replacement. |
19:55.37 | Tommy3 | Fender: Understand the business use.... I just wanted something to experiment with (cisco). I have NEC voip phones an service at my office. have not researched them for sip yet. |
19:55.48 | J4k3 | dd-wrt has instructions for the changeover, I think openwrt does too |
19:55.51 | ThatKidKel | i just noticed that my cdr-csv folder is filling up with a bunch of different files, each named after a particular accounting code |
19:55.57 | J4k3 | but you're stuck with the limitations of 8mb ram and 2mb flash |
19:55.59 | ThatKidKel | how can i stop that, and only log to Master.csv |
19:56.11 | SteveTotaro | yes but you said they all ran linux |
19:56.20 | J4k3 | dd-wrt does an amazing lot in 2/8, openwrt is a bit less tight. |
19:56.25 | SteveTotaro | can you ever admit you are wrong? |
19:56.29 | x86 | tzafrir: you can run linux on the WRT54G, the only difference with the WRT54GL model is more flash |
19:56.29 | J4k3 | SteveTotaro: they all *can* run linux. |
19:56.39 | J4k3 | and no, I never said they all ran linux out of the box... who cares if they do? |
19:56.45 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
19:56.53 | stansmith | u guys made the author leave |
19:57.05 | x86 | SteveTotaro: yeah, it's true... all WRT54G's can run Linux |
19:57.06 | SteveTotaro | (02:51:25 PM) J4k3: tzafrir: they'll all run linux, its just a matter of having any ram/flash left over :) |
19:57.19 | x86 | SteveTotaro: missed that part :) |
19:57.21 | J4k3 | SteveTotaro: you took the line out of context. |
19:57.23 | SteveTotaro | i know they can |
19:57.28 | steliosk | J4K3 : Actually its not the CFE that gets warped, but the area where the variables are stored |
19:57.28 | SteveTotaro | i guess i did |
19:57.43 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
19:57.45 | J4k3 | steliosk: yeah, and most manufacturers had the ability to clear/ignore that enough to tftpboot again |
19:57.51 | SteveTotaro | but if you read the article, they have a linux version for linux people with more ram |
19:58.09 | SteveTotaro | so your argument is/was /dev/null |
19:58.10 | J4k3 | if you can tftpboot, you can make a distro that runs 'mtd erase nvram' and get that whole problem fixed |
19:58.21 | x86 | SteveTotaro: more flash, iirc |
19:58.21 | J4k3 | now, asus's broke when you cleared nvram, but otherwise it was fairly bulletproof. |
19:58.28 | J4k3 | err no, toshiba's die when you clear nvram |
19:58.30 | J4k3 | asus's recover |
19:58.40 | J4k3 | linksys's recovers, but usually if the nvram is corrupt it won't ever tftpboot. |
19:58.49 | J4k3 | and tftpboot is disabled by default on the linksys, which is ghey |
19:58.55 | J4k3 | saves about 2 seconds of boot time, though. |
19:59.02 | SteveTotaro | that is why you enable it first |
19:59.12 | J4k3 | real hardware has it enabled already |
19:59.21 | J4k3 | hence the difference between linksys and the rest of the wrt-alikes. |
19:59.37 | J4k3 | (linksys sells the worst of the worst for the highest price, sans buffalo) |
19:59.43 | SteveTotaro | real hackers enable stuff that can help them from bricking stuff before the hack |
20:00.02 | J4k3 | of course, but some did it anyways since it *can help legitimate end users too* |
20:00.17 | J4k3 | real hackers with lots of time on their hands |
20:00.25 | J4k3 | 8mb over jtag = slow operation |
20:00.29 | J4k3 | or even 2mb |
20:00.44 | steliosk | well not the one you get for 2$ woth of parts |
20:00.59 | J4k3 | its still slower than just upgrading the sumbitch via network |
20:02.11 | stansmith | SteveTotaro: why did you think i was a fedora guy? |
20:02.27 | SteveTotaro | ~fedora |
20:02.28 | jbot | i guess fedora is stevetotaro is <action> tells stevetotaro that becoming RHCP certified, is The One True Way to Supreme Linux Knowledge |
20:02.33 | SteveTotaro | can't you read? |
20:02.57 | SteveTotaro | i actually am a fedora guy |
20:03.08 | SteveTotaro | fc9 beta |
20:03.22 | SteveTotaro | although i guess it is all beta |
20:03.39 | stansmith | arch is rolling release |
20:03.41 | dexpdx | is cdr_odbc supposed to connect and disconnect with every record |
20:04.38 | SteveTotaro | i don't much care what distro so long as i don't wind up in dependency hell |
20:05.09 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
20:05.28 | SteveTotaro | allright, if I don't get some motorcycle time in today, I will regret it, later everyone |
20:05.38 | stansmith | ~bye SteveTotaro |
20:05.39 | jbot | no |
20:05.42 | stansmith | wtf |
20:08.47 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
20:11.04 | *** join/#asterisk Idle_ (n=brian@S010600a024969312.ed.shawcable.net) |
20:11.23 | Idle_ | is there a way to create distinctive rings on asterisk with my wildcard? |
20:13.30 | stansmith | Idle: what kind of rings? |
20:15.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
20:17.12 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
20:17.12 | *** mode/#asterisk [+o angler] by ChanServ |
20:17.28 | b11d | ok.. I have this problem. I have a PRI going into a Sangoma A104d card.. when I enable the hardware EC, FAXing stops working, but echo is gone. When I disable hardware EC, FAXing works, but I get echo on voice calls. Any advice? |
20:19.04 | ManxPower | b11d: The EC should AUTOMATICALLY disable EC when it detects a fax tone because EC causes faxes to not work. This is the way every single EC on the planet should work this way. |
20:19.31 | b11d | it doesnt though.. |
20:19.38 | b11d | I have "faxdetect = both" on the PRI channels |
20:20.46 | b11d | i have "echocancel = yes" "echocancelwhenbridged = no" and "echotraining = no" |
20:20.48 | b11d | on those channels as well |
20:21.19 | b11d | should i pastebin my zapata? |
20:21.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
20:24.49 | tzafrir | b11d, the echo cancelling code in zaptel has an independent fax tone detection that should disable the EC |
20:24.56 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
20:25.12 | tzafrir | Regardless of the asterisk chan_zap faxdetect |
20:25.58 | kraypius | <PROTECTED> |
20:26.10 | kraypius | and i get a busy signal |
20:26.11 | tzafrir | b11d, but if the channel is dedicated for faxing, just disable EC |
20:26.46 | ManxPower | kraypius: I'm sorry, but we do not support Amp, FreePBX, Asterisk-GUI, or other GUIs here. |
20:27.02 | b11d | well it passes a fax down the next available channel |
20:27.09 | b11d | so I never know what one will be fax and what one will be voice |
20:27.10 | b11d | its random |
20:27.58 | ManxPower | b11d: you need to contact Sangoma to find out why their EC is not being autpmatically disabled |
20:28.27 | b11d | zap show channel shows it being disabled though.. |
20:28.32 | b11d | yet it doesnt |
20:28.38 | b11d | so maybe it actually isnt being shut off |
20:29.25 | b11d | http://www.pastebin.ca/926725 |
20:29.37 | b11d | thats a sample of the zapata.conf -- make sure im not on crack and missing something stupid please :) |
20:29.50 | tzafrir | b11d, what is the complete line you see there? How many taps? |
20:30.06 | b11d | 128 taps or something similar.. |
20:30.08 | b11d | one sec |
20:30.44 | b11d | Echo Cancellation: 128 Taps unless TDM Bridged, currently OFF |
20:30.51 | tzafrir | So there is currently no active EC, because there is no call. But the channel is configured to use an EC |
20:31.09 | b11d | one shows "ON" right now |
20:31.45 | b11d | so I should be enabling hardware EC, but setting 'echocancel = no' in zapata? |
20:32.56 | b11d | all I want is EC for voice and no EC for fax ;) |
20:33.20 | dexpdx | anyone know the best way to trace a cdr_odbc error |
20:33.34 | [TK]D-Fender | b11d: "echocancel=yes" and it should deactivate automatically |
20:33.53 | b11d | well.. I have echocancel = yes now.. |
20:34.00 | b11d | and when I enable hardware EC, faxing stops! |
20:34.06 | vap0rtranz | why are bloody inbound cids being matched by the last peer entry?! |
20:34.27 | ManxPower | b11d: you ALREADY have your answer. |
20:34.35 | vap0rtranz | eh? |
20:34.42 | b11d | i guess im misreading.. im stressed out today :) |
20:34.56 | dexpdx | vap0rtranz: dialplan fallthrough? |
20:35.11 | vap0rtranz | dexpdx: really? i'll disable and test ... |
20:35.37 | dexpdx | vap0rtranz: IAX peers? |
20:35.41 | b11d | I need to contact Sangoma then you figure? |
20:35.53 | vap0rtranz | dexpdx: all sip; sorry didn't clarify that |
20:36.52 | dexpdx | vap0rtranz: exactly what is going wrong |
20:37.26 | vap0rtranz | seems to be a dialplan thing. the global context is incoming, but the only inbound cid that doesn't get rejected is whatever trunk (peer) is listed last in users.conf |
20:37.32 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
20:38.46 | dexpdx | what is the source of the inbound calls? |
20:39.28 | vap0rtranz | dexpdx: you mean has is registered upstream correctly?? |
20:40.14 | dexpdx | vap0rtranz: let me get this right: you have a bunch of sip peers, correct? |
20:41.19 | *** join/#asterisk zobia (n=laurashr@222.212.67.156) |
20:41.26 | zobia | Hello everyone |
20:41.45 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
20:42.02 | vap0rtranz | dexpdx: a dozen or so in users.conf. i can always get which one is last to answer the incoming did correctly. all of them will do this if they're last in the line. i'm matching the incoming cid in the dialplan b/c we have to do different things to different incoming calls. it really seems like a dialplan problem but i can get each line to work if they're last in the users.conf ... :( |
20:42.06 | zobia | i want to change the voicemail's filename. anyone knows how can i change this. or make a copy when generate the voicemail file |
20:42.40 | zobia | i have read the app_voicemail.c but don't know where to change. and i want to use the callerid as the filename. |
20:42.49 | dexpdx | are the incoming calls are coming over what kind of trunk: sip, iax, zap? |
20:43.04 | dexpdx | is that trunk's default context different? |
20:43.15 | vap0rtranz | dexpdx: all sip, that's what i meant before |
20:43.28 | dexpdx | what is the type of error that the other sip peers give when issueing a dial? |
20:43.31 | vap0rtranz | dexpdx: yes they are! |
20:44.06 | vap0rtranz | dexpdx: i was just about to cram the trunks into one context ... *crosses fingers* |
20:44.42 | dexpdx | what is the failure message when you try and send a call to one of the SIP peers |
20:44.51 | docelmo | Anyone in here in the UK? |
20:45.03 | vap0rtranz | dexpdx: no sending. this is all incoming |
20:45.42 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
20:46.04 | dexpdx | vap0rtranz: when you have one call coming in from one sip peers and you want to route it to another you generally issue a Dial(SIP/${INSERT_SIP_EXTEN_HERE}/${EXTEN}) |
20:47.04 | vap0rtranz | dexpdx: i'm confused. this is for inbound processing by asterisk from a upstream registrar. dialing out works. |
20:47.13 | mvanbaak | does anyone know if * can do videoconferencing ? |
20:47.38 | dexpdx | vap0rtranz: paste your extensions.conf to pastebin and give the link |
20:49.37 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
20:50.20 | *** join/#asterisk RoyK (n=roy@ip-122-26-149-91.dialup.ice.no) |
20:51.27 | vap0rtranz | dexpdx: we've got alot of internal numbers in there. not sure how the boss would feel. anyways, it really sounds like your troubleshooting the trunk dialing outbound which is not the problem here. it's incoming call processing. i have _[DID] matching each incoming # in extensions.conf under an [incoming] context which is the default; asterisk should just match that and continue processing (to a menu) |
20:52.05 | *** join/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
20:52.44 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
20:52.54 | stansmith | i love it when a plan comes together! |
20:53.32 | *** join/#asterisk egecko (n=sam@cpe-76-176-238-225.san.res.rr.com) |
20:53.50 | dexpdx | vap0rtranz: I don't know how you end indent on a sending a call from one peer to another with out having a Dial cmd |
20:53.59 | draygon | is there a good doc on installing asterisks as a non-root user on CentOS? |
20:54.29 | dexpdx | draygon: I don't think you will have any problems as long as you don't bind to any ports lower than 1024 |
20:54.42 | dexpdx | or whatever the system port cutt off |
20:54.49 | stansmith | draygon: the book goes through the installation using centos |
20:54.50 | stansmith | ~book |
20:54.51 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
20:55.36 | vap0rtranz | dexpdx: ah, i think i see where you're going. i'm not at the provider end right now. i'm at the far end where the cid of the did (or whatever) has already been sent downstream. i need asterisk to correctly process the cid that has already been sent |
20:58.01 | dexpdx | vap0rtranz: your not making sense |
20:58.17 | vap0rtranz | dexpdx: i know nothing |
20:58.26 | dexpdx | vap0rtranz: then go read |
20:58.30 | dexpdx | voip-info.org |
20:58.35 | vap0rtranz | ok, so where does asterisk put an incoming call? |
20:58.43 | dexpdx | where ever your tell it |
20:58.54 | vap0rtranz | the context in sip.conf is default, right? |
20:59.03 | dexpdx | extensions.conf is where you configure your dialplan contexts |
20:59.33 | dexpdx | vap0rtranz: should be 'default' but I always use a custom context for inbound traffic per "peer" |
20:59.56 | dexpdx | exten => 5552221212,1,Dial(SIP/1212/${EXTEN}); |
21:00.02 | vap0rtranz | dexpdx: good, that's how this was setup; for each peer, there was its own context |
21:00.09 | vap0rtranz | dexpdx: precisely |
21:00.11 | dexpdx | would be an example of how to route calls coming in on 5552221212 to exten 1212 |
21:00.29 | vap0rtranz | dexpdx: only diff is i'm Goto (something else) |
21:00.54 | dexpdx | so you issue a Goto to jump to another context? |
21:01.11 | vap0rtranz | dexpdx: the problem is asterisk doesn't get as far as 5552221212, even with _5552221212; it's always "extension rejected" |
21:01.29 | *** join/#asterisk iamthelostboy (n=nathan@125-236-212-46.adsl.xtra.co.nz) |
21:01.45 | dexpdx | well maybe you are not matching the inbound number correctly |
21:02.35 | vap0rtranz | dexpdx: true, what i had thought. but the error message is explicit. the number i dailed for inbound is incorrectly seen as coming from the peer that's listed last in users.conf *ACK* |
21:02.44 | dexpdx | try doing a Verbose('DEBUG: number is matching') instead of a Dial or what cmd and if you don't see the msg in the cli then you are doing something wrong |
21:03.24 | *** join/#asterisk NirS (n=NirS@87.68.17.141.cable.012.net.il) |
21:03.24 | dexpdx | so the inbound number is falling into the the wrong context? |
21:03.52 | vap0rtranz | dexpdx: weird thing is asterisk knows what extension i'm trying to send the inbound call to. so it goes: "Call from '[last-peer-in-users.conf]' to extension '[correct-inbound-cid]' rejected because extension not found |
21:04.20 | NirS | g'day all |
21:04.25 | NirS | actually, g'night all |
21:04.26 | NirS | :) |
21:04.27 | vap0rtranz | dexpdx: [last-peer-in-users.conf] should be the same as [correct-inbound-cid]? |
21:07.05 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
21:07.18 | dexpdx | users.conf not where you specify dialplan contexts |
21:07.39 | *** join/#asterisk enjay5150 (n=chatzill@ip70-190-63-195.ph.ph.cox.net) |
21:08.22 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
21:08.32 | stansmith | vap0rtranz: i love it |
21:08.34 | vap0rtranz | dexpdx: right, i've only specified the context for the trunks there. i'm assuming that context is for inbound & outbound dialing? b/c it's type = peer |
21:08.47 | vap0rtranz | stansmith: eh? |
21:08.56 | *** join/#asterisk lirakis (i=lirakis@66.252.24.133) |
21:09.07 | *** join/#asterisk jlb (n=jlb@75.148.162.90) |
21:09.13 | lirakis | anyone know a good open source sip stack / library |
21:09.23 | dexpdx | vap0rtranz: rtfm |
21:09.31 | stansmith | lol |
21:09.43 | *** join/#asterisk Tebi (n=tebi@gw.aller.fi) |
21:09.44 | *** join/#asterisk newmember (n=chatzill@static-66-11-81-65.ptr.terago.ca) |
21:09.50 | *** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com) |
21:10.05 | vap0rtranz | whatever. it works by setting the context in users.conf to default and the default context in sip.conf to default |
21:10.08 | vap0rtranz | thanks anyways |
21:10.12 | shmaltz | who overher is using switchvox? |
21:10.17 | shmaltz | ~switchvox |
21:10.28 | newmember | Does a Cisco 7960 phone need a TFTP server or can I just set the SIP proxy manually on the phone? |
21:10.37 | shmaltz | ~time |
21:10.38 | jbot | i heard time is 1 dimensional, or everlasting, an illusion, or 2008.03.03 21:10:38 GMT |
21:10.45 | shmaltz | ~swithcvox |
21:11.00 | shmaltz | ~switchvox |
21:11.07 | lirakis | shmaltz: its not there! |
21:11.14 | jlb | The voip-info wiki claims that you can only store static config OR realtime config in a database, but not both. Anyone know if this still true in 1.6 (or even 1.4) ? |
21:11.15 | shmaltz | ok, |
21:11.36 | shmaltz | jlb, why use realtime? |
21:12.06 | lirakis | shmaltz: because changes are dynamic ? |
21:12.18 | shmaltz | lirakis, what type of changes? |
21:12.25 | lirakis | shmaltz: .. or rather, the uptake is dynamic |
21:12.28 | *** join/#asterisk DaleG (n=dale@hlfxns0149w-142177089010.ns.aliant.net) |
21:12.38 | shmaltz | lirakis, what type of changes? |
21:13.25 | jlb | shmaltz: because users/etc. are in the database |
21:13.40 | shmaltz | jlb, why not change the text files and issue a reload? |
21:14.18 | jlb | shmaltz: because asterisk isn't the only consumer of the data |
21:14.27 | lirakis | shmaltz: peers users queues, etc... read the webpage |
21:14.52 | shmaltz | jlb, so why not do both? duplicate it to a database as well and have a deamon update the config files, I know Verizon does it that way |
21:15.00 | stansmith | redundant |
21:15.17 | jlb | because that's a terrible architecture, isn't sensitive to frequent updating, etc. |
21:15.35 | shmaltz | jlb, really? how so? |
21:15.47 | shmaltz | jlb, if Verizon can do it so can you |
21:16.31 | stansmith | jlb: couldnt you use DB as master record, and use a lil perl or some other script magic to query results into text file? |
21:16.41 | stansmith | cron it up to run whenever |
21:16.41 | *** join/#asterisk Dovid (n=Dovid@bzq-79-180-58-141.red.bezeqint.net) |
21:16.42 | stansmith | wammy |
21:16.45 | stansmith | all updates go to DB and we are all happy |
21:16.47 | jlb | well if I have to pick a file to update and reload, I would pick the one that changes infrequently (sip.conf) and not the realtime user data |
21:16.54 | DaleG | hey, I created a small prog that's a web-server to provision IAXy's (for Linux/OS X, others). Point your browser at it, fill in the fields, click provision, and it does it's thing. Looking for people to test it out... |
21:16.54 | Dovid | hi. i am running tests with a new carrier: which one of these is for Xeon ? |
21:16.55 | Dovid | http://asterisk.hosting.lv/#bin |
21:17.46 | jlb | I can definitely do something that will work... I was mostly wondering if the odd mutually-exclusive behavior of static and realtime config had gotten changed/fixed recently |
21:18.24 | shmaltz | anyone here using switchvox |
21:18.26 | shmaltz | ? |
21:18.37 | lirakis | shmaltz: dude... come on |
21:18.40 | shmaltz | #switchvox is as dead as they come |
21:18.48 | shmaltz | lirakis, comeon what? |
21:19.15 | stansmith | @#$@~! |
21:19.53 | shmaltz | anyone here using switchvox? |
21:20.36 | lirakis | shmaltz: are you a robot? |
21:20.41 | stansmith | lol |
21:21.22 | lirakis | shmaltz: check for a robotic brain too |
21:21.28 | stansmith | shmaltz: i wasnt cursing at u, i was cursing at the fact that ${CHANNEL(channeltype)} only displays "Zap" and not the exact line |
21:22.09 | [TK]D-Fender | stansmith: Its only supposed to give you the TYPE, just like its name implies |
21:22.33 | shmaltz | stansmith, ${CHANNEL} should have that |
21:22.40 | shmaltz | I guess I'm wrong for 1.4 |
21:23.02 | shmaltz | anyone here using switchvox? |
21:23.27 | stansmith | i typed "core show [tab twice]" and didnt see variables.. i was hoping there was something inside asterisk that would allow me to see all variables available, is this feasible? |
21:23.50 | [TK]D-Fender | shmaltz: Ask a few more times, its not like you're wasting your breath. |
21:23.58 | lirakis | shmaltz: i use swithvox |
21:24.12 | lirakis | shmaltz: .. oh.. wait.. nevermind |
21:24.18 | [TK]D-Fender | stansmith: "see variables |
21:24.20 | [TK]D-Fender | "? |
21:24.21 | shmaltz | lirakis, are you? |
21:24.25 | lirakis | shmaltz: you almost had me fooled last time you asked |
21:24.41 | stansmith | no such command |
21:25.00 | [TK]D-Fender | stansmith: that was a QUESTION. What do you mean "see variables"? |
21:25.09 | JenniferAkemi | if i type core show version and it says 1.4.18 does that mean i'm not running 1.4, but am running the trunk version or something? |
21:25.14 | tzafrir | stansmith, core show function <tab><tab>? |
21:25.24 | stansmith | yea |
21:25.26 | [TK]D-Fender | JenniferAkemi: 1.4 is a series. |
21:25.27 | stansmith | like in the bash shell |
21:25.37 | stansmith | [TK]D-Fender: yes |
21:25.42 | stansmith | like how u can see all the functions |
21:25.51 | [TK]D-Fender | stansmith: "core show functions" |
21:25.56 | stansmith | no i know that |
21:26.17 | stansmith | i was hoping/wondering there was a way to see variables? rather than looking at http://www.voip-info.org/wiki-Asterisk+variables |
21:26.22 | stansmith | which is half-deprecated |
21:26.36 | stansmith | as in.. core show variables? |
21:26.44 | mvanbaak | I think there's a txt file in the source that list variables |
21:26.44 | tzafrir | stansmith, in 1.4 you can see globals |
21:26.50 | stansmith | ah yes |
21:27.01 | tzafrir | in 1.6 you can also see channel-specific variables |
21:27.08 | tzafrir | see and set |
21:27.33 | newmember | Can I run a Cisco 7960 without a TFTP server and just manually add a proxy? |
21:27.44 | [TK]D-Fender | stansmith: On moment you are asking about variables, the next you're asking about functions. Make up your mind. |
21:27.53 | JenniferAkemi | [TK]D-Fender: the reason i asked is cuz i got the g729a codec, and it says that it was compiled against an older version of asterisk and may cause instability, so i was wondering if i needd to download the one from the unsupported directory (the trunk one) |
21:28.18 | JenniferAkemi | [TK]D-Fender: I think i might have done an svn asterisk at some point trying to get something else to work so i'm not sure if i'm running something newer than 1.4 or not |
21:28.19 | stansmith | [TK]D-Fender: i never asked about functions...i was asking if there is a way to view available variables..similiar to how you view functions ( "core show functions" ) |
21:28.33 | vap0rtranz | what's an * looping function? something like do .. until |
21:28.36 | stansmith | JenniferAkemi: if you are running 1.4.18, u are running 1.4 |
21:28.41 | [TK]D-Fender | stansmith: "core show channel [channel]" |
21:28.54 | lirakis | newmember: http://phoenix.labri.fr/documentation/sip/Documentation/Material/Clients/Hardphone/Cisco/Admin/AdminGuide-us.pdf |
21:28.59 | tzanger | [TK]D-Fender: I get an error "[channel] not found" :-p |
21:29.17 | [TK]D-Fender | tzanger: Smart-ass :p |
21:29.20 | JenniferAkemi | ok. how come i get the message about g729a being compile against an older version of asterisk? |
21:29.22 | newmember | lirakis: ty I will take a look |
21:29.27 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
21:29.31 | lirakis | newmember: next time ask google first |
21:29.58 | newmember | lirakis: you are assuming I haven't |
21:30.00 | JenniferAkemi | vap0rtranz: probably while. |
21:30.11 | lirakis | newmember: yeah.. b/c that took me like 3 seconds to get on google |
21:30.19 | JenniferAkemi | vap0rtranz: there was an app_while or something i noticed in modules.conf |
21:31.03 | stansmith | [TK]D-Fender: rest assured..the info you pass onto me, i pass onto others when possible, we all appreciate it :) |
21:32.16 | MatBoy | someone good experience with the TE-410P cards ? |
21:32.18 | vap0rtranz | JenniferAkemi: oh wow. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+While? that is very cool. |
21:33.36 | stansmith | MatBoy: ask a more specific question |
21:33.48 | MatBoy | stansmith, stability |
21:33.59 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
21:34.06 | JT | pretty specific... |
21:34.06 | vap0rtranz | JenniferAkemi: the book had some silly GoTo statements. it was like a blast from the past ... BASIC that is :P |
21:34.44 | MatBoy | stansmith, I can buy 2 for $500,- |
21:35.21 | JenniferAkemi | vap0rtranz: glad to help |
21:35.33 | MatBoy | new ones actually |
21:35.44 | JenniferAkemi | How do i know if my g729a is still registered |
21:35.53 | stansmith | legally? |
21:36.15 | JenniferAkemi | yeah |
21:40.25 | *** join/#asterisk Nasra (n=maxshipp@190.166.70.107) |
21:40.57 | *** part/#asterisk lirakis (i=lirakis@66.252.24.133) |
21:41.00 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
21:42.41 | Nasra | SteveTatero |
21:42.49 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
21:43.11 | MatBoy | yep, legally |
21:43.41 | J4k3 | since when did you have to register your g729a installation? |
21:43.41 | Nasra | Is that you? |
21:43.56 | J4k3 | afaik all you needed was enough licenses to cover specific usage |
21:44.16 | J4k3 | it'd also help greatly if digium's g729 versions actually worked. |
21:44.28 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
21:45.05 | Nasra | I am in the process of installing a small server... |
21:45.19 | Nasra | with linux |
21:45.29 | Nasra | debian |
21:45.37 | DaleG | Any people out there have any large installations of IAXy's? |
21:45.46 | *** join/#asterisk SparFux (n=raoul@e182030185.adsl.alicedsl.de) |
21:46.55 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
21:47.04 | *** join/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
21:48.11 | adeel | you know, if your end point & sip provider support g.729, you don't need a license for 'pass-through' mode |
21:48.19 | *** join/#asterisk angom (n=angom@200.79.141.128.dsl.dyn.telnor.net) |
21:48.26 | stansmith | the answer to the question i had a little bit ago was ${CDR(channel)} ... ${CDR(channel)} displays whatever line is being used..in my case it is = to "Zap/4-1" |
21:48.30 | JenniferAkemi | well i bought g729a licenses |
21:48.35 | stansmith | just in case anyone else has trouble with that later on ^^^ |
21:48.39 | JenniferAkemi | but now my g729 function thing is gone from the CLI |
21:48.41 | SparFux | Licenses? |
21:49.08 | adeel | JenniferAkemi, do you still have the codec loaded? |
21:49.12 | *** part/#asterisk stansmith (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
21:49.15 | JenniferAkemi | how do i tell adeel |
21:50.52 | adeel | JenniferAkemi, show modules should print out the loaded modules |
21:51.14 | adeel | JenniferAkemi, should start with codec_ and have g729 somewhere in it |
21:51.38 | JenniferAkemi | yeah i have codec_g729a.so |
21:52.49 | *** join/#asterisk Servergod (n=servergo@70.97.159.120) |
21:53.19 | Servergod | anyone ever have a cisco186 only ring twice then give a 480? |
21:53.43 | enjay5150 | JenniferAkemi: have you restarted since you registered/installed? |
21:53.49 | adeel | JenniferAkemi, hmmm...well then you should have access to your g729 |
21:54.38 | *** join/#asterisk bkw__ (n=brian@adsl-76-196-203-125.dsl.tul2ok.sbcglobal.net) |
21:54.39 | SparFux | License? I have no one , but /usr/lib/asterisk/modules/format_g729.so |
21:54.41 | JenniferAkemi | yes i restarted many times since i registered |
21:55.04 | JenniferAkemi | but since then i've been doing stuff in modules.conf |
21:55.21 | adeel | JenniferAkemi, can you try doing a 'module reload codec_g729a.so' |
21:55.44 | JenniferAkemi | codec_g720a.so does not support reload |
21:56.09 | JenniferAkemi | sorry typo, codec_g729a.so does not support reload |
21:56.10 | adeel | JenniferAkemi, so do a 'module unload codec_g729a.so' followed by 'module load codec_g729a.so' |
21:56.12 | SparFux | first unload then load |
21:57.17 | JenniferAkemi | i get unregistered translator lintog720 lines, does that mean it's not registered? |
21:57.43 | adeel | JenniferAkemi, if you get that while unloading, then yes, it has removed it |
21:57.44 | Dovid | hi. i am running tests with a new carrier: which one of these is for Xeon ? |
21:57.45 | Dovid | http://asterisk.hosting.lv/#bin |
21:57.57 | JenniferAkemi | when i load it |
21:57.59 | JenniferAkemi | oh wait |
21:58.04 | JenniferAkemi | i think we're good |
21:58.15 | JenniferAkemi | it says it found license, and how many there are |
21:58.40 | JenniferAkemi | except i still don't have the g729 cli command |
21:58.53 | adeel | JenniferAkemi, type in 'help' and see if anything shows up |
21:59.44 | JenniferAkemi | yes, i tried that, but no it isn't listed |
22:00.01 | JenniferAkemi | nm |
22:00.15 | JenniferAkemi | i was confused. it was show g729 that i type, not g729 show |
22:00.22 | JenniferAkemi | thanks for your help adeel |
22:00.26 | adeel | JenniferAkemi, no problems |
22:00.47 | Servergod | anyone ever have a cisco186 only ring twice then give a 480? |
22:00.55 | *** part/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
22:01.01 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
22:01.09 | enjay5150 | servergod, you have a timeout? |
22:02.44 | Servergod | http://pastebin.com/m79f86ac0 |
22:02.57 | Servergod | not that i can see |
22:03.11 | Servergod | it just sems to do this with the cisco ata186/8 |
22:03.41 | Servergod | lemme do one w sip debug |
22:03.55 | dexpdx | Ok, I'm at my witts end with this cdr_odbc thing |
22:04.48 | dexpdx | [Mar 3 14:03:39] ERROR[14748]: cdr_odbc.c:358 odbc_load_module: cdr_odbc: Unable to connect to datasource: cdr |
22:04.56 | dexpdx | isql f'ing works! |
22:04.58 | dexpdx | wtf |
22:06.57 | JT | dexpdx: why are you telling us? |
22:07.04 | dexpdx | I need some help |
22:07.18 | JT | dexpdx: how can anyone help you with so little info? |
22:07.19 | adeel | dexpdx, can you connect to your odbc from the console, outside of asterisk? |
22:07.31 | dexpdx | adeel: yum isql works perfect inserts/selects etc |
22:07.40 | dexpdx | s/yum/yup |
22:07.52 | dexpdx | JT: that's all the info I get |
22:07.57 | JT | rubbish |
22:07.59 | JT | you have configs |
22:08.03 | dexpdx | with debug and verbose set higher than 10 |
22:08.03 | JT | pastebin them |
22:08.07 | dexpdx | JT: sure |
22:08.08 | dexpdx | np |
22:08.09 | dexpdx | one sec |
22:08.51 | adeel | dexpdx, okay, so you can do a direct isql, have you tried testing your odbc config? |
22:09.26 | dexpdx | adeel: which odbc config - asterisks |
22:10.32 | vap0rtranz | anyway to get the output of reloads to a text file w/out enabling mega-verbose debugging? asterisk doesn't pipe the info to stdout from -rx :( |
22:10.55 | lanning | screen |
22:11.20 | vap0rtranz | lanning: heh. besides the screen |
22:11.38 | dexpdx | JT: http://pastebin.ca/926909 |
22:11.45 | lanning | not "the screen" |
22:11.49 | lanning | "screen" |
22:12.26 | lanning | http://www.gnu.org/software/screen/ |
22:12.43 | dexpdx | or 2>&1 |
22:12.48 | dexpdx | ie redirect STDERR to stdout |
22:13.41 | vap0rtranz | dexpdx: doesn't play nice with sudo :) |
22:13.51 | vap0rtranz | lanning: awesome, already installed too |
22:13.56 | dexpdx | asterisk -rx 'module reload' 2>&1 |
22:14.02 | dexpdx | vap0rtranz: the hell it doesn't |
22:14.39 | dexpdx | [jason@pstn-gw1 ~]$ sudo /usr/sbin/asterisk -rx 'exit' 2>&1 >foo |
22:14.40 | dexpdx | [jason@pstn-gw1 ~]$ cat foo |
22:14.40 | dexpdx | No such command 'exit' (type 'help' for help) |
22:14.46 | dexpdx | proof right there |
22:15.16 | vap0rtranz | dexpdx: screen is good. had forgotten about it. |
22:15.21 | lanning | swap them |
22:15.42 | lanning | sudo /usr/sbin/asterisk -rx 'exit' > foo 2>&1 |
22:16.02 | lanning | redirect STDOUT, then copy the new STDOUT to STDERR. |
22:16.03 | ManxPower | lanning: too bad that won't work |
22:16.23 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
22:16.39 | ManxPower | exit is NOT valid in an -rx |
22:16.49 | x86 | stop now is ;) |
22:16.50 | dexpdx | ManxPower: I know |
22:16.52 | SparFux | lanning: first of all, you need quote with sudo, to have user id still with stuff after > |
22:16.53 | ManxPower | "stop now" would be what you want. |
22:16.53 | dexpdx | ;) |
22:16.54 | vap0rtranz | ManxPower: thank you. |
22:16.57 | dexpdx | No such command |
22:17.08 | dexpdx | ManxPower: I didn't want to stop my instance of asterisk ;) |
22:17.13 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
22:17.22 | x86 | dexpdx: just run asterisk from inittab ;) |
22:17.42 | ManxPower | dexpdx: what exactly DO you want to do? |
22:17.43 | lanning | the error (no such command) should goto the file (foo), unless asterisk is writing to /dev/tty. |
22:18.00 | x86 | inittab+asterisk==lovinit |
22:18.07 | dexpdx | ManxPower: I was showing vap0rtranz that you can redirect the output of -x to a file with sudo |
22:18.15 | ManxPower | Ah. |
22:18.21 | ManxPower | poor thing didn't even know that? |
22:18.23 | SparFux | from inittab? |
22:18.40 | Katty | o hai! |
22:18.47 | dexpdx | x86: you go ahead and drop what ever calls you want on any of your instances just to should the output of a command |
22:18.58 | dexpdx | s/should/show |
22:19.52 | x86 | dexpdx: inittab doesn't drop calls |
22:19.53 | vap0rtranz | ManxPower: i did. that hasn't worked. but screen looks to be good |
22:20.06 | x86 | dexpdx: inittab just assures that if asterisk dies, it's immediately restarted |
22:20.28 | x86 | dexpdx: if you want to run commands, use AMI |
22:20.49 | dexpdx | x86: so if I issued a 'stop now' in the CLI because asterisk gets restarted under inittab it wouldn't drop any calls |
22:20.53 | dexpdx | ? |
22:20.56 | dexpdx | *cough* |
22:21.24 | dexpdx | I don't use inittab anyways I just use safe_asterisk |
22:21.25 | x86 | dexpdx: those were two different things |
22:21.39 | x86 | dexpdx: of course you use inittab... unless you use upstart |
22:22.07 | dexpdx | x86: of course I use inittab but I don't start asterisk directly from it |
22:22.08 | dexpdx | err |
22:22.16 | x86 | you should, is what I'm saying ;) |
22:22.20 | dexpdx | why |
22:22.24 | lanning | x86, the issue was not "restarting asterisk" it was "demo a command output redirection, without any restart" |
22:22.26 | dexpdx | don't need to |
22:22.31 | x86 | so if it dies (which happens from time to time), it will automagically get restarted |
22:22.41 | x86 | lanning: gotcha |
22:22.44 | dexpdx | x86: just like safe_asterisk restarts it |
22:22.56 | x86 | dexpdx: except inittab is better |
22:22.59 | vap0rtranz | dexpdx: did you actually try redirection of reload? it doesn't even work as root |
22:23.13 | dexpdx | except not |
22:23.19 | x86 | dexpdx: sure it does |
22:23.23 | dexpdx | vap0rtranz: yes, I have done it |
22:23.24 | vap0rtranz | exit works |
22:23.24 | x86 | s/does/is/ |
22:23.30 | *** join/#asterisk mosty (n=mostyn@ppp191-34.static.internode.on.net) |
22:23.36 | dexpdx | x86: why? |
22:23.39 | x86 | dexpdx: use AMI to capture the output of a command |
22:23.49 | x86 | dexpdx: because what happens if safe_asterisk dies? |
22:24.11 | x86 | dexpdx: have you ever seen init die on a box? |
22:24.17 | lanning | yes |
22:24.17 | dexpdx | x86: yes |
22:24.19 | dexpdx | ;) |
22:24.40 | dexpdx | x86: I don't make a habbit of putting my auto-restarting processes in inittab |
22:25.14 | x86 | dexpdx: scrap safe_asterisk for asterisk+inittab |
22:25.32 | dexpdx | x86: how often do shell scripts simple die on your systems? |
22:25.32 | x86 | dexpdx: i guess if you're just playing around, it doesn't really matter |
22:26.01 | lanning | x86, you are barking up the "personal preference" tree... |
22:26.03 | dexpdx | I also prefer not to have to edit inittab and -HUP everytime I want asterisk to "stay off" |
22:26.04 | x86 | dexpdx: it's plausible if oom_killer decides to chomp it :) |
22:26.11 | lanning | use "vi" not "emacs" |
22:26.19 | x86 | lanning: nope... oom_killer wont nuke init... period |
22:26.19 | vap0rtranz | lanning: *applauds* |
22:26.31 | x86 | lanning: no, i have reasoning |
22:26.38 | x86 | not just personal preference |
22:26.55 | lanning | either way works. |
22:26.59 | x86 | sure |
22:27.03 | x86 | one works better though ;) |
22:27.06 | lanning | enough said |
22:27.18 | vap0rtranz | x86: oh bother |
22:27.18 | x86 | *nods* |
22:28.10 | dexpdx | you do not run trivial restartable processes from inittab and you do not run programs that kill non trivial programs |
22:28.19 | dexpdx | programs/services/processes |
22:28.51 | *** part/#asterisk ice_croft (n=nolan@213.132.86.246) |
22:28.51 | dexpdx | i.e if asterisk is important enough to run from inittab your don't run a program that will kill it |
22:28.57 | *** join/#asterisk nirz (n=nir@89-138-78-170.bb.netvision.net.il) |
22:29.02 | x86 | dexpdx: i never said to |
22:29.02 | dexpdx | that will - that could |
22:29.18 | dexpdx | then why are you concerned that oom_killer will kill it? |
22:29.19 | x86 | dexpdx: from a high-availability standpoint, inittab makes sense |
22:29.29 | x86 | dexpdx: oom_killer is part of the linux kernel |
22:29.41 | dexpdx | x86: it's a matter of taste - I don't like running things from inittab |
22:30.06 | x86 | dexpdx: then dont :) |
22:30.20 | dexpdx | I also don't run other services on my asterisk machines ;) |
22:30.30 | x86 | me either |
22:30.32 | vap0rtranz | dexpdx: not even ntp! |
22:30.34 | vap0rtranz | *gasp* |
22:30.38 | dexpdx | client |
22:30.39 | dexpdx | ;) |
22:30.52 | x86 | well, I run apache just to do CDR reporting stuffs |
22:31.02 | dexpdx | my pbx's sync to an "operator" host that has ntp/dns etc on it |
22:31.26 | dexpdx | x86: that's part of the reason why I'm trying to get cdr_odbc working |
22:31.27 | vap0rtranz | dexpdx: an my ip phones sync to the asterisk box, per the docs of tiered ntp synching :) |
22:31.52 | x86 | dexpdx: what DB? |
22:31.59 | dexpdx | x86: oracle |
22:32.06 | x86 | nice :) |
22:32.15 | J4k3 | oracle? and you're on irc trying to get support?!?! :D |
22:32.21 | x86 | hahaha |
22:32.23 | x86 | no kidding |
22:32.23 | dexpdx | J4k3: the problem is not with oracle |
22:32.31 | J4k3 | you can pay it, shut up. |
22:32.33 | x86 | you'd think he'd be using CM ;) |
22:32.36 | J4k3 | err, afford it. |
22:33.06 | J4k3 | x86: he should be using a consultant :P |
22:33.13 | x86 | *nod* |
22:33.14 | J4k3 | I mean, if you can afford a money bucket like any oracle product |
22:33.22 | J4k3 | you can afford a money bucket like a consultant. |
22:33.24 | x86 | or Cisco Call Manager |
22:33.32 | dexpdx | J4k3: ok enough already - I'm perfectly comfortable w/ oracle it's asterisk that I have less xp with |
22:33.46 | J4k3 | buying ccm is just the art of buying a lot of support time. |
22:33.53 | x86 | *nod* |
22:34.02 | J4k3 | of course, that goes for all cisco products |
22:34.02 | x86 | dexpdx: so what's the problem? |
22:34.05 | dexpdx | I've never used CCM - I used Avaya in the past |
22:34.08 | *** join/#asterisk eth01 (i=foobar@gentoo/user/eth01) |
22:34.16 | J4k3 | cisco: we don't make it better, we just give you a lot of white-shirt-wearing support. |
22:34.32 | dexpdx | x86: http://pastebin.ca/926909 take a look |
22:34.34 | dexpdx | no connect |
22:34.43 | x86 | J4k3: and sometimes they give you propaganda t-shirts as well, free of charge! |
22:35.03 | x86 | dexpdx: do you see any connection attempt at all? |
22:35.06 | dexpdx | it really shouldn't matter that I'm using oracle as long as I'm going through the unixODBC layer |
22:35.07 | [TK]D-Fender | x86, No, thats a "bundled expense" |
22:35.26 | dexpdx | x86: from the oracle side or asterisk side? |
22:35.33 | J4k3 | x86: oddly my shirts quit coming when I quit paying for smartnet |
22:35.42 | J4k3 | 'free' my ass :D |
22:36.26 | J4k3 | (any of you have seen kung fu hustle?) |
22:36.38 | J4k3 | cisco my ass, smartnet my ass, free my ass, support my ass... |
22:36.47 | dexpdx | people wouldn't need cisco or avaya if someone would make hardware for asterisk that supports hot-swapping hardware |
22:36.51 | dexpdx | i.e. new pri ports |
22:37.21 | J4k3 | dexpdx: well, I seriously doubt any of the non-* related pbx's out there run on top of linux |
22:37.37 | J4k3 | they may use linux for certain parts of the system, but not the primary kernel. linux is a huge limiting factor IMHO. |
22:38.01 | dexpdx | J4k3: the how-swapping in the OS is not the problem... |
22:38.18 | J4k3 | dexpdx: but I bet the kernel goes *hiccup* when it occurs :) |
22:38.25 | dexpdx | in fact all someone would have todo is build a "firewire" chassis of some sort |
22:39.02 | x86 | dexpdx: ATCA? :) |
22:39.11 | dexpdx | J4k3: I'm not arguing that there are problems with it... I'm just saying that kind of stuff needs to happen before softswitches can really compete |
22:39.17 | J4k3 | but, IMHO, asterisk's primary market is a lot smaller than that |
22:39.24 | x86 | dexpdx: there are ATCA chassis that are more than capable of running linux and asterisk |
22:39.38 | J4k3 | * sells well in places that would normally pay out the ass to their telco for centrix services. |
22:39.48 | x86 | J4k3: centrex? |
22:39.58 | dexpdx | x86: does asterisk support the addition of new hardware with out reload? |
22:40.05 | J4k3 | x86: err, wasn't centrix bell's name for switch-based-pbx? |
22:40.32 | x86 | J4k3: i'm using Asterisk in an organization with 7 sites doing outbound callcenter stuff... about 50,000 calls a week going out 7 Asterisk boxes |
22:40.57 | x86 | J4k3: Centrex is basically a POTS line with crazy features on it |
22:41.04 | J4k3 | around here the telco peddles switch-based pbx services for about $100/month/line + rediculously high standard PSTN LD rates. |
22:41.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
22:41.39 | x86 | heh |
22:45.35 | *** part/#asterisk eth01 (i=foobar@gentoo/user/eth01) |
22:48.03 | dexpdx | res_odbc seems to connect just fine |
22:48.17 | dexpdx | *splat* |
22:49.10 | dexpdx | oh, fucking weird |
22:49.24 | dexpdx | if I load up res_odbc and it connects cdr_odbc connect just fine |
22:49.31 | dexpdx | no res_odbc and cdr_odbc won't connect |
22:49.34 | dexpdx | strange |
23:00.39 | vap0rtranz | can group be used to call any sip channel in it? i mean, what does channelavail checking if not some macro that plays with statuses returned by dial() ... |
23:02.18 | mosty | vap0rtranz, huh? |
23:04.17 | vap0rtranz | mosty: if there's a dozen channels that can dial out, but some might be in use, what's doing the logic for which channel to pick? the wonderful world of *now is missing the channelvariables.txt files and my answer is probably in there |
23:05.04 | mosty | vap0rtranz, what channel specification are you dialing with? |
23:05.05 | vap0rtranz | sip |
23:05.11 | vap0rtranz | but will add zap later |
23:06.10 | *** join/#asterisk PepOSX (n=angeldav@201.243.76.220) |
23:06.22 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
23:12.13 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
23:16.29 | mosty | so Dial(SIP/100&SIP/101) for example? |
23:17.30 | vap0rtranz | mosty: that looks messy for 12 lines ... but is that the only way? |
23:17.42 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
23:18.35 | vap0rtranz | i had started writing a While() loop checking the returning channelstatus for each trunk configured for outbound ... just wondered if there was something else |
23:18.58 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
23:20.08 | [TK]D-Fender | vap0rtranz, Can't see how you would loop those..... |
23:20.29 | *** part/#asterisk LjL (n=ljl@ubuntu/member/ljl) |
23:22.09 | vap0rtranz | [TK]D-Fender: yea i got stuck on the returned statuses, that why i was hoping the channelvariables.txt would say something. looks like i'll just create the amount of trunks on the fly (write a dial string with the amount of trunks that any customer has bought) |
23:22.15 | SteveTotaro | fffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff\gk |
23:22.34 | vap0rtranz | SteveTotaro: you collaspe? |
23:22.52 | kraypius | lawl |
23:23.08 | SteveTotaro | dog was laying on my wireless keyboard on the floor |
23:23.41 | SteveTotaro | not sure how he got that combination of characters though |
23:24.29 | *** join/#asterisk xcompass (n=compass@sr-78.srsv01.resnet.ubc.ca) |
23:27.08 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
23:27.29 | *** join/#asterisk Washy (i=Washy@gateway/tor/x-276b0656a12bea20) |
23:28.20 | Washy | Question: Is it possible to get PTSN access without buying service? |
23:29.02 | Jason99 | Hello all... have a question about Asterisk 1.4.. Last week we upgraded to 1.4 from 1.2 and we've been hearing random DTMF tones on the line but no one is pressing keys.. has anyone else heard of this? |
23:29.45 | *** join/#asterisk angryuser (i=nononon@df01t2-213-44-161-86.d4.club-internet.fr) |
23:30.03 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
23:30.19 | *** join/#asterisk xcompass (n=compass@sr-78.srsv01.resnet.ubc.ca) |
23:30.25 | vap0rtranz | Jason99: yes. it's awful. who's your carrier? |
23:30.51 | [TK]D-Fender | vap0rtranz, No. You should not be doing a "multi-dial". That will be quite bad. Just dial them 1 at a time, back to back. if the 1st gets answered then it will not proceed to the 2nd. |
23:31.09 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
23:31.27 | Jason99 | vap0rtranz: We're using an AudioCodes Mediant 2000 which has Rogers PRI connected to it |
23:31.27 | tzafrir | Washy, sure |
23:31.45 | Washy | how? do you need home phone service? |
23:31.51 | vap0rtranz | [TK]D-Fender: so i should not use Dial(SIP/trunk_1& .. &SIP/trunk_n)? i should just list the trunks in a dialplan |
23:32.05 | Washy | do you need any sort of paid service? |
23:32.06 | mosty | vap0rtranz, you can use call queues with the ringall policy, or you could use variables, Dial(${SALES}) where you have previously set SALES=SIP/100&SIP/101 etc |
23:32.15 | [TK]D-Fender | vap0rtranz, No, you should dial them sequentially |
23:32.18 | Washy | how do you get a normal phone number? |
23:32.36 | vap0rtranz | [TK]D-Fender: mosty and you are saying conflicting things ... |
23:32.50 | vap0rtranz | mosty: that's what i was thinking ... but evidently bad idea |
23:32.55 | tzafrir | Washy, all you need is a cousin at some service provider |
23:32.55 | ManxPower | You don't usually want to do paralell dial instead of a queue |
23:33.11 | ManxPower | it will REALLY piss people off when their phone keeps ringing but someone else catches the call |
23:33.23 | mosty | vap0rtranz, no it's not a bad idea, it's just a different behaviour- it depends what you want this to do exactly |
23:33.25 | [TK]D-Fender | vap0rtranz, You seem to be asking about going through a number of outbound providers to get to the PSTN, is that correct? |
23:33.46 | Washy | tzafrir: shut up |
23:34.41 | vap0rtranz | [TK]D-Fender: maybe i'm just overthinking. but surely * has some way of knowing you've registered several providers and it picks one to dial out with ... |
23:34.56 | [TK]D-Fender | vap0rtranz, No, it doesn't, and I'll take your answer as a "yes" |
23:35.12 | [TK]D-Fender | vap0rtranz, * dials exactly, and only what you explicitly tell it to in your dialplan. |
23:35.26 | ManxPower | vap0rtranz: No, you Dial() if it fails it will exit and set a dialplan variable DIALSTATUS and HANGUPCAUSE |
23:35.39 | [TK]D-Fender | vap0rtranz, and indeed you do NOT want to be doing "Dial(SIP/sadggh&sip/SJHGSDJS....) and combining them there. |
23:35.56 | vap0rtranz | mosty: ah, well we have some customers who want separate lines for separate things, so i can't just put all the registered sip providers in one group and let them dial out. |
23:36.02 | [TK]D-Fender | vap0rtranz, just dial them sequentially like I have described. |
23:36.14 | [TK]D-Fender | vap0rtranz, No, you can't. |
23:36.18 | vap0rtranz | [TK]D-Fender: ok |
23:36.53 | Washy | anyone? |
23:36.56 | [TK]D-Fender | vap0rtranz, so set up your dialplan to go through your resources in the order you wish to prioritize based on what was dialed. |
23:36.59 | mosty | vap0rtranz, what kind of things do you want to separate the calls based on? |
23:36.59 | *** join/#asterisk joobie (n=joobie@joobie.org) |
23:37.33 | [TK]D-Fender | Washy, somebody is paying for it, and sure there are "free" services, but ehy usually have a different kind of "cost" |
23:38.00 | Washy | examples? |
23:38.07 | vap0rtranz | mosty: in one case, one extensions gets its own "priority" sip line, in & outbound. in another, the lines are used for different customers so has to be separated out (for ease of call accounting it seems) |
23:38.39 | [TK]D-Fender | Washy, ipkall.com offers free inbound service |
23:38.54 | Washy | what's the cathc? |
23:38.58 | Washy | catch? |
23:39.05 | outtolunc | missed the convo, but least cost routing can be 'tiered' for more important clients <G> |
23:39.24 | mosty | vap0rtranz, then each of your sip phones should start in a different context, based on that reasoning. from there you can dial via different services |
23:39.57 | vap0rtranz | mosty: and the sip providers in separate contexts |
23:40.32 | vap0rtranz | i'm going to go KISS and just enumerate through the lines :) under the correct contexts of course |
23:40.33 | [TK]D-Fender | Washy, For this on, not perceivable "catch". GrandCentral is one that can be used with some trickery for free outbound via Gizmo. |
23:40.40 | mosty | eg in [group1] you can Dial(SIP/provider1...etc) followed by Dial(SIP/falloverprovider...etc) |
23:40.59 | joobie | guys |
23:41.11 | joobie | is there a good / cheap device that can convert SIP to analogue? |
23:41.22 | joobie | i want to support around 10-20 phones with the device |
23:41.37 | joobie | so i can run analogue phones around the office, rather than voip |
23:41.44 | [TK]D-Fender | joobie, good != cheap. |
23:41.47 | joobie | hehe |
23:41.50 | ManxPower | Well, you can get the cost down to something like $30/port if you are willing to have 5 - 10 devices, power supplies, etc |
23:41.58 | ManxPower | Sipura 2100 or similar |
23:42.00 | joobie | also are there any down sides to this solution? |
23:42.08 | [TK]D-Fender | joobie, but your best value would probably be 3 x Linksys SPA-8000 (8 FXS each). |
23:42.12 | outtolunc | just get a t1 card and a channel bank |
23:42.16 | ManxPower | joobie: It's analog and you have to manage 1 device for every 2 lines |
23:42.28 | ManxPower | or whatever [TK]D-Fender recommends |
23:42.28 | [TK]D-Fender | outtolunc, far from "cheap" |
23:42.32 | [TK]D-Fender | ManxPower, lol |
23:42.44 | joobie | why one device for 2 lines?.. it's still an RJ12 ya? |
23:42.47 | outtolunc | not all channel banks are expensive |
23:42.51 | joobie | one-to-one? |
23:43.00 | outtolunc | and a sip device * 20 isnt' cheap either |
23:43.00 | vap0rtranz | mosty: thanks for the help |
23:43.06 | [TK]D-Fender | ManxPower, the SPA-8000 does 8 ports at the same cost ratio as 2-port ATA's so I suggest it these days... and its a lot less to configure... |
23:43.07 | vap0rtranz | and [TK]D-Fender |
23:43.43 | joobie | hmm |
23:43.45 | joobie | what about the quality tho |
23:43.54 | joobie | do u lose anything in quality / features going this path |
23:43.59 | joobie | as opposed to true digital handsets? |
23:43.59 | adeel | is it true that * doesn't work well on a machine with more than 1 NIC? |
23:44.28 | ManxPower | joobie: you lose almost every single feature a digital set has |
23:44.39 | outtolunc | anyways <G> |
23:44.59 | Jason99 | has anyone heard of random dtmf tones on asterisk 1.4 even if no digits were pressed? Using rfc2833... Has anyone found a solution? |
23:45.15 | ManxPower | I do agree with outtolunc, if you are going be stupid and short sighted enough to go with analog at least use a channel bank |
23:45.23 | [hC] | Can I add an extra field to a CDR that gets passed forward through the other * boxes a call might go through? I have a CPE asterisk box that sets an account code right now for if they used a 'long distance code' to dial out, but that accountcode is trampled when the call comes to me, to send to the pstn |
23:45.24 | outtolunc | nods |
23:45.34 | mosty | Jason99, i have heard of people with that issue, have you checked bugs.digium.org ? |
23:45.36 | Washy | Is IPkall doing something with your incoming #? |
23:45.44 | Washy | something naughty |
23:45.45 | [TK]D-Fender | outtolunc, new math for you : http://www.voipsupply.com/product_info.php?products_id=2912 |
23:45.48 | adeel | Jason99, can you elaborate your setup? |
23:46.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
23:46.17 | drmessano | Naughty? |
23:46.17 | [TK]D-Fender | outtolunc, $250 = 8 ports * 3 = $750 for 24 ports, no need to mess with hardware in your server or hit-or-miss ebay-ing |
23:46.31 | adeel | Jason99, i had that problem when i was using * 1.2 machines to connect to * 1.4 and NAT/reinvite's |
23:46.34 | joobie | what are some of the major features ManxPower |
23:46.38 | mosty | [hC], i forget if there's a way to pass channel variables over IAX besides mangling the dial string |
23:46.41 | [TK]D-Fender | outtolunc, And relieves IRQ load etc off your server. |
23:46.44 | joobie | this will be for a call center.. so it's just outbound calls.. nothing fancy |
23:46.50 | Jason99 | ATA --> Asterisk 1.4 --> Asterisk 1.4 --> AudioCodes Mediant 2000 --TDM--> Rogers PRI |
23:46.58 | joobie | i mean, is there quality of audio loss or something? |
23:47.05 | ManxPower | joobie: go to polycom, cisco, or aastra's site to learn about their IP phones. Polycom calls them "SoundPoint" |
23:47.12 | Washy | I dunno, like listening to your calls or something |
23:47.16 | outtolunc | first is it even out, and the cost is $300 for 8 ports so $300 * 3 = $900 check your math |
23:47.24 | drmessano | Washy: No, why? |
23:47.37 | joobie | ManxPower, I know.. ive been looking at the 330 |
23:47.38 | [TK]D-Fender | Jason99, Set your mediant to rfc2833 encoding towards * and you should not get DTMF issues. Avoid in-band at all costs... |
23:47.39 | outtolunc | why is trying to help so flippin hard in here nowdays? |
23:47.41 | vap0rtranz | adeel: Jason99 hits this problem on the head. i don't think it's necessarily 1.2 -> 1.4; it seems to just pop-up sometimes even with all 1.4 |
23:47.50 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
23:47.57 | ManxPower | joobie: we did ONE analog Asterisk install. NEVER again. Users could not remember how to do call waiting or transfer calls. |
23:48.07 | [TK]D-Fender | outtolunc, Page says $250. |
23:48.08 | joobie | ManxPower, but im just curious.. if i went this analogue way with a sip->analogue device.. would i lose any voice quality.. noticable that is.. vs jsut straight digital to the end-user? |
23:48.09 | adeel | vap0rtranz, well random dtmf should imply an incorrect setup |
23:48.13 | Washy | why do they provide free service then? |
23:48.18 | ManxPower | and of course they blamed their problems with memory on the phone system |
23:48.28 | Jason99 | [TK]D-Fender: i am using rfc2833 on the Audiocodes towards * |
23:48.34 | vap0rtranz | Jason99: are the beeps typically towards the beginning of the conversation? |
23:48.37 | joobie | ManxPower, that's not an issue here as it's only outbound calls |
23:48.42 | joobie | but good to know:) |
23:48.45 | [TK]D-Fender | Jason99, sure you set * to it as well to avoid double detection? |
23:48.51 | drmessano | Washy: http://lists.digium.com/pipermail/asterisk-biz/2005-July/007081.html |
23:48.51 | ManxPower | joobie: for the most part, assuming you do not need to compress the voice (like if it went over the internet) there is not a significant audio quality loss |
23:48.57 | joobie | ManxPower, what about the voice quality.. ? both mic and speaker? are they noticable degraded if you go analoge? |
23:48.58 | vap0rtranz | [TK]D-Fender: i bet he is rfc2833 all throughout the network |
23:49.17 | Jason99 | vap0rtranz: it's anywhere in the call.. people report that it happens when people are speaking loudyly/laughing... |
23:49.23 | adeel | Jason99, you might want to setup rfc2833compensate |
23:49.28 | Washy | But asterisk doesn't enable you to interface with PTSN for free, right? |
23:49.30 | joobie | I do need to compress the voice |
23:49.32 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-144-229.bltmmd.east.verizon.net) |
23:49.38 | joobie | given this, will there be a difference? |
23:49.40 | ManxPower | joobie: Dude, I can't tell you what the quality of the speaker/mic of the phones you buy. |
23:49.44 | adeel | Jason99, hmmm....i wonder if the loud speech is the problem |
23:49.45 | mosty | Washy, no of course not |
23:49.46 | drmessano | Washy: Why not? |
23:49.48 | joobie | ahh k |
23:50.02 | vap0rtranz | adeel: isn't that relaxdtmf? |
23:50.09 | mosty | Washy, no pstn phone company will give you pstn access for free |
23:50.26 | SteveTotaro | you can splice into a 200 pair |
23:50.34 | joobie | im just curious man.. i mean, if i go the digital -> analogue .. I know there's a feature loss, but that's not a consideration here because it's only a single outbound call at a time that's required from these phones. I'm wondering if there's any other noticable differences with a setup like this.. any pros / cons |
23:50.38 | Jason99 | adeel: will rfc2833compensate break anything else? Is that something I should set as default in sip.conf ? |
23:50.48 | Washy | what's a CLEC? |
23:50.58 | ManxPower | Washy: a non-bell phone company |
23:50.58 | [TK]D-Fender | joobie, Using ATA's for phones is jsut fine if you have the extra phones laying around. |
23:50.58 | SteveTotaro | google know |
23:50.59 | adeel | vap0rtranz, they're 2 separate settings...relaxdtmf is one thing, while rfc2833compensate is supposed to compensate for pre-1.4 DTM transmission |
23:51.03 | SteveTotaro | ~clec |
23:51.03 | jbot | i guess clec is Created by the Telecommunications Act of 1996, a CLEC is a service provider that is in direct competition with an incumbent service provider. CLEC is often used as a general term for any competitor, but the term actually has legal implications. To become a CLEC, a service provider must be granted "CLEC status" by a state's Public Utilities ... |
23:51.16 | ManxPower | you don't ever want to use relaxdtmf |
23:51.21 | [TK]D-Fender | joobie, I linked to a product would would quite suitably fit your needs. |
23:51.23 | ManxPower | if you need to use it, then you have OTHER issues. |
23:51.27 | vap0rtranz | adeel: i meant people's voices causes beeps, which seems to be one thing Jason99 notices |
23:51.33 | SteveTotaro | sometimes you want to use relaxdtmf |
23:51.38 | SteveTotaro | never say never |
23:51.40 | ManxPower | SteveTotaro: no you don't. |
23:51.44 | [hC] | anyone know if cdruserfield is carried forward in an iax call? |
23:51.48 | joobie | [TK]D-Fender, so you think buy like 2-3 of those devices and wack them all together? |
23:51.50 | SteveTotaro | speak for yourself buddy |
23:51.52 | ManxPower | If you have to use relaxdtmf then you have a gains issue with your telco |
23:52.07 | *** join/#asterisk xcompass (n=compass@sr-78.srsv01.resnet.ubc.ca) |
23:52.08 | SteveTotaro | whatever works |
23:52.10 | ManxPower | fix the audio level problems and the need for relaxdtmf goes away |
23:52.11 | adeel | vap0rtranz, i guess the question is what kinds of phones Jason99's clients are using |
23:52.28 | SteveTotaro | if relax dtmf gets the job done |
23:52.29 | drmessano | SteveTotaro: Rewire your central office, lazy ass |
23:52.37 | drmessano | :) |
23:52.44 | SteveTotaro | rather than fighting with your carrier for months about gains |
23:52.54 | Jason99 | adeel: just to add to the mix.. the beeps are always on the far end.. my clients don't hear them, its the other end |
23:52.58 | [TK]D-Fender | joobie, Yup. If I was budget conscious and wanted analog, that's what I would do. |
23:52.59 | SteveTotaro | and them having no idea what the hell you are talking about |
23:53.14 | joobie | [TK]D-Fender, what if it wasnt budget conscious and wanted analog?:P |
23:53.29 | adeel | Jason99, so that means you're transmitting them... |
23:53.38 | adeel | Jason99, what type of phones are you using? |
23:53.48 | [TK]D-Fender | joobie, then I'd spend like double the money on an AudioCodes MP-124 24-port redundant gateway :) |
23:53.57 | Jason99 | adeel: not sure, it happens to several different clients, they all use different phones and different ATAs |
23:54.02 | joobie | eheh thanks TK |
23:54.02 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
23:54.08 | [TK]D-Fender | joobie, thats what I'd use in large installs. |
23:54.10 | Jason99 | adeel: if you think that could be a cause, i will find out |
23:54.16 | SteveTotaro | quintum tenor ax rulez |
23:54.32 | Yourname`` | So, in the CLI of a 1.4 dualcpu duocore box, remote unix connection keeps happening and disconnected. No manager enabled.. what's going on?? |
23:54.45 | SteveTotaro | or just go with an adtran or adit 600 |
23:54.52 | SteveTotaro | for your analog |
23:54.54 | [TK]D-Fender | joobie, for SMB use I more than happy to stack a few SPA-8000's together. Once you hit 5 units or so it DOES start to get a little ridiculous though. |
23:55.06 | joobie | hehe yea i can imagine |
23:55.14 | joobie | thanks heaps TK |
23:55.17 | joobie | also the uplink to those |
23:55.18 | [TK]D-Fender | joobie, The solution should be scaled to the need. |
23:55.24 | joobie | do u just consolidate via a switch / router |
23:55.27 | joobie | and just push out the net? |
23:55.38 | joobie | or do u go to an asterisk box.... and then out the net |
23:55.43 | joobie | yea - i hear ya |
23:55.45 | [TK]D-Fender | joobie, I only use elephant guns on ant problems when I want a free light & sound show as well :) |
23:55.45 | SteveTotaro | the solution should not be scaled to the need but the future need |
23:55.46 | adeel | Jason99, you're not using canreinvite anywhere on the path are you? |
23:55.56 | joobie | lol |
23:56.02 | Washy | y is stanaphone not accepting new customers? |
23:56.07 | [TK]D-Fender | joobie, I don't usually use or advocate using an ITSP for corporate PSTN |
23:56.12 | joobie | it's a good approach TK.. nto overkill. |
23:56.26 | SteveTotaro | maybe stan is going broke |
23:56.28 | SteveTotaro | ? |
23:56.33 | drmessano | stanaphone is dead |
23:57.03 | Jason99 | adeel: from the ATA through the 2 Asterisk servers, canreinvite=no, on the AudioCodes canreinvite=yes |
23:57.07 | joobie | [TK]D-Fender, i hear ya.. the cost problem is a factor here.. it's turned out to be pricy for the PSTN install so the client wants to go voip over itsp |
23:57.25 | [TK]D-Fender | joobie, lines are nasty in your area? |
23:57.25 | joobie | [TK]D-Fender, if that is a given though.. do u see the need to go through asterisk or just route straight out to the ITSP from the linksys device |
23:57.32 | joobie | [TK]D-Fender, they are expensive |
23:57.48 | joobie | it's around 1k install on a 24-m contract.. and about 250$/month ongoing.. minus calls |
23:57.59 | joobie | -24m +12m |
23:58.10 | [TK]D-Fender | joobie, you'll probably want * there so as to handle the calls.... you don't want literally independant jacks do you? like 100% separate lines from each other that can't dial between themselves, etc... |
23:58.10 | SteveTotaro | just drop a tenor AX on your lan |
23:58.14 | [TK]D-Fender | joobie, Do you |
23:58.17 | [TK]D-Fender | ? |
23:58.21 | SteveTotaro | and have it register to your itsp |
23:58.30 | adeel | Jason99, so that means your using old analog telephones? i wonder if your ATA has any dtmf settings |
23:58.38 | [hC] | argh. how annoying. So there's no way to carry forward custom CDR fields between IAX/SIP connections? |
23:59.01 | mosty | [hC], there are indirect methods |
23:59.05 | joobie | TK, there's no requirement for phones to be able to dial each other.. they can be 100% independant |
23:59.07 | ManxPower | [hC]: SIP and IAX are not the same thing. |
23:59.14 | ManxPower | There is IAXVARS or something like that |
23:59.22 | ManxPower | SIP has custom headers you can set/get |
23:59.25 | joobie | .. of course the feature wouldn't hurt, but it's not a requirement. It's outbound calls only... like a call center |
23:59.32 | Jason99 | adeel: yes, the ATA are set to rfc2833.. using Mediatrix, Linksys SPA and Dlink DVG |
23:59.35 | vap0rtranz | Jason99: do other IVR's work? as in, any buttons pressed are always recongized correctly |
23:59.44 | Jason99 | vap0rtranz: yes |
23:59.47 | [hC] | ManxPower: Hm. So I guess the way to do it is to check for the presence of those on the otherside and write to cdr then. |