00:00.25 | bird_of_Luck | jmardonesk: what does 'show function ODBC_PREGUNTA1' says? |
00:01.48 | jmardonesk | <PROTECTED> |
00:01.49 | jmardonesk | [Syntax] |
00:01.49 | jmardonesk | ODBC_PREGUNTA1(<arg1>[...[,<argN>]]) |
00:01.49 | jmardonesk | [Synopsis] |
00:01.49 | jmardonesk | Runs the referenced query with the specified arguments |
00:01.49 | jmardonesk | [Description] |
00:01.51 | jmardonesk | Runs the following query, as defined in func_odbc.conf, performing |
00:01.53 | jmardonesk | substitution of the arguments into the query as specified by ${ARG1}, |
00:01.55 | jmardonesk | ${ARG2}, ... ${ARGn}. The values are provided either in whole as |
00:01.57 | jmardonesk | ${VALUE} or parsed as ${VAL1}, ${VAL2}, ... ${VALn}. |
00:01.59 | jmardonesk | This function may only be set. |
00:02.02 | jmardonesk | SQL: |
00:02.05 | jmardonesk | INSERT INTO asterisk.encuesta (respuesta1, hora) VALUES ('${SQL_ESC(${ARG1})}',NOW()) |
00:03.38 | jmardonesk | the query is fine http://pastebin.com/d26b28664 for more info |
00:05.06 | bird_of_Luck | jmardonesk: can you post function definition in func_odbc.conf to pastebin.com ? |
00:06.01 | jmardonesk | bird_of_Luck, http://pastebin.com/d26b28664 |
00:09.52 | bird_of_Luck | jmardonesk: try using Set(XX=${ODBC_PREGUNTA1(0)}) |
00:14.14 | jmardonesk | bird_of_Luck, when i put that, I see the error [Sep 23 19:53:51] ERROR[5917]: pbx.c:1552 ast_func_read: Function ODBC_PREGUNTA1 cannot be read |
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00:16.34 | bird_of_Luck | jmardonesk: dsn 'astrealtime' exists and works ? 'module reload func_odbc.so' shows no warnings/errors? |
00:18.19 | jmardonesk | astrealtime works for me in the cdr, and the reload dont show any error |
00:21.09 | bird_of_Luck | jmardonesk: something is wrong in func_odbc.conf. See 'This function may only be set' in function desctiption |
00:22.05 | phix | jeev: awesome |
00:22.29 | bird_of_Luck | jmardonesk: for valid functions with read= 'function show ..' should report that in can be read |
00:23.14 | bird_of_Luck | jmardonesk: are you sure you posted real func_odbc config on pastebin ? |
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00:29.13 | bird_of_Luck | jmardonesk: you should get something like http://pastebin.com/m22df4df0 for function with read=.. and write=.. |
00:30.13 | Qwell | you can't insert on a read.. |
00:34.47 | bird_of_Luck | Qwell: odbc module is not interested in what SQL query do. It simply substitutes variables and execute query. Yes, you cant't get valuable result but you can do INSERT into db. |
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01:23.49 | jmardonesk | in read in can only select? and in write i only can insert or update? |
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01:34.53 | drmessano | i do not how open the source |
01:34.56 | drmessano | halp? |
01:35.18 | Qwell | drmessano: Don't you have testing to do or something? |
01:36.23 | jmardonesk | bird_of_Luck, I dont understand whats mean "This function may only be set". I have in func_odbc.conf |
01:36.25 | jmardonesk | write=INSERT INTO asterisk.encuesta (respuesta1, hora) VALUES ('${SQL_ESC(${ARG1})}',NOW()) |
01:36.31 | drmessano | Setting back up my DNS for the 3rd time in 5 days |
01:37.01 | jmardonesk | and the context say: exten => 102,2,Set(ODBC_PREGUNTA1(0)) |
01:37.34 | drmessano | I did check all the VoIP spam tabloid websites and have heard no juicy astricon news |
01:37.40 | drmessano | WTF is up with dat? |
01:38.37 | jmardonesk | when i put exten => 102,2,Set(${ODBC_PREGUNTA1(0)}) dont work, when i change write for read, dont work... I dont know what i need to do.. |
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01:42.49 | bird_of_Luck | jmardonesk: Set() wants evaluated expression like X=Y |
01:43.45 | jmardonesk | bird_of_Luck, how can only call the query ODBC_XXXX? |
01:44.18 | bird_of_Luck | jmardonesk: save function as read, do 'module reload func_odbc.so', verify you get it right (by doing show function ...) |
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01:44.53 | bird_of_Luck | jmardonesk: change your set to SET(UNNEEDEDSTUFF=${ODBC_PREGUNTA1(0)}), save, do 'dialplan reload' and retry |
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01:48.02 | Qwell | umm, here's a stupid question |
01:48.12 | Qwell | if both read and write do the same query... why bother with fixing it? |
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01:55.42 | puga | anyone knows Russell Bryant? |
01:56.19 | puga | bah... have to go.. Oo' |
01:57.56 | jmardonesk | bird_of_Luck, muchas gracias dude ;-) |
01:58.30 | jmardonesk | bird_of_Luck, it works very fine, now show only a warning |
01:59.01 | jmardonesk | bird_of_Luck, but is now storing data in mysql |
02:02.27 | bird_of_Luck | jmardonesk: good. another way is to write write=.. with ${ARGX} and ${VALX} and call function like Set(FUNC_NAME(ARG1,ARG2,..)=VAL1\,VAL2\,..) |
02:04.41 | bird_of_Luck | jmardonesk: http://pastebin.com/m2464efb0 for example |
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02:05.47 | pcrane | does anyone have experience with Linksys spa8000s and DTMF in asterisk? |
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02:20.05 | jmardonesk | how can detect the answer in a Zap channel? i forgot what are the option in /etc/asterisk/zapata.conf |
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03:06.57 | jameswf-home | Marketing at its best http://www.voipspeak.net/2008/fonality-provides-free-replacement-phone-systems-to-customers-devastated-by-hurricane-ike/ |
03:08.05 | eric2 | hmm, I restarted zaptel and it took the whole machine down |
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03:08.17 | jameswf-home | dont do that |
03:08.29 | hardwire | when did we start naming hurricanes after dead presidents? |
03:09.05 | eric2 | what's the best way to restart zaptel and wanpipe? |
03:09.11 | jameswf-home | reboot |
03:09.18 | eric2 | hmm... :( |
03:09.27 | eric2 | is my machine toast? |
03:09.39 | eric2 | messed up kernel ? |
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03:10.20 | frogonwheels | I'm trying to work a way of having an incoming call with Ringing() Dial(SIP/etc ,20,rd ) but have it Answer() after (say) 10 seconds |
03:12.23 | glaz | wait() ? |
03:13.02 | frogonwheels | ok.. if I have Ringing() Dial(SIP/etc,5,rd) Answer() wait(15) then it stops ringing as soon as the DIal() finishes |
03:13.09 | drmessano | So any big annoucements from Astricon today? |
03:13.38 | frogonwheels | glaz: I'm either missing something obvious, or just trying to be too smart :| |
03:14.09 | frogonwheels | glaz: was that what you meant by using wait? |
03:15.49 | frogonwheels | .. and iif I do Ringing() Dial(SIP/etc,5) ANswer() Dial(SIP/etc,15,rd) then, at least for my sip client on my mobile), the second Dial doesn't connect |
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03:19.15 | frogonwheels | There's an old email that hints on forking a dialplan.. http://tinyurl.com/4snrz5 but I'm not sure I can then direct that incoming channel to Answer() |
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03:21.09 | hardwire | hi |
03:21.16 | wscholar | hey |
03:21.28 | wscholar | <PROTECTED> |
03:21.48 | wscholar | [Sep 23 23:09:27] WARNING[16692]: app_channelredirect.c:112 asyncgoto_exec: ChannelRedirect failed for SIP/provider-08237a80 |
03:22.53 | hardwire | wscholar: read up on this |
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03:22.55 | hardwire | http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Transfer |
03:23.00 | hardwire | krokodilerian: fancy seeing you here |
03:23.07 | Qwell | drmessano: it doesn't actually "start" until tomorrow |
03:23.18 | Qwell | today was developer day, and other random "pre-show" stuff |
03:23.19 | drmessano | :( |
03:23.45 | mosty | is it possible to get asterisk to refuse sip 302 redirects? |
03:23.54 | drmessano | I just want to know if Skype is buying Paltalk.. then I will shut up... |
03:24.11 | jaytee | hehe |
03:24.29 | drmessano | THIS IS IMPORTANT!!!!!ONES!!!!!111!!!! |
03:24.42 | file | wscholar: you are sending it to the transfer context, extension "Local/90005119077482879@transfer" with a priority of 1 |
03:25.10 | wscholar | yes |
03:25.14 | wscholar | trying todo a blind transfer |
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03:34.04 | jeev | Qwell |
03:34.07 | jeev | i cancelled my appearance |
03:35.00 | jaytee | I bet the fans were shattered. |
03:35.02 | hardwire | wscholar: figure it out? |
03:35.10 | hardwire | jeev: where are you atm? |
03:35.24 | jeev | i'm at the airport |
03:35.26 | jeev | i'm coming back home |
03:35.30 | hardwire | aww |
03:35.31 | jeev | they only sent me 150k |
03:35.35 | hardwire | hah |
03:35.36 | jeev | we aggreed on 250k |
03:35.45 | jeev | something about russellb complaining that he's not getting his 100k hooker |
03:35.45 | hardwire | twofiddy? |
03:36.00 | hardwire | jeev: hah.. I'd hooker to him for that much. |
03:36.14 | jeev | lol |
03:36.18 | jeev | you'd hooker to anything for nothing |
03:36.18 | jaytee | writes that down in his journal |
03:36.31 | jeev | journal |
03:36.33 | jeev | rollerblading |
03:36.33 | jeev | wow.. |
03:36.35 | jeev | you sure you're not? |
03:36.59 | drmessano | jeev: I should apologize |
03:37.03 | jaytee | only in your deepest fantasies |
03:37.24 | drmessano | jeev: I know I acted like you are a stark raving lunatic pathological liar |
03:37.41 | drmessano | jeev: Now I realize you're just ill and need help :( |
03:39.06 | drmessano | jeev: Now please, stop trying to stab yourself with that potato chip and get off the roof of the car.. mommy and daddy love you :`( |
03:39.45 | jaytee | I think he's just locked into the emotional state of a 14 year old. That's probably when the "trauma" occurred. |
03:40.30 | drmessano | jaytee: Do we need to break out the "Where did he touch you" bear? |
03:41.08 | jeev | what trauma |
03:41.11 | jaytee | I don't want him acting out his inner demons in here particularly |
03:41.21 | drmessano | jaytee: or is this more like "tell us where left Uncle Joe "sleeping", son" |
03:41.29 | jeev | break yoself |
03:41.30 | wscholar | well if i pass in EXEC "ChannelRedirect" "SIP/provider-081ca788|transfer|SIP/provider/90005119077482879|1" |
03:41.33 | wscholar | i get failure |
03:41.37 | wscholar | if i pass in |
03:41.37 | wscholar | <PROTECTED> |
03:41.39 | wscholar | srry |
03:41.46 | wscholar | <PROTECTED> |
03:41.52 | wscholar | i get congestion / fast busy |
03:42.59 | wscholar | [transfer] |
03:43.00 | wscholar | exten => _X.,1,Noop(Enter transfer context, gr_callid: ${gr_callid}) |
03:43.00 | wscholar | exten => _X.,n,Agi(agi://domU-12-31-39-00-50-32.compute-1.internal/transfer.agi?gr_callid=${gr_callid}) |
03:43.20 | wscholar | if that helps |
03:43.58 | jaytee | would have been nicer if you'd just have wrapped up all that in a nice pretty package on a pastebin |
03:44.02 | jaytee | ~pb |
03:44.02 | 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 |
03:44.29 | hardwire | wscholar: hmm.. |
03:45.59 | frogonwheels | ok, is there a way I can cause an incoming channel to get 'answered' without stopping the Dial() (i'll pb what I've tried) |
03:46.04 | hardwire | Dial(SIP/provider/90005119077482879) |
03:46.06 | hardwire | yar |
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03:48.31 | frogonwheels | http://pastebin.com/d7bb8f70b |
03:51.26 | [TK]D-Fender | frogonwheels: That will grab it after 5s, hold for 20 and jsut flat out hangup. |
03:51.49 | [TK]D-Fender | frogonwheels: I see no point to this at all. Just answer the call then dial like normal. |
03:52.11 | frogonwheels | [TK]D-Fender: yep, I know that is wrong 'cause I tried it, and It makes sense. |
03:52.33 | [TK]D-Fender | frogonwheels: Either way the call will get answered withing 5s regardless unless they hang up nearly instantly, which, really, why do youc are about that so much? |
03:52.37 | frogonwheels | [TK]D-Fender: Normally, I prefer not to Answer() straight away - so people on mobiles can hang up and not incur a charge. |
03:52.46 | frogonwheels | [TK]D-Fender: ok - so make it 20 seconds |
03:53.47 | [TK]D-Fender | frogonwheels: there should also be no need for "r" or "ringing". Your first priority would be the Dial. |
03:54.07 | [TK]D-Fender | frogonwheels: Net your nested dial will not work to try to leave your original channel ringing. |
03:54.17 | [TK]D-Fender | frogonwheels: So no need for the split local channel. |
03:54.36 | [TK]D-Fender | frogonwheels: You'll have to do this as back-to-back dials with an Answer in the middle |
03:54.38 | frogonwheels | [TK]D-Fender: Ok - I know that. But you get what I'm trying to do, and I'm wondering if there is a away to do it. |
03:54.43 | frogonwheels | ok |
03:54.55 | [TK]D-Fender | frogonwheels: As I've just described. That's all there is |
03:55.05 | *** part/#asterisk [jarrod] (n=jarrod@c211-28-176-99.chirn1.vic.optusnet.com.au) |
03:55.06 | frogonwheels | .. and that doesn't appear to work with my sip client on the mobile phone. |
03:55.25 | frogonwheels | - the second dial doesn't work. (still engaged) |
03:56.26 | frogonwheels | I thought there could be an outside chance, that [cont_Vista] context could send (something like) a ChannelRedirect to cause the connection to Answer |
03:57.50 | [TK]D-Fender | frogonwheels: All the redirecting in the world won't cause an answer. An ANSWER will cause an answer. |
03:59.23 | frogonwheels | [TK]D-Fender: makes sense. I found this http://tinyurl.com/4snrz5 - obviously somebody has thought of something similar before me :) |
03:59.35 | frogonwheels | [TK]D-Fender: thanks for the help though. |
04:00.42 | krokodilerian | hm |
04:00.53 | krokodilerian | does some one feel like looking at some C code |
04:01.14 | krokodilerian | asterisk app, "cleaned up" for astricon |
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04:02.16 | geoff2010 | i have a question on g.729 and asterisk... anyone willing to listen? |
04:03.21 | krokodilerian | is not that drunk yet |
04:03.33 | phix | krokodilerian: fail |
04:03.42 | phix | you should drink more |
04:03.55 | krokodilerian | shouldn't, i'll try for a dcap tomorrow |
04:04.00 | krokodilerian | i need at least to get to the hall |
04:04.05 | krokodilerian | preferrably standing |
04:04.22 | mosty | geoff2010, ask |
04:05.01 | [TK]D-Fender | frogonwheels: And that post states failure at every tern |
04:05.26 | frogonwheels | [TK]D-Fender: it was just a proof of concept. |
04:05.35 | frogonwheels | [TK]D-Fender: a debugging straw-man |
04:05.50 | geoff2010 | i want my system to be able to handle 100 concurrent calls, but i only want to purchase 50 g729 licenses. will asterisk automatically start negotiating g711 once all the licenses are used up? |
04:06.06 | frogonwheels | [TK]D-Fender: not much point in dotting i's and crossing t's if the concept is flawed |
04:06.08 | [TK]D-Fender | frogonwheels: If you answer in the multi-dial you cut on the one you ran in "parallel". Complete waste of time. You may as well have dialed for the same amount of time and gone on to the next priority. There is no interrupting dial and changing the rules. |
04:06.23 | [TK]D-Fender | frogonwheels: Dial for X, Answer, Dial for Y. |
04:06.36 | [TK]D-Fender | geoff2010: No. |
04:06.39 | frogonwheels | [TK]D-Fender: gotcha, really I understand. |
04:06.46 | phix | krokodilerian: standing and bladder control is overrated |
04:07.04 | geoff2010 | Fender: what will the behaivor be once i run out of licenses? |
04:07.12 | [TK]D-Fender | frogonwheels: On to more productive things then. |
04:07.19 | frogonwheels | exactly |
04:07.24 | [TK]D-Fender | geoff2010: Calls will be rejected |
04:07.33 | geoff2010 | lame |
04:07.37 | jameswf-home | obscure errors |
04:08.04 | geoff2010 | so it's all or nothing with asterisk, can't mix/match codecs? |
04:08.06 | krokodilerian | wscholar , did you charge your phone? I might need some juice in a while :) |
04:08.36 | [TK]D-Fender | geoff2010: It will match the same compatibility list every time. Then upon attempt it will fail, and thats the end of the call. |
04:09.02 | [TK]D-Fender | geoff2010: IMO * should "reserve" a license in its negotiation", but it does not. |
04:09.18 | wscholar | i'm good |
04:09.22 | wscholar | take it |
04:10.04 | geoff2010 | Fender: agreed. sounds like a somewhat intentional limitation to force people to buy more g729 licenses... perhaps I am wrong. |
04:11.23 | jameswf-home | hmmm http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
04:14.23 | mosty | geoff2010, will this box be primarily for incoming or outgoing calls? |
04:17.25 | geoff2010 | mosty: i actually have 6 asterisk systems which work as one unified platform. it's the telephony interface for a large virtual call center solution. |
04:17.42 | geoff2010 | mosty: the servers do everything, inbound, outbound, IVR, etc |
04:18.44 | mosty | you may as well just buy 100 g729 licences |
04:19.04 | mosty | instead of messing around with a solution which might not work 100% of the time |
04:19.51 | geoff2010 | yeah, 100 was a 'fake' number... we would need a lot more licenses if we switched to g.729. problem is bandwidth is becomming very expensive. not cheap to burst to 500Mb/sec |
04:19.51 | oilinki | how much are the g729 licenses? |
04:20.12 | hardwire | how many goats you got? |
04:20.21 | jameswf-home | Free< |
04:20.30 | hardwire | oilinki: prices are listed on digiums site for their licenses |
04:21.24 | mosty | geoff2010, do your phones support g729? |
04:21.33 | drmessano | lol |
04:21.45 | drmessano | yeah, thats legal.... |
04:22.13 | geoff2010 | we don't have any phones, it's all virtual call center stuff, so we are taking calls from SIP gateway and then calling agents out through the SIP gateway going back to the PSTN |
04:22.26 | geoff2010 | asterisk stays in the middle of all the media |
04:22.38 | geoff2010 | agents use regular home phones |
04:28.57 | oilinki | g729 seems to be usd10/channel |
04:29.13 | hardwire | oilinki: using g729 with a provider? |
04:31.22 | oilinki | hardwire: same situation as geoff2010 has. man in the middle |
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04:36.59 | hardwire | oilinki: mitm happens |
04:38.12 | jameswf-home | is there a way to see the codec in use? |
04:38.57 | krokodilerian | jameswf-home , show channel xxxx |
04:39.02 | krokodilerian | jameswf-home , and then look at the format part |
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04:39.53 | jameswf-home | no format field first thing i tried |
04:40.01 | mosty | jameswf-home, try "sip show channels" |
04:40.04 | krokodilerian | hmmm, should I be able to order room service with food and them carrying me from the code zone to the room |
04:40.06 | mosty | and "iax2 show channels" |
04:40.09 | krokodilerian | well, not on a tray |
04:40.16 | krokodilerian | jameswf-home , what do you mean no format field? |
04:40.31 | krokodilerian | goes to look for an asterisk with channels |
04:40.57 | jameswf-home | hmm 0x0 (nothing) |
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04:42.34 | krokodilerian | <PROTECTED> |
04:42.34 | krokodilerian | <PROTECTED> |
04:42.34 | krokodilerian | <PROTECTED> |
04:42.45 | krokodilerian | jameswf-home , what's your channel driver |
04:44.18 | jameswf-home | sip/zap. was checking out an unofficial g729 build |
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04:48.14 | *** mode/#asterisk [+o russellb] by ChanServ |
04:49.08 | krokodilerian | jameswf-home , can't really help you there |
04:49.13 | krokodilerian | seems like a bug |
04:49.33 | krokodilerian | the one I tested on has g729 |
04:50.36 | krokodilerian | <PROTECTED> |
04:50.36 | krokodilerian | <PROTECTED> |
04:50.36 | krokodilerian | <PROTECTED> |
04:50.40 | krokodilerian | this is from a zap channel |
04:52.39 | krokodilerian | so, whatever, i'll sleep and try to dcap tomorrow :) |
04:52.53 | jameswf-home | fogot g723 hmm |
04:52.57 | jameswf-home | works |
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05:27.59 | hardwire | hi |
05:30.14 | drumkilla | hi2u2 |
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05:43.42 | miloux | o |
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06:23.26 | jameswf-home | heh http://www.russellbryant.net/blog/2008/09/12/the-voice-of-asterisk-does-monty-python/#more-136 |
06:24.00 | Strom_C | NOBODY EXPECTS A MONTY PYTHON JOKE !!! |
06:25.19 | jeev | i dont even know what monty python is |
06:25.49 | jblack | that's a joke, right? |
06:26.23 | jeev | negative |
06:26.38 | jblack | You should know. I suggest you google it up |
06:26.46 | jeev | i'm fuckin <50 years old |
06:27.04 | jblack | No need to be obnoxious. |
06:27.05 | jameswf-home | oy! |
06:27.07 | jeev | lol |
06:27.12 | jeev | i saw a wikipedia thingy |
06:27.13 | jeev | 1969 |
06:27.15 | jeev | jeebus h. christ |
06:27.30 | jameswf-home | Monty Python is timeless |
06:27.32 | Strom_C | monty python was a group of british guys who wrote brilliant jokes which geeks have, through tireless repetition, made lame and unfunny. |
06:27.47 | jameswf-home | it gave us python |
06:27.52 | jameswf-home | okay i made that up |
06:27.52 | jeev | ahh cool |
06:28.01 | jeev | jameswf-home, are you ever out of your home? :D |
06:28.07 | jeev | jameswf No such nick/channel |
06:28.11 | jeev | oh BY THE WAY |
06:28.13 | jameswf-home | yes except when im not |
06:28.14 | jeev | i'm going to england next month |
06:28.19 | jeev | is it cool thereor what |
06:28.31 | jblack | You're home except when you're not home? |
06:28.41 | jameswf-home | correct |
06:29.05 | jeev | http://www.africahit.com/news/index.php?mod=article&cat=othersenglish&article=4452 |
06:29.05 | jeev | ! |
06:29.07 | jeev | catch me there |
06:29.08 | jeev | farting |
06:29.21 | jblack | i suppose you're also paraplegic, except for when you're not. |
06:29.34 | jblack | jeev: Didn't you say you were gonna behave? |
06:29.44 | jameswf-home | LIES! |
06:30.06 | jblack | Oh.. africa hit. Not africa shit. |
06:30.11 | jblack | I'm sorry. |
06:30.25 | jeev | heh |
06:30.30 | jeev | i am behaving |
06:31.01 | jblack | I can apologize again, if you like |
06:31.21 | jeev | dood |
06:31.23 | jeev | london better be fun |
06:31.24 | jeev | i'll be pissed |
06:31.40 | jblack | Plenty to do in london, if you like getting drunk. |
06:31.44 | jeev | i dont drink |
06:31.45 | jeev | never drank |
06:31.46 | jeev | never smoekd |
06:33.18 | jeev | hahah, i told my girlfriend i give her too much freedom |
06:33.19 | drmessano | I'm going to jupiter in August |
06:33.20 | jeev | she's the cutest thing ever |
06:33.23 | jameswf-home | london smells bad |
06:33.49 | drmessano | jeev and I are going to go... as long as they don't make us take the orange pills again |
06:33.53 | drmessano | :( |
06:33.56 | jblack | I never noticed london smelling bad, but I smoke heavily. |
06:34.05 | jblack | well, did at the times I was there. |
06:34.24 | drmessano | We're taking our Neptune Express Black Cards |
06:34.36 | jameswf-home | an odd mix of trash and urine |
06:35.09 | jblack | 6-13 more days. |
06:35.16 | drmessano | Kinda like the smell of a man that just peed in his own vomit, set it on fire, then rolled around in it? |
06:35.19 | drmessano | Thats NYC |
06:35.22 | jeev | heh |
06:51.30 | hardwire | blah |
06:51.46 | hardwire | phoenix smells like sulfer to me |
06:51.53 | creativx | i thought nyc smelled like food |
06:51.59 | creativx | food food and more food |
06:52.02 | hardwire | and even though it's humid.. it's making my nose dry |
06:52.18 | creativx | atleast you dont have rocket fuel in your drinking water ;) |
06:52.29 | hardwire | creativx: put down the vodka. |
06:54.23 | fiddur | Hi. After a queue-call is finished, is there any way to identify what agent or interface that took the call, still in the dialplan? I want to make a custom wrap-up-handling.... |
06:56.34 | hardwire | yup |
06:56.48 | hardwire | check out the "asterisk variables" section in voip-info |
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06:57.07 | hardwire | -> sleep |
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07:03.00 | fiddur | I'm sorry, but I can only find QUEUESTATUS being set from application QUEUE, and none of the QUEUE_*-functions say anything about what agent took the call on the channel either... What am I missing? |
07:08.31 | fiddur | ah, now I found some info about setinterfacevar... |
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07:15.12 | yang | Is there a magic key for deleting all messages from the voicemail ? |
07:16.02 | Strom_C | the closest thing to magic on my telephone is "transfer" |
07:16.25 | Strom_C | although "8" can be pretty fucking magical if you get it drunk first |
07:17.19 | jeev | lol |
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09:11.19 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc6, 1.4.22-rc5 (2008/09/09), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12.1 (2008/09/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon |
09:15.28 | mort_gib | Again, how to I change the ringing fro a caller to indicate that callee is in another call?? |
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09:17.44 | qp | morning all |
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09:51.41 | mort_gib | <PROTECTED> |
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10:32.08 | badcfe | <PROTECTED> |
10:32.11 | badcfe | <PROTECTED> |
10:32.14 | badcfe | i have that when i reload |
10:32.35 | badcfe | and i have a file specified in the custom config |
10:32.48 | badcfe | however, nothing goes into that file when calls end |
10:32.51 | badcfe | 8 - ( |
10:33.04 | badcfe | anyone comes up with a possible reason? |
10:40.33 | badcfe | cdr-custom is *not* shown by "cdr status" |
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10:46.48 | festr_ | hi |
10:48.09 | festr_ | anyone know, why i'm hearing ringing tone immediatly after "SIP/SDP Status: 180 Ringing, with session description" although the device does not send ring for the first five seconds? |
10:48.25 | festr_ | it is some GSM gateway which sends first RTP packet after five seconds |
10:48.44 | festr_ | but in this first seconds asterisk assumes something and generates ring which is wrong |
10:48.47 | festr_ | 1.4 |
10:48.58 | festr_ | any idea what to change or if this is normal behaviour? |
10:51.49 | badcfe | your phone will generate rinning on 180 |
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10:52.28 | badcfe | more info about my problem descirbed above: custom cdr *was* enabled on make menuselect under Call Detail Recording |
10:54.14 | festr_ | badcfe: dont understand, 180 generates ring although it is "with session description?" |
10:54.55 | festr_ | aha |
10:54.57 | festr_ | i got it |
10:54.57 | festr_ | https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-January/011850.html |
10:55.03 | festr_ | If a 180 (Ringing) has been received but there are no incoming media packets, generate local ringing. |
10:55.12 | festr_ | If a 180 (Ringing) has been received and there are incoming media packets, play them and do not generate local ringing. |
10:56.16 | badcfe | and sometimes the behaviour is configurable on the phone |
10:56.16 | festr_ | unfortunately it is not. it seems that i need 183 |
10:58.26 | badcfe | festr_: you have used custom-cdrs? i have a little puzzle enabling it right now. |
10:59.59 | badcfe | festr_: on reload i see that /etc/asterisk/cdr_custom.conf are being parsed but without any effect ... you know why this could be so? |
11:00.38 | festr_ | i'm using /etc/asterisk/cdr_mysql_custom.conf |
11:01.15 | badcfe | hmmm. when i issue "cdr status" the cdr-custom doesnt show even |
11:01.31 | festr_ | ast-mezivoda*CLI> cdr status |
11:01.31 | festr_ | CDR logging: enabled |
11:01.31 | festr_ | CDR mode: simple |
11:01.31 | festr_ | CDR output unanswered calls: no |
11:01.31 | festr_ | CDR registered backend: cdr_manager |
11:01.34 | festr_ | CDR registered backend: cdr-mysql-custom |
11:02.08 | badcfe | i have nothing in /etc/asterisk/cdr.conf since its enabled by default, and my cdr_custom.conf now contains the example Master.csv |
11:02.41 | badcfe | festr_: i suppose if you uncomment the example in cdr_custom.conf, the file shows up under /var/log/asterisk/cdr-custom ? |
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12:22.21 | yang | I am wondering is there a device to integrate asterisk with hamradio ? |
12:24.16 | boolean12 | This sounds fun. |
12:24.23 | boolean12 | Well. |
12:24.28 | boolean12 | Yeah. :-p |
12:24.34 | boolean12 | You could use the console. |
12:24.48 | yang | I would probably need a device |
12:24.49 | boolean12 | Take the sound output from the soundcard into your base. |
12:25.00 | yang | sort of a receiver to trigger it on some frequency |
12:25.40 | boolean12 | If your base has a serial interface, you could just run a system command to key it, transmit then unkey. |
12:26.45 | boolean12 | Hmm. |
12:27.02 | yang | boolean12: i mean triggering the numbers would be possible via DTMF tones |
12:27.06 | boolean12 | I wonder if you could use talking detection to key it. |
12:27.09 | boolean12 | Yeah, it would be. |
12:27.32 | boolean12 | Hm. |
12:27.38 | boolean12 | Sounds fun ^^ |
12:31.18 | tzafrir_laptop | boolean12, pciradio, baiscally |
12:31.34 | tzafrir_laptop | err, the above was for yang |
12:31.37 | boolean12 | Heh ^^ |
12:31.46 | tzafrir_laptop | ask SteveTotaro |
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12:45.44 | oktay | hello.. what does 5 very rapid tones when I take the line off-hook mean? |
12:45.58 | oktay | or 8 |
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12:52.05 | srt | hi, i am looking for dids in portugal. can anybody give me a hint? |
12:52.46 | eric2 | didww |
12:53.53 | srt | unfortunately they don't have anything in portugal, same for didx |
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13:03.10 | oktay | Outbound Routes/Dial Patterns -> 88 |
13:03.19 | oktay | gives me a PSTN dial tone when I dial 88 |
13:03.21 | oktay | why is that? |
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13:05.47 | oktay | erm. sorry. that was a freepbx question. |
13:06.26 | [TK]D-Fender | oktay: For which they have their own channel ; #freepbx . Go give them a visit I'm sure they can tell you what you need to know |
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13:07.38 | oktay | yes. i typed in the wrong window. that's why i said sorry. |
13:07.44 | oktay | thanks |
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13:12.09 | oktay | bye |
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13:15.38 | traxxos | hello |
13:17.44 | Katty | wobbles |
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13:17.52 | Katty | who stole my caffeine iv |
13:18.08 | blaylock | anyone have 1.6 working with unified messaging? |
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13:18.46 | seanbright | anyone know how to set callerid name on an outbound analog line? |
13:18.49 | seanbright | teehee |
13:18.54 | seanbright | <-- s.o.b. |
13:19.00 | Katty | bonks seanbright |
13:19.22 | Katty | Sean NotSo Bright |
13:19.28 | seanbright | clever! |
13:19.44 | seanbright | my middle initial is R |
13:19.49 | hi365 | is there a limit to the ammont of includes voicemail.conf can have? |
13:19.53 | Katty | there's two new litters of GSDs in the newspaper :> |
13:19.54 | seanbright | so when i say "Sean R. Bright" |
13:19.57 | coppice | seanbright: go to Japan :-) |
13:20.18 | seanbright | jackasses says "shouldn't that be Sean *is* Bright??" |
13:20.21 | Katty | plots phone calls around 10 |
13:20.43 | seanbright | coppice: what is in japan besides really frightening porn? |
13:20.52 | Katty | lolita goths. |
13:20.57 | Katty | ipod knockoffs |
13:21.05 | coppice | you actually can set caller ID on some analogue lines there |
13:21.36 | Katty | really bad food. |
13:21.36 | seanbright | ohhh |
13:21.36 | Katty | their mcdonalds carry soy burgers. |
13:21.55 | Katty | walmart labor camps. |
13:21.55 | seanbright | coppice: i was just asking that question as a friendly ribbing towards Katty |
13:21.55 | Katty | coppice: yes. he was giving me hell. |
13:21.56 | Katty | coppice: in a nice, sort of way. |
13:22.01 | gr0mit | mmmmh soy-burgers |
13:22.08 | Katty | i love you too sean. |
13:22.16 | seanbright | <3 |
13:22.23 | Katty | boca burgers aren't too bad. |
13:22.28 | Katty | you can even get those at Denny's! |
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13:22.49 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) [NETSPLIT VICTIM] |
13:22.49 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) [NETSPLIT VICTIM] |
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13:22.49 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
13:22.49 | *** mode/#asterisk [+o file] by irc.freenode.net |
13:23.03 | coppice | some markets kinda make veggie burgers essential for MacD. A big mac would have a limited market in India |
13:23.03 | *** join/#asterisk miloux (n=miloux@213.88.194.123) [NETSPLIT VICTIM] |
13:23.03 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) [NETSPLIT VICTIM] |
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13:23.11 | coppice | I think they only serve rice in .ph, though |
13:23.26 | Katty | sacred-god-burgers |
13:23.40 | Katty | another example of religion holdin the man down |
13:24.02 | tzafrir_laptop | ham-burgers? |
13:24.10 | coppice | how exactly does not eating cows keep someone down? |
13:24.12 | Katty | cow-burgers |
13:24.23 | Katty | coppice: if they ate the cows, they'd have less famine |
13:24.34 | gr0mit | munches a ratburgeer |
13:24.40 | coppice | huh? it works the other way around |
13:24.53 | Katty | coppice: perhaps we're not on the same page. |
13:25.00 | Katty | coppice: it's still really early in the Angeliverse. |
13:25.38 | coppice | it takes 10kg of veggie to make 1kg of beef. eating animals causes starvation |
13:25.52 | gr0mit | not for me though ;-) |
13:26.00 | coppice | do you eat horse? |
13:26.07 | gr0mit | nope. |
13:26.12 | Katty | isn't that illegal? |
13:26.18 | Katty | i know it's illegal to shoot a horse now. |
13:26.20 | gr0mit | the English do not eat horse |
13:26.20 | srt | no horse is great |
13:26.29 | Katty | i've heard horse is really good. |
13:26.30 | gr0mit | at least not knowingly |
13:26.50 | coppice | no. most western people just don't do it. similarly much of asia treats its traditional farm working animals with the same respect |
13:26.51 | Katty | i also hear cat is really good. |
13:27.01 | viraptor | is there any official / proven way to add join/leave sounds to app_conference? |
13:27.06 | coppice | donkey is better than horse |
13:27.14 | Katty | viraptor: show application meetme |
13:27.16 | gr0mit | likes ostrich |
13:27.21 | Katty | viraptor: also perhaps, core show applicaiton meetme |
13:27.33 | Katty | viraptor: show core application meetme? |
13:27.36 | Katty | viraptor: something like that. |
13:27.38 | coppice | kiwi is better than oistrich |
13:27.39 | viraptor | Katty: I don't want meetme -> is there any way to add it to app_conference? |
13:27.54 | Katty | viraptor: i haven't the froggiest. |
13:28.01 | Katty | viraptor: all the cool kids use meetme. |
13:28.25 | Katty | coppice: you mean... the kiwi fruit? |
13:28.38 | Katty | coppice: i'm having horror flashbacks of kiwi, from youtube. |
13:28.48 | coppice | no. the bird. similar meat to ostrich, but more flavourful and more tender |
13:28.52 | viraptor | Katty: doesn't work good enough here... app_conf can handle 2 times more users unfortunately |
13:28.56 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:29.24 | jameswf-home | never call an aussie a kiwi or a kiwi an aussie |
13:29.48 | gr0mit | or a Brit an ozzie |
13:29.49 | Katty | viraptor: best of luck in your search. |
13:29.54 | coppice | i think most of the kiwis eaten in asia actually come from .au |
13:30.11 | Katty | i must google this kiwi bird. |
13:30.24 | jameswf-home | most ausies were brits granted long ago |
13:30.38 | Katty | it is the youtube kiwi :< |
13:30.45 | Katty | unacceptable! |
13:31.10 | Katty | jbot: kiwi? |
13:31.11 | jbot | kiwi is, like, A framework for Python applications with graphical user interfaces. URL: http://www.async.com.br/projects/kiwi/ |
13:31.28 | Katty | jbot: epic fail. |
13:31.44 | jameswf-home | jbot fist jbot |
13:31.44 | jbot | ACTION uses jbot as a handpuppet |
13:32.11 | blaylock | unified messaging and asterisk 1.6 anybody? |
13:32.38 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
13:32.44 | Katty | lanman |
13:32.59 | [intra]lanman | Katty |
13:33.03 | Katty | how's the wifey? |
13:33.11 | [intra]lanman | hugs Katty |
13:33.13 | Katty | and all your critters |
13:33.13 | [intra]lanman | she's good |
13:33.15 | Katty | hugs intralanman |
13:33.26 | [intra]lanman | ooohhhh, the critters aren't doing so good |
13:33.30 | Katty | oh? :< |
13:33.34 | [intra]lanman | well, at least the chickens |
13:33.42 | Katty | what happened to the chickens? )_= |
13:33.44 | [intra]lanman | all the furries are still cool |
13:33.55 | [intra]lanman | well, in short, the dogs happened to the chickens |
13:34.05 | Katty | oh dear :< |
13:34.10 | [intra]lanman | uh huh |
13:34.18 | Katty | i hope it was quick. |
13:34.53 | [intra]lanman | well, i've seen the dogs play with toys and rip them to shreds in seconds... .so i'm sure it was |
13:34.59 | Katty | nods |
13:35.05 | Katty | i'm getting a GSD soon. |
13:35.09 | coppice | don't eat dog with beans |
13:35.10 | Katty | inside pup, tho |
13:35.21 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
13:35.30 | [intra]lanman | coppice: why? give you gas? |
13:36.01 | coppice | yeah. it messes up your stomache really really badly |
13:36.07 | Katty | also, randomly, (emily elbert)++ |
13:36.20 | gr0mit | begins to get very worried at the direction this conversation is heading |
13:36.23 | [intra]lanman | coppice: thanks, i'll keep that in mind |
13:36.37 | Katty | gr0mit: it's too early for geekery. |
13:36.43 | [intra]lanman | indeed |
13:36.55 | Katty | i still think seanbright stole my caffeine iv. |
13:37.03 | gr0mit | returns to Electronics for Dogs |
13:37.07 | [intra]lanman | this might not be the best place to say this... but i finally moved my voicemail server away from asterisk ;-) |
13:37.08 | Katty | gr0mit: oh! |
13:37.10 | coppice | it would be hard to outdo someone living in east asia for eating strange things :-) |
13:37.18 | Katty | gr0mit: target has a robotic dino |
13:37.26 | Katty | gr0mit: $135ish, it'd be a super dog toy |
13:37.43 | Katty | gr0mit: it wasn't the common white one from best buy and such, either. it was BIG with lights, and fun noises. |
13:37.54 | adr3nalin3 | How does one disable moh? |
13:38.08 | Katty | musiconhold.conf? |
13:38.40 | adr3nalin3 | Katty: thanks for some reason I was looking for moh.conf |
13:38.51 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
13:39.00 | Katty | early morning score for katty! |
13:39.11 | Katty | ...oh. that sounded bad. |
13:39.15 | [intra]lanman | yup |
13:39.23 | Katty | seanbright: iv :< |
13:40.10 | seanbright | i admit nothing |
13:41.40 | gr0mit | ponders an after lunch nap |
13:42.02 | Katty | volunteers seanbright for after lunch nappery. |
13:43.09 | seanbright | i'm down |
13:45.24 | Katty | poodles are so incredibly ugly. |
13:45.38 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
13:46.49 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
13:46.52 | sehh | hey people |
13:47.04 | Katty | hi |
13:47.48 | gr0mit | hello |
13:49.17 | sehh | q: i've got an asterisk server (v1.4.21) which uses a Beronet ISDN card to connect to the ISDN line. It uses mISDN to talk to asterisk. My problem is that some incoming callers can't use the IVR because the DTMF tones aren't recognized. |
13:49.35 | sehh | i've enabled debug,verbose,dtmf mode in logger.conf |
13:50.10 | sehh | and i've successfuly tested the system by calling from a landline and a mobile phone (dtmf tones are recognized and also logged properly) |
13:50.40 | *** join/#asterisk mog (n=mog@nat/digium/x-2924d08d04cd8846) |
13:50.40 | *** mode/#asterisk [+o mog] by ChanServ |
13:51.10 | sehh | unfortunately, the users who complain that the IVR doesn't work are calling from their own ISDN line and when they hit dial keys, my asterisk doesn't even log anything! its as if asterisk can't hear the tones |
13:51.53 | *** join/#asterisk relas (n=hemanuth@port-92-203-120-191.dynamic.qsc.de) |
13:52.37 | gr0mit | sehh |
13:52.50 | gr0mit | are you sure you are receiving DTMF? |
13:53.08 | gr0mit | if you listen via chanspy do you hear the incoming dtmf? |
13:53.09 | relas | hello! I've got two hfc-s isdn-cards with Asterisk 1.4.21.1-BRIstuffed-0.4.0-RC3b. I'm getting this warning: WARNING[2172]: chan_zap.c:2510 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! |
13:53.36 | relas | Why do I get this warning? |
13:53.57 | gr0mit | one of your isdn2 lines is not working? |
13:54.01 | Katty | [TK]D-Fender: where are you this morning? |
13:54.22 | gr0mit | Relas, paste your zaptel.conf and zapata.conf pls |
13:54.31 | relas | gr0mit: mom |
13:54.35 | [TK]D-Fender | Katty: Work, like usual... same sh!t, different day... |
13:54.42 | Katty | [TK]D-Fender: ah right. |
13:54.50 | Katty | [TK]D-Fender: well i hope your wounded thumb is a bit better. |
13:55.12 | sehh | gr0mit, i haven't tried to use chanspy (never used it before) but i'm sure the caller produces tones because i've made a call to my mobile phone for testing and i can hear them fine |
13:55.43 | *** join/#asterisk Specialist1 (n=me@119.160.105.172) |
13:55.43 | Katty | disappears for awhile |
13:55.45 | Specialist1 | hi all |
13:56.03 | relas | http://rafb.net/p/52LpFT55.html |
13:56.15 | sehh | gr0mit, i've also asked the caller to call some local company that uses an IVR on a panasonic PBX and he could navigate their IVR without a problem |
13:57.00 | [TK]D-Fender | Katty: Yes, its vissibly better daily. Also funky when you can FEEL your body repairing itself... |
13:57.43 | gr0mit | Relas, looks ok |
13:57.47 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
13:58.48 | sehh | gr0mit, interestingly when the tones ARE recognized, i see this in the log (when pressing an 8): DTMF[7993] channel.c: DTMF end accepted without begin '8' on mISDN/1-u1389 |
13:58.59 | gr0mit | hmmm, relas, let me pastebin you my config. |
13:59.41 | gr0mit | http://pastebin.com/d77202254 |
14:00.21 | gr0mit | spans 1 and 2 are NT mode, span 3 goes to my ISDN line |
14:00.36 | gr0mit | wonder if it relates to where you are taking your clock from |
14:00.55 | gr0mit | you must take your clock from the public networl |
14:01.12 | hi365 | is there a limit to the ammont of includes voicemail.conf can have? |
14:01.24 | gr0mit | network, but i cant recall what the numbers in span= line do without looking it up |
14:01.39 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
14:02.14 | gr0mit | and i am running the same version of bristuff as you, relas |
14:02.50 | gr0mit | sehh, never used misdn, i use bristuff |
14:03.27 | *** join/#asterisk zydoon (n=zydoon@41.225.153.114) |
14:03.40 | sehh | gr0mit, ok no problem |
14:04.15 | tengulre | anybody here make CallCenter with asterisk ? |
14:04.18 | *** part/#asterisk zydoon (n=zydoon@41.225.153.114) |
14:05.23 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-9728310ef3a85813) |
14:05.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:06.06 | *** join/#asterisk gambolputty (n=BC43599@cpe-24-175-90-49.tx.res.rr.com) |
14:06.13 | *** part/#asterisk gambolputty (n=BC43599@cpe-24-175-90-49.tx.res.rr.com) |
14:07.22 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
14:07.29 | [TK]D-Fender | tengulre: Plenty of us. |
14:08.26 | *** join/#asterisk daveburr (n=dave@208.53.57.91) |
14:11.48 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
14:13.46 | geoff2010 | g729 licensing question: i am making outbound calls which do AMD, which will require a license reservation to perform the audio analysis, once i get an answer i am going to be doing straight g729 pass-thru to another g729 UA, will asterisk give back the license once it is done decoding for AMD? |
14:14.11 | geoff2010 | or will that channel hold onto the license for the duration of the call? |
14:14.14 | [TK]D-Fender | geoff2010: It should |
14:14.28 | [TK]D-Fender | geoff2010: Only when actual decode/encode is required |
14:21.26 | *** join/#asterisk BobPierce (n=WESTMAN\@216.36.132.162) |
14:21.39 | geoff2010 | Fender: Thanks for the help (last night and this AM) |
14:23.38 | gr0mit | any luck, relas? |
14:24.49 | *** join/#asterisk buzzyd (n=psp-man@86.54.239.113) |
14:24.59 | buzzyd | Hi all, |
14:25.00 | *** join/#asterisk ToTo (n=ToTo@207.176.6.103) |
14:25.17 | *** join/#asterisk tftech (n=mjolnir1@63.214.236.169) |
14:26.00 | buzzyd | I'm using the followme app in asterisk 1.4 and it asks push 1 to accept call 2 to reject however I can't do this on my bike headset so anyway to stop it asking that and just connect on answer? |
14:28.39 | tftech | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? |
14:29.53 | *** join/#asterisk mv2 (n=mv2@83.240.229.38) |
14:31.08 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
14:31.30 | mv2 | why asterisk opens two udp ports beside 5060 ? |
14:31.44 | km- | backdoor. |
14:31.52 | km- | it's so digium can hax your gibson. |
14:31.53 | *** join/#asterisk ph0enix (i=disalvo@foster.stonedcoder.org) |
14:32.28 | mv2 | really ? |
14:32.39 | km- | haha, no. What ports are open? |
14:32.58 | mv2 | 4569 |
14:33.01 | mv2 | 2727 |
14:33.11 | km- | 4569 is IAX2, another voip protocol that asterisk supports |
14:33.20 | *** join/#asterisk jpastore (n=jpastore@adsl-69-106-247-50.dsl.pltn13.pacbell.net) |
14:33.29 | mv2 | ok |
14:33.32 | tftech | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? |
14:33.43 | mv2 | and 2727 manager ? |
14:34.02 | km- | nope, 5038 is manager I think |
14:34.05 | BobPierce | I'm just in the process of rebuilding our Asterisk server here at the office. We're currently running 1.4.21.2, but I'm tempted to move to 1.6 for some of the new features. Anyone have any opinions or info on how stable it might be for production use? |
14:34.08 | km- | not sure what 2727 is. H323 maybe? |
14:34.22 | km- | ah, no, MGCP |
14:34.32 | *** join/#asterisk jpastore (n=jpastore@adsl-69-106-247-50.dsl.pltn13.pacbell.net) |
14:34.32 | mv2 | km: ok thanks |
14:34.34 | km- | MGCP is another voip protocol |
14:34.38 | km- | np |
14:36.03 | *** join/#asterisk jpastore (n=jpastore@adsl-69-106-247-50.dsl.pltn13.pacbell.net) |
14:36.13 | JerJer | 4569 is the segfault port |
14:36.56 | km- | guaranteed to segfault your crap when you connect to it? |
14:37.32 | JerJer | well not necessarily a guarantee but pretty damn close |
14:37.41 | km- | hahaha |
14:38.25 | JerJer | http://www.securityscraper.com/pingpoke/iaxControlNew.txt |
14:39.57 | km- | interesting |
14:41.17 | tftech | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info |
14:41.34 | JerJer | km-: try it :) |
14:41.36 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-570d86b7c4963b77) |
14:41.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
14:41.49 | JerJer | pick any version of asterisk - even svn head |
14:42.46 | mv2 | JerJer : not working with me |
14:44.23 | km- | hmm |
14:44.35 | km- | its spamming iax packets to my server but no crash yet. |
14:46.10 | km- | it's up to 40,000 packets |
14:46.19 | km- | is there a certain point where it'll fall over? |
14:49.03 | km- | hehe, there's definitely a memory leak |
14:51.06 | tftech | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info |
14:52.40 | adr3nalin3 | Anybody have any tips on tweaking zaptel settings? I am having some problems with dialing out. I have the patterns correct but the phone co doesn't seem to be registering all the numbers dialed when they are sent out my tdm400p card. |
14:54.00 | adr3nalin3 | about 50% a call will go through the rest of the time I will get a recording from the phone co. That says cannot complete the call as dialed. |
14:54.11 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
14:54.14 | *** join/#asterisk wampie (n=nancyvb@vanbaak.xs4all.nl) |
14:54.37 | *** join/#asterisk propellerhead (n=yogurt2u@190.210.28.193) |
14:54.44 | wampie | mvanbaak, awake yet? |
14:54.58 | glaz | which file is use for echo tuning? |
14:55.04 | tftech | adr3nalin3 - What verison are you using? |
14:55.26 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
14:55.37 | wampie | Hi there btw! Anyone there who is actually on the Astricon? |
14:56.13 | adr3nalin3 | tftech: current stable |
14:56.22 | seanbright | wampie: bunch of people floating around in #astricon |
14:56.36 | adr3nalin3 | tftech: Zaptel 1.4.12.1 |
14:56.38 | wampie | ah thanks! |
14:56.47 | tftech | adr3nalin3 : I had an issue where I had to update the Zaptel drivers to the very latest. I built a box that worked at one place and would not work at another |
14:56.50 | wampie | hadn't checked there yet ;) |
14:57.32 | adr3nalin3 | tftech: were they using "digital" phones from a cable co? |
14:57.53 | tftech | adr : so they are not striaght POTS lines? |
14:58.23 | adr3nalin3 | tftech: they are terminated as such. |
14:58.47 | adr3nalin3 | I was using them without issue with a 3COM analog line card adaptor |
14:58.55 | tftech | adr : Can you call out on a regular analog phone? |
14:58.58 | adr3nalin3 | yes |
14:59.35 | adr3nalin3 | tftech: I will give updating zaptel a shot. |
14:59.42 | relas | gr0mit: I changed it to the same in your config. Seems to work |
14:59.42 | tftech | I know more about telephony than Asterisk. Sorry. I just know I had the issue and had to update the zaptel drivers |
15:00.01 | adr3nalin3 | tftech: thanks for your help |
15:00.13 | tftech | wish I could help more |
15:03.31 | gr0mit | relas, cool! |
15:03.51 | *** join/#asterisk brodiem (n=brodiem@67.18.114.226) |
15:06.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
15:06.18 | *** part/#asterisk BobPierce (n=WESTMAN\@216.36.132.162) |
15:06.35 | relas | gr0mit: I've antoher problem. DTFM don't work. |
15:07.28 | *** join/#asterisk hfb (n=hfb@pool-96-247-117-68.lsanca.dsl-w.verizon.net) |
15:10.17 | gr0mit | relas - aaahh not you as well! |
15:10.23 | gr0mit | dtmf in or out?! |
15:11.29 | relas | in |
15:11.33 | relas | out works fine |
15:11.46 | gr0mit | ok pls paste your zapata.conf file |
15:12.08 | gr0mit | 1 sec let me look back |
15:12.09 | relas | http://rafb.net/p/52LpFT55.html |
15:12.52 | relas | toneduaration=300 did not take any effect |
15:13.33 | gr0mit | let me check my config |
15:14.47 | gr0mit | does it not work for your isdn line or your phone or both? |
15:15.14 | relas | for isdn-line |
15:15.27 | gr0mit | and for your isdn phone? |
15:15.37 | relas | it works |
15:15.46 | relas | for my isdn-phone |
15:16.02 | *** join/#asterisk thor (n=mjolnir1@63.214.236.169) |
15:16.38 | thor | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info |
15:17.53 | gr0mit | so if someone calls your isdn line, it does not get detected under all conditions, or just some? |
15:18.26 | gr0mit | let me paste my config |
15:18.36 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-15-215.red.bezeqint.net) |
15:19.21 | *** join/#asterisk ManxPower (n=manxpowe@5.sub-70-222-245.myvzw.com) |
15:19.29 | gr0mit | http://pastebin.com/d180f80c1 <-- Relas, here is my config for a UK ISDN2 line |
15:22.00 | relas | gr0mit: I'll try your config later. I have to go right now. I will report you on success. |
15:22.10 | thor | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info |
15:22.12 | gr0mit | ok. tot ziens |
15:22.34 | gr0mit | or tchus |
15:22.39 | gr0mit | or wotever it is over there! |
15:22.43 | [TK]D-Fender | adr3nalin3: Most common issue is that * dials the 1st digit too fast and it gets dropped. When dialing out add "ww" before your actual number to delay it 1s |
15:24.14 | adr3nalin3 | [TK]D-Fender: Thanks, I called digium and they told me the same and was fixed immediately! |
15:26.51 | thor | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info |
15:27.45 | blaylock | is anyone using asterisk 1.6 to connect to unified messaging server without sipx? |
15:29.43 | [TK]D-Fender | adr3nalin3: Excellent |
15:32.21 | tftech | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info |
15:33.00 | *** join/#asterisk oilinki (n=oil@ppp-124-120-17-190.revip2.asianet.co.th) |
15:33.05 | [TK]D-Fender | tftech: You should go ahead and pastebin that now. |
15:33.07 | [TK]D-Fender | ~pb |
15:33.08 | 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 |
15:33.57 | tzafrir_laptop | jbot's back! |
15:34.06 | creativx | wb jbot |
15:34.50 | iCEBrkr | umm |
15:35.22 | iCEBrkr | has Cordy's func_mysql stuff been replaced with the app_mysql addon?? |
15:35.33 | tftech | It is not really over 3 lines |
15:35.37 | tftech | just will explain |
15:36.29 | tftech | getting this message: ERROR: Could not open H.323 listener port on 1720 |
15:36.29 | tftech | [Sep 24 04:22:14] ERROR[13127]: chan_h323.c:3170 load_module: Unable to create H323 listener |
15:36.48 | tftech | have installed via SVN today |
15:37.00 | tftech | recompiled 3 times with no installation errors |
15:40.15 | *** join/#asterisk a-s (n=user@89.38.174.194) |
15:40.28 | a-s | how can I install ilbc in asterisk? |
15:40.41 | a-s | I cannot find it in asterisk's sources |
15:41.09 | iCEBrkr | Ahh I found it.. func_odbc |
15:41.58 | Katty | YES! |
15:42.04 | Katty | gsd pups have been CONFIRMED! |
15:42.09 | Katty | [TK]D-Fender: PUPPY!!!! |
15:42.11 | Katty | [TK]D-Fender: THIS WEEKEND! |
15:42.17 | Katty | [TK]D-Fender: I"M GOING TO BE A MOM! |
15:42.38 | Katty | [TK]D-Fender: i hope. they're sending me pictures :> |
15:42.50 | Katty | boingboingboing |
15:43.18 | *** join/#asterisk riksta (n=rick@92.63.131.41) |
15:43.20 | Katty | ..i swear i'm not insane. |
15:43.26 | Katty | looks around |
15:43.48 | creativx | craaaaahaazy |
15:44.03 | riksta | can someone tell me how i can set up an extension in the dialplan which when i call into it, it runs a stop now (or shuts down asterisk) ? |
15:45.24 | *** join/#asterisk aaroninnes (n=aaroninn@im.jobdig.com) |
15:45.37 | *** join/#asterisk bijit (n=benji@200.122.158.243) |
15:46.03 | bijit | how do i get the colors for the cli? |
15:47.47 | aaroninnes | Have you checked google yet? If not I just found this: http://osdir.com/ml/debian.packages.voip.devel/2005-04/msg00151.html |
15:48.16 | bijit | yeah i checked |
15:48.49 | bijit | acually i tried that and it did not wok |
15:48.56 | *** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it) |
15:48.56 | aaroninnes | Nice... |
15:49.27 | aaroninnes | Can someone tell me if it is normal to have errors on cat/proc/interrupts on an asterisk server? |
15:49.45 | a-s | nobody can help me about how to install ilbc please? |
15:50.38 | Specialist1 | any folks have any idea about a good reseller control panel for asterisk |
15:51.03 | a-s | I cannot find its sources here... http://ilbcfreeware.org/ |
15:53.39 | geoff2010 | anyone know anything about "frame type 64"? i am getting the following error after installing some g729 licenses on my server: |
15:53.41 | geoff2010 | chan_sip.c:3707 sip_write: Asked to transmit frame type 64, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
15:54.38 | *** join/#asterisk ana_micho (n=sniper_v@87.236.144.38) |
15:54.45 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
15:54.51 | c4t3l | hello all |
15:55.23 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
15:56.02 | ana_micho | Hi all, I need that someone take a look to http://pastebin.com/m285374ec.....When making a call to a DID number the asterisk server in not entering in DID mode...Any comments? |
15:56.37 | iCEBrkr | Katty: This is like the 100th time you claim to not be insane.. who you trying to fool? |
15:56.41 | iCEBrkr | :D |
15:56.51 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
16:00.48 | MikeJ | Katty isn't insane.. |
16:00.55 | MikeJ | really.. i swear |
16:02.00 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
16:03.36 | ManxPower | ana_micho: You should contact a2billing support |
16:03.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
16:04.02 | ManxPower | geoff2010: "show codecs" |
16:04.07 | ManxPower | or core show codecs |
16:04.24 | ManxPower | aaroninnes: what kind or errors? |
16:05.01 | ManxPower | riksta: exten => 666,1,System(/usr/sbin/asterisk -rx "stop now") |
16:05.06 | *** join/#asterisk ant (n=ant@209.11.127.13) |
16:06.10 | *** join/#asterisk CunningPike (n=arodgers@64.251.77.9) |
16:06.26 | *** join/#asterisk ManxPower (n=manxpowe@5.sub-70-222-245.myvzw.com) |
16:06.48 | ManxPower | oops 8-) |
16:07.01 | aaroninnes | ManxPower: I am getting the info right now, thank you. |
16:07.16 | *** join/#asterisk spokra (n=spokra@gumby.sea0.speakeasy.net) |
16:07.16 | *** join/#asterisk Tomo1657 (n=Tomo1657@63-255-103-7.ip.mcleodusa.net) |
16:07.21 | riksta | ManxPower: thanks, ididnt know you could call System() |
16:07.45 | ManxPower | riksta: "core show applications" tells you all the applications available in YOUR Asterisk install |
16:07.56 | ManxPower | aaroninnes: copy the errors to pastebin.ca NOT TO THIS CHANNEL |
16:08.10 | aaroninnes | Ok |
16:08.59 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
16:13.21 | aaroninnes | ManxPower: here is the pastebin link: http://pastebin.com/m3eafe8f3 |
16:13.56 | ManxPower | aaroninnes: You are either having problems with the PRI line or you are having problems with locked interrupts. What version of Zaptel are you using? |
16:14.38 | ManxPower | BTW, I believe this problem is something that would be included with the free support for your card. |
16:15.54 | bijit | anyone know how I can get the cli colors working? |
16:15.56 | ManxPower | aaroninnes: a Red Alarm almost always means line problems on the telco side. Sometimes it can be a marginal cable. |
16:16.12 | ManxPower | bijit: not me. Seems like something stupid to spend lots of time trying to fix |
16:16.48 | *** join/#asterisk lgj (n=leif@76.7.56.12) |
16:16.53 | *** join/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net) |
16:16.54 | aaroninnes | Ok, thank you very much. We believe the problem was with locked interrupts. I'm looking for the zaptel info now. |
16:17.15 | ManxPower | aaroninnes: I have never actually seen locked interrupts causing a red alarm. |
16:18.20 | ManxPower | ALL Red alarms that I have had to diagnose and fix were caused by the telco side. |
16:19.53 | aaroninnes | ManxPower: zaptel version 1.4.10.1 asterisk version 1.4.20.1 and libpri 1.4.4 |
16:19.56 | bijit | ManxPower: thought it would be fun.. anyhow. What I really want to spend time is in blf. I changed call-limit limitonpeers. But its kind of delayed. Dont know if its the phone firmware. |
16:21.22 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
16:23.10 | a-s | why ilbc is not included in tha latest version of * any more? |
16:23.24 | [TK]D-Fender | a-s: Licensing |
16:24.26 | coppice | ilbc is very messily licenced. its hard to know what you can and can't do with it. this is probably intentional |
16:24.27 | lgj | hello channel, have a sip auth/peer question if anyone has time? |
16:24.51 | ManxPower | ~ask |
16:24.52 | jbot | ask is probably Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
16:25.28 | a-s | [TK]D-Fender: waw! however if I want to compile it , can I? |
16:25.50 | [TK]D-Fender | a-s: Yes, and there is plenty of documentations telling you how |
16:26.01 | ManxPower | a-s: dude, this should be all covered in the Asterisk change log or release notes or upgrade.txt that is included with the Asterisk source code. If it is not then you should file a bug. |
16:26.05 | lgj | anyone know if inbound sip calls which have a peer/user entry matching host and insecure=port, don't match all the time and thus send call to default context even when allow guest calls is no? |
16:26.30 | lgj | seems like it only gets the context when I use the domain=<domain>,<context> setting in global |
16:27.28 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
16:27.53 | aaroninnes | ManxPower: Thanks for the info, I'm a little confused though. I don't see how the telco can cause na IRQ error, plus I feel it is very unlikely that they are having issues with 2 T1 lines. Is it more possible its a PRI issue? |
16:28.07 | lgj | which seems not the right default function of chan_sip |
16:28.26 | ManxPower | aaroninnes: maybe you are having two UNRELATED problems? |
16:28.41 | ManxPower | lgj: Not all that many people use insecure= |
16:29.20 | aaroninnes | ManxPower: That is possible...thank you. |
16:29.22 | lgj | what then is the best way to talk to opensips/openser currently the test is on the same box thus the insecure=port |
16:29.48 | ManxPower | lgj: no idea. |
16:29.53 | lgj | nods |
16:30.30 | lgj | insecure is the issue then i would assume? |
16:31.06 | ManxPower | lgj: best of luck. |
16:31.20 | c4t3l | lgl: have you been over to the openser channel |
16:31.24 | *** join/#asterisk southtel (n=southtel@68-114-19-101.dhcp.gwnt.ga.charter.com) |
16:31.59 | a-s | I understand that ilbc must be compiled as a system library, and asterisk will link something against it... in the past its sources were included in *. is it true? |
16:32.06 | lgj | yeah I got the openser side working perfectly. now down to why asterisk only sends the call to the correct context when I use the global value for domain=<domain>,<context> |
16:32.27 | lgj | seems insecure to do the context via URI domain instead of a defined peer? |
16:32.45 | c4t3l | agreed |
16:33.00 | c4t3l | that's a wierd one |
16:33.10 | *** part/#asterisk southtel (n=southtel@68-114-19-101.dhcp.gwnt.ga.charter.com) |
16:33.30 | lgj | the reason I am asking here is I don't understand why the peer wouldn't match all the time, it matches when an outside call is forwared to asterisk e.g. inbound mainmenu |
16:34.02 | lgj | but when the phone registered with opensips calls the asterisk box it sends it to the default context even when there is a peer with host match ? |
16:34.31 | lgj | i am thinking I overlooked somthing but i wanted to know if anyone had wrestled with this before |
16:34.39 | c4t3l | sounds like some sipheader foo goin on there... |
16:35.04 | lgj | prob so, but it was very late last night and I was tired so I figured I would ask fresh here today |
16:35.14 | c4t3l | i last workd with openser/asterisk over 3 years ago. Do you hang out on this channel often? |
16:35.31 | c4t3l | i know a guy who does this sort of thing exclusively |
16:35.51 | c4t3l | he runs a business on it actually |
16:35.53 | lgj | nope, unfortunatly don't have time to chat usualy, only try to ask informed questions when I cannot get it myself |
16:35.56 | a-s | [TK]D-Fender: can you give me a concretely page where the installation of ilbc is described? |
16:36.03 | a-s | I cannot find its sources |
16:36.51 | lgj | c4t3l, yeah and they changed a LOT from openser to opensips so almost all the docs out there are slightly wrong |
16:37.06 | c4t3l | crap |
16:37.49 | lgj | but I got all my regex and dialplan code working correctly just this wired bug/feature of how the calls are matched to the peer host records |
16:38.06 | lgj | thus it forcing it into default context even when the global context is not default |
16:38.12 | lgj | that is why I came here to chat about it |
16:38.26 | lgj | seems like there may be a wired condition within the code |
16:38.51 | ManxPower | But code is not really discussed on this channel. |
16:38.59 | ManxPower | That's why they created #asterisk-dev |
16:39.17 | lgj | yeah on there but none of the available people want to chat about my issue right now :) |
16:39.22 | c4t3l | lgl: sorry man. I cant raise my buddy on the phone. the only other option i can think of is maybe running in debug logging mode (if not done so already) or perhaps gdb... |
16:39.56 | ManxPower | a-s: please report the lack of docs in the code about ilibc on bugs.digium.com |
16:39.58 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
16:40.18 | lgj | c4t3l, no need to bother anyone on the phone, yeah I will dig deeper into it if I have to. the work around for now with the gobal domain= settings works just don't feel to comfortable about it |
16:40.24 | lgj | er global |
16:40.38 | lgj | thanks for the help people! keep up the good work with asterisk! |
16:40.52 | lgj | l8r |
16:41.08 | qp | [TK]D-Fender: re my connection issue from a couple of days back, we did end up using conference rooms for our inhouse sip softphone, quite handy too as others can "hop" into calls |
16:41.23 | ManxPower | a-s: of course if you had read the CHANGES and UPGRADE.txt you would know everything you need to know about why and how to fix your problem. Shall I also give you line numbers of the information>? |
16:41.23 | qp | (managed transfers) |
16:41.35 | *** join/#asterisk Segnale007 (n=Segnale0@host112-254-dynamic.32-79-r.retail.telecomitalia.it) |
16:46.23 | *** join/#asterisk voxter (n=voxter@63-255-103-7.ip.mcleodusa.net) |
16:49.56 | *** join/#asterisk kbdman (n=carlos@unaffiliated/kbdman) |
16:51.34 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
16:53.01 | *** join/#asterisk wscholar (n=wayne@143.sub-75-209-212.myvzw.com) |
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16:54.07 | StephenF | How do you guys handle teleworkers? hardware VPN tunnel back to *? |
16:54.16 | mort_gib | Hi, is there any way to alert a caller if callee is already in a call. I KNOW i can see if the Channel is in use, but playing a sound file is not really what I want |
16:54.32 | ManxPower | mort_gib: what DO you want? |
16:55.56 | mort_gib | Me?? -To be left alone! But I need to alert a caller to the fact that the "extension" they are calling is busy, without limiting the ingoing/outgoing calls |
16:55.59 | tftech | Hello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info |
16:56.30 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
16:57.09 | ManxPower | mort_gib: you just said you don't want to play a sound file. How do you want to notify the caller? |
16:57.10 | [TK]D-Fender | mort_gib: Well you're calling this parrty (or in the dialplan about to do so), besides playing a sound, what else could yousexpect? |
16:57.32 | ManxPower | puts on his tin foil hat and glares at [TK]D-Fender |
16:58.04 | mort_gib | Dunno, options I guess... |
16:58.25 | [TK]D-Fender | sues ManxPower for infringing his licensed telepathic band-space. |
16:58.38 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
16:58.47 | mort_gib | In the end I have to record the sexy reception girl saying something :-) |
16:58.47 | ManxPower | mort_gib: IIRC ChanIsAvail has an option that will let you know the peer is in use, even if call waiting is enabled on the phone. |
16:59.03 | [TK]D-Fender | mort_gib: Guess you'd better be able to come up with what you expect if you want any kind of help |
16:59.10 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
16:59.46 | mort_gib | I know, I have that working, my client wants, like a busy tone or a different ring tone... |
17:00.03 | ManxPower | mort_gib: THEN PLAY A BUSY TONE. |
17:00.06 | mroth_imm | Anyone know a straightforward way to stop Asterisk from replying to a REGISTER with a '100 TRYING'? |
17:00.06 | ManxPower | this is not rocket science. |
17:00.33 | mroth_imm | It is not playing nice with our SBC. |
17:00.53 | [TK]D-Fender | mroth_imm: "vi chan_sip.c" |
17:01.24 | mort_gib | ManxPower: Not a problem, just wondered... |
17:01.31 | *** join/#asterisk errr_ (n=mike@fedora/errr) |
17:01.39 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
17:01.41 | ManxPower | mort_gib: but you already knew the answer. |
17:01.43 | mroth_imm | Yes, that's quite straightforward. I'm sure it's right in a define. |
17:02.04 | ManxPower | mroth_imm: I imagine not replying would violate the SIP RFCs. |
17:02.26 | [TK]D-Fender | mroth_imm: I'm sure it isn't TOO hard to hunt down, but it is that dirty a job... |
17:02.37 | mroth_imm | From the vendor: 'this a known issue in the 3.5.1 release, as the 100 Trying is not normally provided to the REGISTER method.' |
17:02.43 | ManxPower | Asterisk was never designed to give access to low level SIP stuff. |
17:02.51 | errr_ | when using the manager for QueueStatus it says it takes a parameter of ActionID.. what are the valid actionids? |
17:02.55 | ManxPower | Use OpenSER if you want low level protocol stuff |
17:02.58 | mroth_imm | I think it expects just an OK or a 40[3-4] |
17:03.14 | ManxPower | mroth_imm: 404 means "not found" |
17:03.22 | *** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com) |
17:03.29 | mroth_imm | You get the point, but thanks for flexing you e-muscle. |
17:03.58 | mort_gib | ManxPower: No I did not. I knew ONE way of doing this, but clients ALWAYS come up with another request. That is way we use * because of the flexibility |
17:04.50 | [TK]D-Fender | mort_gib: Flexibility is nice, but knowing what you want is more important. what you want is infinitley |
17:04.52 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
17:04.58 | [TK]D-Fender | dang split line. |
17:05.05 | [TK]D-Fender | mort_gib: You catch my drift... |
17:05.10 | mroth_imm | Anyway, no would've been a good enough response, but I guess that's not in your vocabulary. |
17:05.30 | *** part/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
17:05.57 | mort_gib | [TK]: Yes, I just wondered how other people delt with this.... |
17:06.19 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
17:06.42 | [TK]D-Fender | mort_gib: We personally eith just chove them to VM, or just call the phone. |
17:06.57 | [TK]D-Fender | ok, I'm going to go caffienate some more... |
17:08.00 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
17:08.27 | Katty | MikeJ: ;) |
17:08.32 | Katty | iCEBrkr: well of course i'm not insane. |
17:08.41 | mort_gib | Yeah, which I did initially.... |
17:09.14 | anonymouz666 | app_queue is definitely mad. it reports members as "IN-USE" and that members has a call-limit=99 configured and "show channels" does not show this member in ANY call. |
17:09.31 | Katty | anonymouz666: offer it some chicken salad. |
17:10.31 | anonymouz666 | Katty: heh, how are you? |
17:10.33 | StephenF | So is using something like a Cisco 800 router with a VPN tunnel back to the * box the cheapest way to have a telework's phone connect to the office network? |
17:10.45 | Katty | anonymouz666: starving. also, eating chicken salad. |
17:11.05 | Katty | anonymouz666: there's also promise of getting a puppy this weekend. i'm quite excited. |
17:11.11 | Katty | iCEBrkr: but not insane. |
17:11.27 | iCEBrkr | hehe |
17:11.46 | Katty | i went to petco, and i was looking through the puppy sweaters. |
17:11.59 | Katty | that's how excited i am. |
17:12.13 | Katty | i don't even have pictures of the pups yet |
17:13.22 | *** join/#asterisk krokodilerian (n=vasil@63-255-103-7.ip.mcleodusa.net) |
17:16.19 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
17:16.20 | anonymouz666 | Katty: can I send to the picture of my puppy? |
17:16.23 | puzzled | hi |
17:16.24 | anonymouz666 | you |
17:16.41 | mort_gib | StephenF: I have 8 users that connect via Cisco 877/ADSL lines |
17:16.58 | StephenF | individual users? So they each have an 877? |
17:17.03 | StephenF | or is that a branch office |
17:17.13 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
17:17.16 | puzzled | anyone know which option in Dial(ZAP/<option>/${EXTEN} makes asterisk use the last available line in the group? |
17:17.35 | mort_gib | StephenF: No, it's a branch office, they do Outlook/MS Exchange over the same line :-( |
17:17.44 | mort_gib | 512 kBit :-( |
17:17.45 | StephenF | gotcha |
17:17.47 | StephenF | ouch |
17:18.17 | mort_gib | I would be tempted to use Soekris or similar running OpenBSD |
17:18.18 | krokodilerian | puzzled , yep, use G1 instead of g1 |
17:18.28 | puzzled | krokodilerian: thanks |
17:18.31 | mort_gib | Gives you MUCH better QOS |
17:20.19 | mort_gib | You are going to end up having to deal with explaining to users that limewire and VOIP are not good bed fellows |
17:20.52 | Katty | anonymouz666: SURE! |
17:21.10 | Katty | anonymouz666: i love puppeh pictures |
17:24.15 | mort_gib | StephenF: But you have to keep an eye on latency (of course) |
17:28.12 | Katty | thinkgeek and jinx need to make doggy shirts. |
17:28.49 | Katty | Qwell: i could see a dalmation wearing a More Dots! More Dots! shirt. |
17:28.57 | Qwell | groans |
17:29.54 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
17:30.08 | nny_1 | ManxPower: hey mind if I pm you? |
17:30.30 | Qwell | Katty: I may have to photoshop that later |
17:30.33 | ManxPower | mort_gib: most of the network I manage is 384K |
17:30.38 | Katty | Qwell: :> |
17:30.44 | ManxPower | nny_1: Only if it involves me receiving money. |
17:31.09 | ManxPower | Or something that REALLY is secret, I guess. |
17:32.21 | nny_1 | ManxPower: the former |
17:32.27 | ManxPower | then PM away |
17:34.41 | seanbright | holy crap |
17:34.44 | seanbright | clay aiken is gay!? |
17:34.47 | seanbright | why wasn't i told!? |
17:34.48 | seanbright | heh |
17:35.23 | *** join/#asterisk tkbeat (n=tk@p54B97062.dip.t-dialin.net) |
17:42.03 | Katty | achmed the undead terrorist told me. |
17:42.32 | creativx | i thought he was dead |
17:42.37 | creativx | :o |
17:42.44 | Katty | oh right. |
17:42.55 | Katty | Qwell: Achmed, the undead mage. SILENCE!!! I SHEEP YOU! |
17:43.01 | mort_gib | ManxPower: Yeah, but I HOPE you only run IAX/SIP over it! |
17:43.10 | Qwell | Katty: stop breaking my sheep |
17:43.22 | Katty | Qwell: do you have turtur tome? |
17:43.27 | Katty | Qwell: or just the piggy |
17:43.27 | Qwell | no |
17:43.30 | Katty | :< |
17:43.34 | Katty | there was one on the AH. 4500g |
17:43.35 | Qwell | 5kg >.< |
17:43.44 | Katty | yes, indeed. |
17:44.56 | ManxPower | mort_gib: We don't run any voice over it. |
17:44.57 | mort_gib | ManxPower: I got "lucky" because the two ADSL links are on the same exchange.. Still bandwidth drops like a stone @ 17:00 when kids come home from School |
17:45.06 | mort_gib | Ah, ok |
17:45.13 | mort_gib | So only ?? MS Exchange |
17:45.16 | ManxPower | Stupid to try to run VoIP on a 384K frame relay network |
17:45.24 | Katty | someone said exchange |
17:45.29 | Katty | goes to get rifle. |
17:45.44 | krokodilerian | ManxPower , the world is full with such people |
17:46.08 | ManxPower | krokodilerian: Maybe so, but I don't interact with those people. |
17:46.16 | krokodilerian | ManxPower , people try to run voip on an unguaranteed bandwidth and then complain to us when it fucks up at the moment some idiot is downloading porn from torrent |
17:46.24 | krokodilerian | torrents |
17:46.29 | krokodilerian | ManxPower , then you must live in a happier place :) |
17:46.36 | mort_gib | Yeah, I managed to get some 8-9 users to work over a 512 K link, VOIP and all their dear MS stuff |
17:46.47 | ManxPower | krokodilerian: I just don't currently accept stupid clients. |
17:46.58 | krokodilerian | mort_gib , 729 for the voip? |
17:47.17 | mort_gib | I happen to manage some 35+ networks, I don't EVER get to decide what technology they implement. Sure I get to suggest, but... |
17:47.20 | mort_gib | Yes! |
17:47.28 | mort_gib | G729 for VOIP |
17:47.32 | krokodilerian | ManxPower , how do you know if they're stupid? i'll buy you a beer if you tell me how you do that if it doesn't include talking to them :) |
17:47.56 | krokodilerian | mort_gib , hm, so then the adsl device was not a shitty one |
17:48.09 | *** join/#asterisk stephank (n=urk@82-197-207-120.dsl.cambrium.nl) |
17:48.11 | krokodilerian | we had to explain to a customer that the router they had couldn't push the voip traffic as it couldn't handle the pps |
17:48.23 | ManxPower | krokodilerian: if they want to run voice over a 384K network and I cannot change their mind, then they are stupid. |
17:48.49 | mort_gib | Well... QoS on the Cisco Soho devices are mostly mentioned in the pamflet that comes with the device, for good reason! |
17:49.52 | mort_gib | ManxPower: I changed to consultancy BECAUSE I wanted to retain my right to say "Told you so" and not get a "I don't want any negative attitude" |
17:50.30 | mort_gib | I was quite open about the risk this particular client was (is) running |
17:50.48 | mort_gib | in the end we will setup a WiFi link between offices |
17:50.52 | Katty | [TK]D-Fender: I found a very interesting sign. "German Shepherd Security - We Don't Call 911" |
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17:51.43 | *** mode/#asterisk [+o russellb] by ChanServ |
17:53.54 | Katty | hai russell |
17:54.11 | *** join/#asterisk jewfish (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
17:54.45 | jewfish | Hello. I'm trying to figure out the significance of the underscore in the asterisk dialplan. i.e., what's different about _9NXXNXXXXXX and just 9NXXNXXXXXX? Thanks for any help. |
17:57.50 | ManxPower | jewfish: you need to read the Asterisk book. |
17:57.57 | ManxPower | _ means "pattern match" |
17:57.58 | ManxPower | ~book |
17:57.59 | jbot | well, book is 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 |
17:57.59 | *** join/#asterisk roe_ (n=roe___@216-164-160-45.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
17:58.11 | jewfish | tnx |
17:58.25 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
17:59.59 | roe_ | I know this is a tough question to ask and I have done my research. When comparing the polycom 320 with the linksys sp942 - both of them in about the same price bracket (in the US) the features of the spa942 blow the polycom out of the water. I have to be missing something but I don't know what |
18:01.03 | ph8 | hi guys, I have a SIP hard phone in my room at uni, I need to tunnel it through my main PC (i'm fine setting up tunnels) to the registration server - what ports will the phone be using? Just the one for SIP? |
18:01.08 | ph8 | (what is the port for SIP?) |
18:02.30 | Kobaz | 5060 |
18:02.40 | Kobaz | you're better off using iax |
18:02.41 | ph8 | cheers, i'll try just that and hope that's the only blocked thing |
18:02.50 | Kobaz | much more NAT/firewall friendly |
18:02.53 | ph8 | is IAX a better protocol? |
18:02.56 | ph8 | i see |
18:03.02 | ph8 | it's not my firewall unfortunately or else it wouldn't be a problem :p |
18:03.06 | Kobaz | well, it's better for certain things |
18:03.10 | Kobaz | sip has more features |
18:04.04 | Kobaz | ph8: you can set up a little asterisk box so you can use your sip ahrd phone to asterisk. and then send the call over iax to the remote system |
18:04.08 | Kobaz | or just tunnel the whole entire thing |
18:04.18 | ph8 | oh sorry i do |
18:04.22 | ph8 | asterisk takes the phone by SIP |
18:04.25 | ph8 | then makes outbound calls by IAX |
18:04.28 | ph8 | although it receives calls by IAX |
18:04.30 | *** join/#asterisk BobPierce (n=WESTMAN\@216.36.132.162) |
18:04.31 | Kobaz | okay, that should work nicely |
18:04.38 | ph8 | i'm trying to just get the phone out to my remote * server |
18:04.44 | ph8 | via my desktop i guess |
18:05.04 | Kobaz | so what's the real problem |
18:05.05 | [TK]D-Fender | roe_: Polycom can handle the same number fo calls, and better, has a better build & audio quality, microbrowser support, etc, and costs less. |
18:05.21 | ph8 | Kobaz: What i said, i have a phone i can't connect to my * server because the phone's behind a university firewall |
18:05.33 | ph8 | so 5060 is SIP, will outbound be any different? |
18:05.35 | Kobaz | can you tunnel through it? |
18:05.47 | ph8 | yep |
18:05.51 | ph8 | so i've got one tunnel for 5060 |
18:05.53 | Kobaz | well for sip you need 5060, and then the client port on the way back in |
18:05.54 | ph8 | which i'm just testing |
18:06.14 | ph8 | will the client port be random? |
18:06.16 | Kobaz | which you can force to be within a specific range |
18:06.16 | Maliuta | ph8: don't forget to pass the rtp ports aswell |
18:06.22 | Kobaz | yeah the client rtp ports |
18:06.25 | ph8 | i read RTP, wasn't sure what it is |
18:06.27 | ph8 | is it a range? |
18:06.32 | Kobaz | media / audio |
18:06.37 | Kobaz | you can set a range |
18:06.42 | Maliuta | yeah a range of ports used to do the media trasmission |
18:06.47 | ph8 | not in /etc/services |
18:06.58 | Kobaz | nothing in /etc/services to monkey with |
18:07.12 | ph8 | i would have used it to find the rtp ports :p |
18:07.13 | Maliuta | ph8: because it's user defined |
18:07.14 | ph8 | just googling now |
18:07.27 | Katty | [TK]D-Fender: http://shop.cafepress.com/design/12712455 |
18:07.29 | ph8 | ahhh |
18:07.33 | ph8 | the 'local rtp port' on the phone? |
18:08.03 | [TK]D-Fender | Katty: What do you have by way of pets already? |
18:08.31 | ph8 | hmm |
18:08.34 | roe_ | [TK]D-Fender, While there is a $32 difference, the linksys handles 4lines instead of 2 and the screen is 50% and is backlit |
18:08.35 | c4t3l | is interested to know about this after the dog naming from yesterday |
18:08.35 | ph8 | can i tunnel *in* to my phone like this? |
18:08.39 | ph8 | does that make sense? |
18:09.13 | roe_ | I guess that $32 pays for those extra features |
18:09.28 | c4t3l | ph8: is firewall stateful. keep in mind the problems with some firewall and SIP |
18:09.40 | ph8 | not a clue unfortunately |
18:09.42 | ph8 | one would hope so |
18:10.10 | ph8 | right so i've set my phone to use my machine for sip and 'outbound' (what the phone calls it) and my machine forwards 5060 and 5004 to my * server |
18:10.16 | ph8 | s/forwards/tunnels |
18:10.36 | Katty | [TK]D-Fender: just the ferrets. does the boyfriend count? |
18:10.56 | ph8 | hmm no joy :/ |
18:11.07 | Maliuta | Katty: that depends on what you do to him |
18:11.13 | Katty | Maliuta: erm. |
18:11.17 | Katty | Maliuta: it was a joke. |
18:11.18 | [TK]D-Fender | roe_: SPA supports 4 calls, IP320 supports 2x2=4. Same thing. |
18:11.23 | Maliuta | Katty: does Freenode have a #BDSM? |
18:11.29 | Katty | Maliuta: i wouldn't know. |
18:11.40 | Maliuta | Katty: you say that now |
18:12.03 | Katty | c4t3l: i need a german speaking person to help me finish the name thing :< |
18:12.42 | Maliuta | dog naming? |
18:13.09 | [TK]D-Fender | Katty: Depends... does he have a leash too? ;) |
18:13.25 | Katty | Next!!!! |
18:13.32 | *** join/#asterisk stencil (n=stencil@69-196-130-202.dsl.teksavvy.com) |
18:13.43 | Maliuta | [TK]D-Fender: no, he just has a ball gag and some restraints :) |
18:13.51 | l2trace99 | is there a way to get the qualify value for realtime peers ? |
18:15.07 | l2trace99 | like the from the output from a sip show peer | user ? |
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18:18.11 | nny_2 | ManxPower: quick q that power supply is a regular 48v, i know those units use negative 48. Still works though right? |
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18:27.38 | keith4 | where would I find out how much power a Polycom SIP320 draws, over PoE? I don't see it in the PDF manual |
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18:28.47 | *** part/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net) |
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18:31.53 | ManxPower | nny_2: they are what my customer uses |
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18:35.46 | nny_2 | ManxPower: cool sounds good thanks |
18:36.19 | nny_2 | ManxPower: the telco is testing themselves for echo :\ tonight. I don't trust them, but have asked for full details of the test. |
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18:44.08 | StephenF | keith4: no more than 15.4 W? |
18:45.35 | StephenF | i dunno, you probably need to call them unless someone here has measured it |
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18:47.08 | swampwork | I've got a Polycom 501 that I'm trying to upgrade firmware on. I've never done it before. I am a near-total n00b. I have a tftp server on my internal network (same as the phone's on). I have a .cfg file named the same as the phone's MAC address (0004fd.......cfg) . I've told the phone the IP address of the TFTP server, and verified from another machine that the cfg file can be retrieved via tftp. But the phone insists that there's an error retrievin |
18:47.29 | swampwork | Is there a step-by-step, so-specific-a-monkey-could-do-it walk-through for doing this? |
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18:50.04 | jaytee | so far we haven't been successful in creating a phone provisioning guide that monkeys could use, we have however created a political system so simply screwed that even a chimpanzee can be President. |
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19:01.48 | MindTheGap | hello all, asterisk 1.6.0 rc6 getting choppy moh... there is a digium te110p card locaded and correctly configured (although its running w a loopback connector to clear red signal). zttest shows 99.98 all the time, no irq sharing, no . Playback works fine but moh quality is crappy (playing a .wav file). Does the digium board requires a live connection to generate proper timing? I even tried a realtime kernel but it changed nothing... this server is |
19:03.32 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-17-190.revip2.asianet.co.th) |
19:06.09 | *** join/#asterisk pirulo (n=pirulo@70.56.223.76) |
19:07.23 | MindTheGap | ztdummy also wont change a thing, by the way the kernel is a 2.6.24-19-rt and since ztdummy on 2.6 takes the timer from the rtc i thought it would make the problem dissapear... but it says. |
19:07.42 | MindTheGap | s/says/stays |
19:07.58 | swampwork | jaytee: So you're saying I should just get myself elected president and have someone ELSE provision the phone for me? |
19:08.17 | *** join/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net) |
19:08.50 | swampwork | I was afraid of that. So what's the best provisioning howto/walk-through/source of useful information you would suggest ? Even if it's not monkey-usable. |
19:11.16 | [TK]D-Fender | swampwork: There is a decent guide on the WIKI |
19:11.18 | [TK]D-Fender | ~wikis |
19:11.19 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
19:11.36 | jaytee | I like following the provisioning example in "the book" along with the Polycom SIP Admin Guide and their whitepaper on provisioning which makes it all seem pretty simple once you've read that. I now have provisioning working and I just got through writing a shell script that I just type the name of with the mac address 1st line key extension number etc and it generates my master config file for that phone and the unique settings <mac>-phone.cfg file |
19:12.29 | swampwork | jaytee: Nice! Care to share such shell script goodness? |
19:12.48 | jaytee | sure but it has it's limitations |
19:13.13 | jaytee | right now I've got two scripts I'll be merging, one is for 550's and the other is for 330's. |
19:13.37 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
19:14.04 | jaytee | and it "assumes" that the first two line keys are set to the same sip account. |
19:15.13 | swampwork | jaytee: That's no problem; I'm not going to use it blindly, I'm just looking for reference stuff that's actually *working* somewhere to read and compare to. |
19:17.18 | jaytee | first off, you need to read the section in the book |
19:17.21 | jaytee | ~book |
19:17.21 | jbot | [book] 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 |
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19:19.19 | Maliuta | mm book |
19:19.21 | jaytee | swampwork, read pages 85-93 of the book and reference a couple of the howtos on the wiki for the FTP setup and make sure you're using FTP and not TFTP. |
19:19.33 | keith4 | ~sip phones |
19:19.37 | geoff2010 | does MeetMe transcode all audio streams regardless? i.e. if i have two g729 channels will MeetMe do straight pass-thru, or will it reserve 2 licenses and decode/encode everything? |
19:19.39 | keith4 | ~phones |
19:19.39 | jbot | methinks phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
19:20.43 | keith4 | pets jbot |
19:21.04 | jaytee | ~pb |
19:21.04 | 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 |
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19:24.11 | swampwork | jaytee: Cool. I'm off to read. I was trying to use tftp, but I guess that's not a good idea. I'll fire up an FTP server and get with it. I'm trying to work through the voip-info wiki guide, but the tftp failure has been throwing me off. So I'll see if I can get it working with FTP. |
19:24.29 | jaytee | swampwork, hang on a second |
19:26.20 | [TK]D-Fender | geoff2010: yes, it'll kill 2 licenses, and the mix load on CPU will be more significant |
19:26.21 | ManxPower | geoff2010: You cannot mix G729 audio. All audio must be converted to SLIN for mixing |
19:26.55 | swampwork | jaytee: Yeah? |
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19:27.31 | jaytee | this is a shell file I use to create the config files for a 330. http://pastebin.com/ddbec48 |
19:27.45 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
19:27.59 | xai | Anyone know of a webconferencing suite that uses * ? |
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19:29.21 | swampwork | jaytee: Coolness! Thanks much. |
19:29.24 | jaytee | swampwork, and this is the file I named 000000000000.330 http://pastebin.com/m3e2174e0 |
19:29.41 | swampwork | Excellent! |
19:29.42 | gaetronik | hi there |
19:30.24 | jaytee | swampwork, IMPORTANT. on the second line of the file change the 000000000000.cfg to 000000000000.330 |
19:30.34 | gaetronik | i sometime have agents which stay conected whereas the softphone is closed |
19:30.54 | jaytee | swampwork, the second line of the prepphone.sh script I pastbined first |
19:34.48 | swampwork | OK, got it. |
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19:37.14 | jaytee | swampwork, so I use a different master config template for each type of phone, the sed just replaced certain preset strings for the extension number passed in the second command line parm. |
19:38.26 | jaytee | I'm doing a similar thing with 550's and when I get done I'm merging the two script files into one so I'll just pass it phone type, mac address and lines 1-4 or 1,3-4 in my case since I use the first two line keys for the same number. |
19:40.22 | gaetronik | no one have the same issue? |
19:40.46 | swampwork | Nice. BTW, shouldn't there be a space between "/overrides" and CONTACTS_DIRECTORY in the 000000000000.330 file? |
19:41.10 | swampwork | Is the 550 one significantly different than the 330 one you pasted? |
19:42.25 | gongoputch | anyone use Gizmo ? |
19:43.57 | eric2 | hmm, what's the command to uninstall asterisk? (ya, probably a dumb question) |
19:44.07 | Qwell | make uninstall |
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19:45.03 | SexyKen | Anyone ever setup a Trunk based on IP and no registration for terminating or origination? |
19:45.11 | eric2 | ah yes... thanks Qwell |
19:46.32 | geoff2010 | ManxPower: i guess i was wondering if there was only a single speaker in a MeetMe conference, does it do straight packet pass-thru if both users are G.729 |
19:46.35 | SexyKen | I'm failing at every attempt. |
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19:49.00 | SexyKen | Anyone even use iCAll? |
19:49.08 | ManxPower | geoff2010: Meetme always transcodes AFIK |
19:49.35 | ManxPower | SexyKen: you mean like the permit= and deny= in sip.conf/iax.conf? |
19:51.14 | gaetronik | is this a known isue the issue of agent which look like conected in asterisk whereas the softphone is down? |
19:51.25 | jaytee | swampwork, yes the 550 uses a differnet sip.ld file, Polycom broke them out into smaller files per phone |
19:54.05 | SexyKen | Manx: Maybe that's where I'm messing up? You're supposed to have permit= and deny= ? |
19:54.07 | citywok | does anybody do Asterisk consulting type stuff? I wnat to use the ZapScan/ZapBarge feature to monitor channels in a way that would have to link in some weird external shit, that is over my head |
19:54.34 | ManxPower | SexyKen: Try looking in sip.conf.sample, the first place to look for sip.conf questions |
19:54.54 | ManxPower | citywok: I do consulting, but would not touch that project. |
19:55.18 | citywok | i'm guessing its a few hours of work for somebody that knows what they are doing |
19:55.51 | ManxPower | not if you are using zapscan/zapbarge |
19:56.17 | citywok | i can already use it to camp on a line, but i want to be able to build a "pool" of people to scan |
19:57.15 | citywok | I want to bea ble to dial in, and monitor active calls, would it make more sense to use a bridge of some sort, and just bridge in/out calls programmatically from the API? |
19:58.17 | citywok | (i am a call center, and i've built a call recording system, but people want to be able to "camp on" and listen to calls) |
19:58.52 | jaytee | swampwork, this is a template 000000000000-XXXX.cfg file I use for that script for 330's http://pastebin.com/d2eae38ee |
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20:00.41 | brookshire | does anyone have pics of astricon |
20:01.32 | citywok | ManxPower, would it be possible to dial into my asterisk box, and then use a web interface/API to bridge two zap channels together? |
20:01.51 | *** join/#asterisk SamuraiDio (n=diovani@200.180.158.162) |
20:01.53 | SamuraiDio | hi |
20:02.19 | citywok | well, not bridge because i dont want it to be a conference, i want to barge |
20:02.25 | gaetronik | citywok, with a script which make ami call |
20:03.01 | citywok | would it have to be an AMI, or an API call? |
20:03.49 | SamuraiDio | can i have an asterisk cluster (two or more asterisk servers) acting simultaneously (with auto balancing) able to be accessed/used (by users, peers, and cli) in a single address? |
20:05.42 | geoff2010 | does anyone know how asterisk behaves if you have two SIP peers with the same IP address. this is an unregistered peer, just does a sip_poke to determine if it's alive. I am wondering how asterisk decides which peer record is used for inbound calls. |
20:07.36 | swampwork | Well, my problem remains in that the phone refuses to load the 0004f.......cfg . I'm using FTP now, and (again) I've verified that I can ftp in as PlcmSpIp and retrieve that file. But the phone says "Unable to contact boot server, using existing configuration", then a few secs later says "Error loading 0004fblahblah.cfg" |
20:08.01 | swampwork | Which is the same behavior it's given me with the TFTP server as well. So having config files won't help if I can't coax it to connect to the FTP server and load a new one. |
20:09.23 | SexyKen | Hrm - Okay, I got outgoing calls to work - but on incoming, it seems to start this undless loop of calling the DID that was actually supposed to be calling us. |
20:10.02 | citywok | ManxPower, how would i execute ZapBarge(CHANNELNUMBER) from the api/ami, on a specific channel? Say i'm on channel 17, and i want to Barge in on channel 2 |
20:11.13 | *** join/#asterisk CrazyTux (n=root@ip68-111-67-4.oc.oc.cox.net) |
20:12.29 | swampwork | sunuvagun - it's doing *something* now . Apparently it was so hosed that it had forgotten the "default" FTP username/passwd combo that the docs said to use. I put them in and now it's doing things to itself. I'll hope they're healthy things |
20:14.13 | jeev | http://rawr.pastebin.com/d39a3318f anyone know what i could do? |
20:14.27 | jeev | i guess just add it after |
20:16.26 | *** join/#asterisk MindTheGap (n=MindTheG@189.67.233.134) |
20:17.35 | *** join/#asterisk krokodilerian (n=vasil@63-255-123-206.ip.mcleodusa.net) |
20:18.24 | *** join/#asterisk MindTheGap_ (n=MindTheG@201.80.60.227) |
20:18.44 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-39-82-255-22-143.adsl.proxad.net) |
20:25.00 | citywok | does anybody know how i would execute zapbarge on an open zap channel from the api? |
20:25.01 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
20:26.35 | *** join/#asterisk nny_2 (n=Scott@64.253.24.93.dyn-e-pool33.pool.hargray.net) |
20:26.56 | AlexTO | Hi, i'm making and new instalation of asterisk 1.4 on debian and it is fine but on my /etc/asterisk folder the user and group after the installation is root:root and usuarlly is asterisk:asterisk , so i wonder to know if it fine or not? thanks |
20:27.22 | nny_2 | i have a pri channel in an alarm state and a tech heading over to test the card with a loopback cable. Is there anything else I can do to see if this is the telco having issues or asterisk? |
20:27.44 | nny_2 | PRI span 1/0: Provisioned, In Alarm, Down, Active |
20:27.48 | nny_2 | is what it says right now |
20:27.49 | *** join/#asterisk jplank (n=GBove@158.sub-75-208-68.myvzw.com) |
20:28.04 | nny_2 | wcte12xp: Setting yellow alarm |
20:29.24 | nny_2 | willing to paypal anyone that wants to help |
20:29.59 | *** join/#asterisk CrazyTux (n=root@ip68-111-67-4.oc.oc.cox.net) |
20:30.00 | citywok | you could build a loopback cable yourself, and tell them to try and loop it from there end |
20:30.21 | citywok | loopback cable = rj45 connector with a couple wires jumping the pins |
20:30.22 | nny_2 | yeah someone is heading over there right now to hook the loopback up and see if the alram clears |
20:30.30 | nny_2 | yeah made one already |
20:30.38 | citywok | just do that, and they dont need to send anybody over |
20:30.39 | citywok | :-) |
20:30.49 | citywok | they can test it from their CO |
20:30.55 | *** join/#asterisk xenonex (n=xenonex@89.218.233.88) |
20:31.45 | ManxPower | nny_2: RED or YELLOW alarm? |
20:31.51 | nny_2 | ManxPower: Yellow |
20:31.56 | nny_2 | ManxPower: according to dmesg |
20:32.02 | ManxPower | nny_2: no red alarms? |
20:32.02 | citywok | then its a signalling problem? |
20:32.13 | ManxPower | yellow is not a signalling |
20:32.33 | nny_2 | Detected alarm on channel 5: Red Alarm |
20:32.36 | ManxPower | nny_2: I would have the telco "do a loop test to the smartjack" |
20:32.38 | nny_2 | from asterisk messages |
20:32.51 | nny_2 | tech is on site going to test via loopback cable |
20:33.00 | citywok | ManxPower, is it possible to send an extension to a context from API? |
20:33.10 | ManxPower | RED alarm is line problems that SHOULD be seen by the telco when they do the loopback to the smartjack. You don't do anything for this. |
20:33.18 | citywok | i.e. send zap/13-1 to context asdfasdfasdf |
20:33.19 | nny_2 | they said the card is bad |
20:33.23 | nny_2 | the digium card |
20:33.26 | ManxPower | in fact, you don't even have access to the wiring required for a loop test |
20:33.41 | ManxPower | nny_2: then they will find no problems when the telco does that |
20:33.45 | krokodilerian | nny_2 ,yellow means you don't have asterisk running |
20:33.50 | krokodilerian | or not configured right |
20:33.57 | nny_2 | krokodilerian: explains the dmesg, as i restarted both |
20:33.58 | krokodilerian | so either zapata.conf or asterisk |
20:34.05 | krokodilerian | well |
20:34.19 | krokodilerian | just run asterisk, ztcfg -vv once, and see what does asterish say on zap show status |
20:34.59 | nny_2 | http://pastebin.com/m7498440c zapata.conf |
20:35.31 | nny_2 | Wildcard TE122 Card 0 RED 1 0 0 |
20:35.37 | krokodilerian | hm |
20:35.41 | krokodilerian | RED is another thing :) |
20:35.53 | nny_2 | yeah dmesg said yellow, asterisk says red |
20:36.04 | krokodilerian | what does zttool say |
20:36.12 | krokodilerian | or hm, /proc/zaptel/ ... whatever that was |
20:36.19 | jplank | without starting a flame war (and if I am, please stop me) could someone explain the pros and cons of digium cards vs sangoma cards? |
20:36.26 | nny_2 | red |
20:36.31 | nny_2 | has a loop option |
20:36.44 | krokodilerian | jplank , sangoma - better hardware, bigger price, a bit weird drivers |
20:37.08 | krokodilerian | jplank digium - not so good hardware, i think cheaper, somewhat better support in zaptel/dahdi |
20:37.15 | nny_2 | IRQ Misses: 1 Total/Conf/Act: 24/ 24/ 0 |
20:37.31 | nny_2 | bipolar says 0, rx/tx levels say 0 |
20:38.04 | nny_2 | can i use the loop feature of zttool to test it that way |
20:38.25 | krokodilerian | well, i think you also needed someting on the other side to see the loop,etc. |
20:38.40 | jplank | krokodilerian: the sangoma guy I was talking to said dahdi works with their cards, but the prefer netborder |
20:38.45 | nny_2 | i have a tech heading over now and on hold for digium support |
20:39.10 | krokodilerian | the checklist for pri problems goes like this - first, is the cable ok, is the zaptel.conf right (signalling parameters) and if those are ok, you either get YEL or OK |
20:39.12 | Katty | SexyKen: moooooo. |
20:39.21 | krokodilerian | or, of course , the card might be fucked up |
20:39.40 | *** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4) |
20:39.40 | krokodilerian | jplank netborder ? |
20:39.41 | jplank | just bothers me to install a media gateway ontop of asterisk |
20:39.51 | nny_2 | if i loop back test and get green does that clear the card? |
20:39.58 | jplank | netborder.com |
20:40.17 | krokodilerian | oh, their weird driver? |
20:40.29 | jplank | its a software media gateway that works as a driver, and makes Asterisk think its just a SIP trunk |
20:40.38 | jplank | or multiple sip trunks whatever the case maybe |
20:40.41 | *** part/#asterisk CrazyTux (n=root@ip68-111-67-4.oc.oc.cox.net) |
20:41.00 | krokodilerian | why the hell would I want to see my PRI card as a SIP trunk, that's almost retarded |
20:41.34 | jplank | I asked the same question |
20:41.37 | jplank | funny enough |
20:41.46 | FinboySlick | I need a bit of inspiration here. An application here would require us to pipe a pcm stereo signal over TCP/IP at relatively low latency. I'm sure this is something asterisk can do but it's a tad overkill for my need. IceS2 + Icecast is way too slow (>3sec latency), so, any advices? I don't need a whole lot of compression. |
20:42.07 | krokodilerian | FinboySlick , socat/netcat?:) |
20:42.18 | krokodilerian | FinboySlick , should be relatively easy to tune it for latency |
20:42.27 | jplank | basically his answer was it doesn't matter how * sees the trunk, as long as it can communicate with it, and brought up the point * works a lot better as a SIP PBX then a TDM one |
20:42.42 | jplank | then again, he also had a glass of vino in his hand, so who knows |
20:42.45 | FinboySlick | krokodilerian: I've pondered something like that, but what about the endpoints? I think raw PCM might be a tad too big for me. |
20:42.48 | krokodilerian | yes, really, and you do faxing reaaaaally easy with sip and asterisk |
20:42.58 | krokodilerian | FinboySlick , well, it's easy to calculate |
20:43.16 | jplank | thats what I said, but since it wouldn't really be a SIP trunk, faxing wouldn't be a problem |
20:43.25 | jplank | thats what he said at least |
20:43.35 | krokodilerian | jplank , i'll go talk to him again today probably, i spoke with him on running a test for me on one of their BRI cards |
20:43.51 | AlexTO | Hi, i'm making and new instalation of asterisk 1.4 on debian and it is fine but on my /etc/asterisk folder the user and group after the installation is root:root and usuarlly is asterisk:asterisk , so i wonder to know if it fine or not? thanks |
20:43.56 | krokodilerian | jplank , also... how the hell do you use a sip trunk for timing :) |
20:44.18 | *** join/#asterisk BiG_NoBoDy (n=BiG_NoBo@88.223.34.50) |
20:44.32 | BiG_NoBoDy | hy |
20:44.33 | BiG_NoBoDy | (11:26:27 PM) BiG_NoBoDy: does any one knows how to make Asterisk send fax via email when fax is retrieved? |
20:44.33 | BiG_NoBoDy | (11:27:59 PM) BiG_NoBoDy: and how to set up fax server that it should send faxes retrieved via email (postfix) |
20:44.33 | BiG_NoBoDy | (11:28:28 PM) BiG_NoBoDy: using debian and asterisk 1.6 and hylafax and avantfax |
20:44.33 | BiG_NoBoDy | (11:28:36 PM) BiG_NoBoDy: at the moment |
20:44.48 | jplank | funny enough, these are the same questions I asked him |
20:45.20 | krokodilerian | and what did he say about timing? |
20:45.25 | jplank | he said the clocking would be taken care of by the card itself and netborder, since * only sees a SIP trunk, * wouldn't care about the clocking |
20:45.26 | *** join/#asterisk devhen (n=devhen@70-58-99-185.slkc.qwest.net) |
20:45.59 | jplank | and since it would be all sangoma devices (FXO, PRI ect) the timing issue would be taken care of by netborder |
20:46.06 | krokodilerian | so * doesn't need timing at all ?:) wow, i nevr knew that... |
20:46.10 | jplank | * wouldn't ever have to know they are actually TDM devices |
20:47.04 | jplank | if you don't have TDM devices as far as * is concerned, what do you really need the timing for? (taking meetme out of the equation) |
20:47.16 | krokodilerian | jplank , iax trunking |
20:47.32 | jplank | I ran a asterisk server with SIP trunking only for two years before ever having to instal zdummy for timing |
20:47.32 | krokodilerian | jplank , some t38 fax transmitter that needs good timing (that we wrote) |
20:47.37 | krokodilerian | and tons of other stuff |
20:47.57 | jplank | we make it a point not to support t38 |
20:47.59 | krokodilerian | i always tend to have at least ztdummy on my asterisks |
20:48.06 | jplank | and I haven't really messed with IAX all that much |
20:48.17 | krokodilerian | jplank: ... we make a product for that, not really an option :) |
20:48.31 | jplank | yea, I installed ztdummy on ours a couple months ago for meetme |
20:48.33 | jplank | we? |
20:48.34 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
20:48.46 | krokodilerian | jplank , attractel (the thing is called attrafax_ |
20:49.01 | jplank | t38 hardware? |
20:49.01 | krokodilerian | i'll be talking a bit on timing at my talk today (it's in an hour) |
20:49.06 | krokodilerian | software |
20:49.11 | jplank | which conference? |
20:49.44 | jplank | errr |
20:49.45 | krokodilerian | astricon, i'm in cira B |
20:49.48 | jplank | which track should I say |
20:50.22 | jplank | measuring signal quality in a hybrid system? |
20:50.46 | krokodilerian | jplank , yes, although the title is a bit different |
20:50.54 | jplank | I'm just going by the book |
20:51.08 | krokodilerian | and i'm not joachim vanheverzwijn :) we didn't know if i would be able to travel, so that's why it's his name over there |
20:51.23 | jplank | I was going to go to the ss7 one, but I'll come to yours instead |
20:51.51 | krokodilerian | :) |
20:52.05 | krokodilerian | for mine I have the whole talk written, so you can read it afterwards, I don't know if it will be the same for ss7 |
20:52.12 | krokodilerian | and damn, i wanted to attend that one too |
20:52.17 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
20:52.24 | krokodilerian | but not showing for my own talk would be somewhat impolite |
20:52.29 | jplank | lol |
20:52.31 | jplank | somewhat |
20:52.37 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
20:52.37 | *** mode/#asterisk [+o russellb] by ChanServ |
20:53.04 | jplank | so in your personal experience, you'd go which way, digium or sangoma? |
20:53.36 | krokodilerian | jplank , what for, e.g. PRI or TDM, what kind of application? |
20:53.43 | jplank | PRI |
20:53.48 | jplank | or even TDM |
20:54.01 | krokodilerian | i have mostly used digium cards as we sell them, but have seen so many of them dead that it's a bit worrying |
20:54.04 | jplank | for a customer installation |
20:54.53 | jplank | I guess my best bet would be to get one of each and beat them up |
20:54.53 | krokodilerian | digium cards will take less time to setup |
20:54.54 | krokodilerian | pretty much sure about that |
20:54.54 | ManxPower | nny_2: RED means "line not connected" |
20:54.56 | krokodilerian | for the rest... toss a coin :) sangoma is mre reliable |
20:55.07 | jplank | so I've heard |
20:55.08 | krokodilerian | at least hardware wise, on dying ports,etc |
20:55.21 | jplank | reliability is more important then ease of setup |
20:55.39 | krokodilerian | jplank , so then either sangoma or one of the later digium cards |
20:56.02 | jplank | well then, that answers my question :) |
20:56.24 | jplank | </sarcasm> just in case |
20:56.58 | krokodilerian | if you're not paying it, and you won't need the tdm stuff in asterisk for timing, etc, go with sangoma :) |
20:57.06 | krokodilerian | their guy can probably show you how the installation works |
20:57.12 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpfg.cable.mindspring.com) |
20:57.18 | krokodilerian | we can in fact go torment him together later on :) |
20:57.44 | jplank | please |
20:57.44 | jplank | lol |
20:57.44 | VJFROMGT | i need a way to convert SIP to a PRI without a asterisk machine |
20:57.44 | VJFROMGT | can someone tell me of a hardware? |
20:58.47 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
20:59.09 | jplank | krokodilerian: I'm going to hold you to that torment comment |
20:59.21 | *** join/#asterisk neurosys (n=neurosys@c-66-229-91-186.hsd1.fl.comcast.net) |
20:59.23 | krokodilerian | okay:) |
20:59.26 | jplank | lol |
20:59.36 | puzzled | VJFROMGT: look at patton.com |
20:59.47 | jplank | ahhhh |
20:59.49 | jplank | NOT PATTON |
20:59.51 | jplank | please |
21:00.00 | jplank | better off with Adtran |
21:00.00 | nny_2 | http://pastebin.com/m7498440c <-- can someone peep this and see if anything looks out of place for a standard t1 |
21:00.12 | jplank | VJFROMGT: Adtran TA904 |
21:00.25 | puzzled | or cisco |
21:00.37 | nny_2 | krokodilerian: loopback test and patlooptest show green/ no errors |
21:00.45 | jplank | never got to use any of the Cisco IADs, Adtran did the trick |
21:00.52 | jplank | Patton gave us NOTHING but problems |
21:01.03 | jplank | and their config is backwards |
21:01.04 | *** join/#asterisk rasterix (n=IceChat7@80.177.176.254) |
21:01.36 | rasterix | hi is it possible to retrieve config files including white space and comments through the asterisk manager interface? |
21:01.44 | krokodilerian | nny_2 , show also the zaptel.com |
21:01.46 | krokodilerian | conf |
21:02.12 | krokodilerian | zapata looks fine |
21:02.55 | nny_2 | krokodilerian: http://pastebin.com/m5996a5f5 |
21:03.06 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
21:03.46 | nny_2 | krokodilerian: mind you this all worked until 14:30 today |
21:04.01 | krokodilerian | nny_2 ahhaaaa |
21:04.07 | krokodilerian | nny_2 , how many ports do you have on that card |
21:04.13 | nny_2 | krokodilerian: 1 |
21:04.22 | krokodilerian | nny_2 , any replacements handy? |
21:04.26 | nny_2 | krokodilerian: no |
21:04.28 | *** join/#asterisk xenonex (n=xenonex@89.218.233.88) |
21:04.40 | nny_2 | krokodilerian: but all self tests show that it is fine |
21:04.42 | krokodilerian | if it just stops working, most of the time it's an issue with your hardware, the one of the carriers doesn't do that often |
21:05.04 | nny_2 | the carrier has been adjustng things on their ends for other reasons from what we understand |
21:05.29 | *** part/#asterisk BiG_NoBoDy (n=BiG_NoBo@88.223.34.50) |
21:05.56 | krokodilerian | nny_2 , and so there was a technician where, where the machine with the PRI is, or on the other end |
21:06.16 | krokodilerian | they usually have something they can connect on the PRI and see the signal |
21:06.17 | nny_2 | both afaik but they haven't been very helpful |
21:06.33 | nny_2 | krokodilerian: they said the card is transmitting, but the loopback test cleared it |
21:08.20 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
21:08.40 | ManxPower | nny_2: YOUR tech or TELCO tech? |
21:08.40 | nny_2 | krokodilerian: *not transmitting |
21:08.46 | nny_2 | ManxPower: telco tech |
21:08.59 | ManxPower | nny_2: He should have a T-BERD for testing the PRI |
21:09.08 | jplank | what if you plug a loopback into the *? |
21:09.12 | jplank | does the card green up? |
21:09.15 | nny_2 | our tech hooked up a loopback, i restarted ZAP and got green |
21:09.17 | nny_2 | yes |
21:09.30 | jplank | when you connect to the smartjack it stays red? |
21:09.31 | nny_2 | and patlooptest completed 60 second time out no errors |
21:09.34 | nny_2 | yes |
21:09.46 | nny_2 | and the T1 is flashing red regardless of the cable connection to it |
21:09.52 | nny_2 | T1/smartjack |
21:09.54 | jplank | does the alarm light change on the smartjack when you plug the the * in? |
21:10.00 | jplank | got it |
21:10.00 | nny_2 | no it stays red regardless |
21:10.08 | jplank | it doesn't flash or anything? |
21:10.17 | nny_2 | flashes red continuiously |
21:10.35 | jplank | the alarm light on the smartjack flashes red? not solid? |
21:10.38 | nny_2 | yes |
21:10.46 | ManxPower | nny_2: My prediction is that the telco will find nothing wrong, but the problem will be fixed. Either that or the tech will say "Oh!", call the CO and the problem will be fixed. |
21:10.48 | jplank | who's your LEC? |
21:11.03 | nny_2 | ManxPower: I will buy you a beer if you're right, which i suspect you are |
21:11.04 | nny_2 | Embarq |
21:11.06 | nny_2 | :\ |
21:11.09 | rasterix | < thinks manxpower is right |
21:11.09 | jplank | lol |
21:11.21 | jplank | what happens when you plug the loopback directly into the smartjack? |
21:11.23 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
21:11.29 | nny_2 | haven't tred that |
21:11.29 | ManxPower | Red alarms are almost always a telco line/smartjack/mux/etc issue or a bad cable between the smartjack and the telco |
21:11.38 | nny_2 | is that safe to do, it's their equipment |
21:11.42 | nny_2 | replaced the cable and tested it |
21:11.44 | jplank | agrees with ManxPower |
21:11.52 | nny_2 | beer for everyone! |
21:11.56 | nny_2 | and double for me :D |
21:12.08 | jplank | did you plug a loopback into the smartjack? |
21:12.23 | MindTheGap_ | ello all, asterisk 1.6.0 rc6 getting choppy moh... there is a digium te110p card locaded and correctly configured (although its running w a loopback connector to clear red signal). zttest shows 99.98 all the time, no irq sharing, no . Playback works fine but moh quality is crappy (playing a .wav file). Does the digium board requires a live connection to generate proper timing? I even tried a realtime kernel but it changed nothing. ztdummy also wont |
21:12.32 | nny_2 | i can if thats ok, i don't wanna screw with their equipment and create a liability |
21:12.34 | ManxPower | Smartjacks have a telco controlled loopback feature internal to them |
21:12.53 | jplank | ManxPower: yea, but i've seen plenty of cases when the back end of the smartjack was bad |
21:12.54 | jdnWEST | Anyone had any luck getting a Sangoma A 102 Card into a 1U server? |
21:12.55 | ManxPower | MindTheGap_: why have you not reported this as a bug yet? |
21:13.03 | jplank | so when we looped up the smartjack it ran clean |
21:13.33 | jplank | nny_2: you just made a quick loopback plug right? rj45 connector with pins 1-4 2-5? |
21:13.37 | nny_2 | yes |
21:13.45 | jplank | then yea, plug it into the smartjack |
21:13.52 | jplank | see if the alarm light goes out |
21:13.52 | ManxPower | When the telco says "line tests fine, we see your equipment" then I unplug the line and ask them if they see any change. If they don't see a change then they are testing the wrong line (happened to me several times) |
21:14.18 | nny_2 | ManxPower: hahaha |
21:14.26 | MindTheGap_ | ManxPower, because i may be missing something. as i stated, the card has a loopback conenctor to clear alarms, dont know if it will generate the timing asterisk needs while in loopback |
21:14.44 | jplank | ManxPower: we've had the issue where the lec would leave a soft loop up somewhere in their network, so when we try to run to the customers equipment, or loop up their csu, it seems to work, then we send the loop down command and everything starts working magically |
21:15.05 | nny_2 | ManxPower: and these people somehow think they are better than us heh |
21:15.21 | nny_2 | I say all playing fields are level until you make it a point to look stupid |
21:15.37 | jplank | nny_2: put the loopback into the smartjack and tell me what happens |
21:16.11 | jplank | if * greened up when you plugged the loopback into it, I doubt its a problem with the card, at the very least, it should green up |
21:16.16 | jplank | its probably a cable issue |
21:16.19 | jplank | or a smartjack issue |
21:16.34 | nny_2 | jplank: loopback made it red |
21:16.36 | jplank | or your carrier or lec didn't put their cross-connects in (or properly) |
21:16.40 | jplank | solid red? |
21:16.52 | nny_2 | jplank: asking |
21:17.00 | jplank | which lights? |
21:18.19 | nny_2 | jplank: ds1: cable in = green green red amber amber cable out green/red green red loopback = green/red green/red |
21:18.29 | nny_2 | jplank: sorry playing monkey n the middle with my tech on site |
21:18.45 | nny_2 | basically cable out and loopback are the same |
21:18.47 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
21:18.57 | jplank | the lights don't change at all? |
21:19.02 | jplank | with a loopback in |
21:19.06 | jplank | like blink or anything? |
21:19.33 | nny_2 | says t flashes green red |
21:19.38 | jplank | ok |
21:19.42 | *** join/#asterisk knobo (n=bohmersp@148.122.202.168) |
21:19.42 | jplank | thats something :) |
21:20.15 | jplank | whats the label of the light that flashes green/red? |
21:20.20 | jplank | ds1? |
21:20.20 | nny_2 | jplank: stand by my tech is not communicating this well |
21:20.30 | jplank | alarm |
21:20.31 | jplank | ? |
21:20.54 | jplank | realistically, when you plug the lopoback into the smartjack, the alarm should green up |
21:21.23 | jplank | if it doesn't, theres def something wrong with either the smartjack, or at the lec (not necessarily your carrier) |
21:22.03 | nny_2 | jplank: is there a light that specifcally states alarm |
21:22.24 | jplank | yea |
21:22.30 | jplank | what kind of smartjack is it? |
21:22.34 | jplank | adtran? |
21:22.40 | jplank | hyper_whatever |
21:22.44 | jplank | ? |
21:23.01 | nny_2 | jplank: woring on all five light status on 3 different set ups (loopback etc) i will get brand name |
21:23.08 | nhuisman_work | I wish there was a better mechanism for having the same # on multiple phones |
21:23.16 | nhuisman_work | instead of having to make up sip users for each phone |
21:23.33 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
21:23.35 | krokodilerian | nhuisman_work , do a script that generates the config:) |
21:24.22 | nny_2 | jplank: ds1 light is the only one that changes based on cable |
21:24.36 | nny_2 | jplank: Westell |
21:24.40 | jplank | ok |
21:24.45 | nny_2 | case anyways |
21:25.00 | nny_2 | jplank: adtran card |
21:25.04 | jplank | ok |
21:25.13 | jplank | their should be a alarm light |
21:25.32 | nny_2 | jplank: alm |
21:25.32 | jplank | there should also be a ds1 light (which you confirmed) |
21:25.35 | jplank | yea |
21:25.45 | jplank | when you plug the loopback in |
21:25.51 | jplank | does the alarm light do anything? |
21:25.56 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
21:25.57 | jplank | it might flash quickly |
21:26.19 | nny_2 | ok having him retest and confirm light status |
21:26.24 | Katty | [TK]D-Fender: they should make a Kong suitable for children. |
21:26.50 | nhuisman_work | krokodilerian, configuring it isn't the issue, it's just needing to have sip users who somehow represent the X phones that the number rings on |
21:26.54 | nhuisman_work | like X_1 X_2 |
21:27.03 | jplank | if your plugging the loopback into the smartjack and the ds1 light is greening up (which confirms the the smartjack is seeing something being plugged in) but the alarm is staying red |
21:27.12 | nny_2 | jplank: cable out no loopback alm is orange for a sec than red |
21:27.19 | jplank | that means your carrier doesn't have their crosses in |
21:27.20 | krokodilerian | nhuisman_work , e.g. an account per did per phone?? |
21:27.22 | nny_2 | jplank: checking now |
21:27.27 | nny_2 | crosses? |
21:27.39 | jplank | yea, its def a carrier issue |
21:28.04 | jplank | keep your loopback in, and tell them you have a "hard" loop directly into the smartjack and ask them to run patterns to it |
21:28.05 | nny_2 | jplank: with loopback in same thing orange then red |
21:28.09 | nny_2 | ok |
21:28.21 | nhuisman_work | krokodilerian, I have an extension that I need 4 phones to share |
21:28.23 | jplank | assuming your plugged into the right circuit, they wont see their test patterns come back |
21:28.31 | nny_2 | jplank: ds1 goes solid red when loopback is in |
21:28.49 | nhuisman_work | krokodilerian, I already have it configured using usernames like 300_1 300_2 300_3 300_4 for the sip usernamaes |
21:29.05 | nhuisman_work | just complaining ;P |
21:29.11 | krokodilerian | :))) |
21:29.34 | krokodilerian | ok, i'm going to embarass myself in front of some people :) |
21:29.43 | jplank | i'll be there |
21:30.28 | jplank | nny_2: you def need to call your carrier, if your not greening up with a loopback, its not your issue |
21:30.53 | jplank | unless your loopback is bad |
21:30.56 | nny_2 | jplank: alarm cleared on loopback after a minute or two |
21:31.07 | jplank | but you confirmed its not since th * greened up |
21:31.14 | jplank | ok |
21:31.27 | nny_2 | jplank: on T1 |
21:31.39 | jplank | but alm is still red? |
21:31.46 | nny_2 | jplank: now red is gone |
21:31.50 | jplank | ok good |
21:31.58 | nny_2 | jplank: alm is clear and ds1 is green |
21:32.12 | jplank | now make a loopback jack and put it on the end of the cable your plugging into the * |
21:32.16 | jplank | same thing should happen |
21:32.58 | nny_2 | jplank: SWEET CHRIST |
21:33.01 | nny_2 | jplank: it works now |
21:33.05 | nny_2 | just magically started working |
21:33.11 | nny_2 | on t1 hooked to asterisk |
21:33.22 | jplank | either your carrier realized the crosses werent in |
21:33.43 | jplank | or it just took a while for it to sync up (i've seen circuits take 5-10 minutes to green up) |
21:33.48 | jplank | either way good for you |
21:33.50 | jeev | http://rawr.pastebin.com/d39a3318f anyone know what i could do? |
21:34.09 | jplank | now I'm going to sit in on kroko be back in a bit |
21:35.52 | nny_2 | jplank: thanks again i owe you need your email addy |
21:36.21 | jplank | you dont owe me anything, but if you need help gbove@nyigc.com |
21:36.34 | nny_2 | jplank: whats crosses btw? |
21:36.38 | nny_2 | jplank: thank you again |
21:36.41 | jplank | cross connects |
21:36.49 | jplank | digital |
21:36.55 | nny_2 | roger |
21:36.58 | *** join/#asterisk Carlos_PHX (n=Carlos@63-255-103-7.ip.mcleodusa.net) |
21:37.02 | citywok | is it a new t1 nny? |
21:37.22 | citywok | i've not once had a t1 install go without a hitch. Qwest, Verizon, PLDT all screw them up, every single time |
21:37.39 | jplank | hey our turn ups go pretty good :) |
21:37.58 | jplank | most of the time at least |
21:37.58 | citywok | lol our last one we were moving 1 of 4 T1's to another building |
21:38.03 | citywok | they cross patched the wrong one in the middle |
21:38.14 | jaytee | it's always easier when the T1 is already in place and you're just swapping out what it connects to. |
21:38.17 | citywok | so they broke the other companies T1 cuz they cant read circuit ID's |
21:38.31 | jplank | its usually a lec issue not carrier |
21:38.37 | citywok | International T1's are difficult to deal with |
21:38.38 | jplank | lol |
21:38.43 | jplank | sounds about right |
21:38.51 | jplank | t1/e1 sucks |
21:39.01 | nny_2 | ManxPower: jplank carrier says somehow it was our fault and tested for echo (another issue) with a tbird and said it cleared |
21:39.18 | jplank | of course they did |
21:39.23 | jaytee | I once had to deal with Nynex in New Hampshire to straighten out a mess when they put the demarc for the T1 in the laundromat next door. |
21:39.27 | *** join/#asterisk outtolunc (i=outtolun@15-056.143.popsite.net) |
21:39.31 | jplank | you used a magicloopback that fixed it |
21:39.36 | citywok | haha jaytee thats common i think |
21:39.58 | jplank | the lec NEVER installs the circuit in the right place |
21:39.59 | denon | jaytee: so how many million did they rack up in 900# terminations before you got it moved? |
21:40.10 | citywok | how the hell do i bridge 2 channels together, or how do i execute a context on an active channel? |
21:40.18 | denon | or wasn't this a chinese laundry? |
21:40.20 | jplank | we had verizon once place the circuit on the flooor then refuse to mount it |
21:40.37 | citywok | the only thing i can find is bridge, but thats only in 1.6 |
21:40.51 | citywok | i want to execute ZapBarge on an active channel, dont care how i have to do it lol |
21:40.57 | ManxPower | citywok: dude, channels are AUTOMATICALLY bridgbed |
21:41.03 | jaytee | denon, it had never been turned up so $0.00 but I had them move it. Fixing Nynex's mistakes isn't my job, I just point them out and say, "Fix this or we get another carrier" |
21:41.24 | citywok | ManxPower, i want to click a button, and have it change what channel i'm barged into |
21:41.39 | ManxPower | citywok: Zapbarge/Zapscan does not bridge channels |
21:41.48 | citywok | i can zapbarge no problem in my dialplan, but only locally, cant remove exec it |
21:41.57 | ManxPower | citywok: and I want a 21 yr old blond asian chick. We can't always get what we want. |
21:41.57 | citywok | yea, bridge would be both way so you could talk to it, right? |
21:42.21 | citywok | i'll be in the philippines in 2 weeks, i'll get you one if you help me :-) |
21:42.32 | jplank | ManxPower: come to astricon, theres a strip club not to far from the convention that has one :) |
21:42.45 | ManxPower | There are incoming calls and outgoing calls to/from Asterisk. When a call comes into the dial plan and the dialplan executes Dial then the two channels are bridged. |
21:43.04 | citywok | is there any way to execute a command like zapbarge from the API? |
21:43.18 | citywok | tell channel 13 to barge in on channel 27 |
21:43.29 | citywok | as far as i can tell in the api there is no way |
21:43.39 | jaytee | I want, I want!!! that's all they ever say! I want a button I can click on that makes 1000 people I never met in some countries I've never heard of write and mail me a valid check for $9999.99 everytime I click it. |
21:43.40 | ManxPower | citywok: there are like 5 APIs in Asterisk. Which do you want to use? |
21:43.52 | ManxPower | citywok: why do you care what channel the barger is on? |
21:44.25 | citywok | ManxPower, i want to be able to dial into asterisk on my desk phone, and then be able to click a button on a webpage that will figure out what channel i am on, and let me barge in on another channel |
21:44.34 | jeev | gives 5 to ManxPower and gives the finger to jaytee |
21:44.48 | ManxPower | citywok: best of luck with that. The rest of use the dialplan for that |
21:44.49 | citywok | right now i can click a button on a webpage, and have asterisk dial my phone, and barge me in on another call, but i dont want it to call me, i want to call it |
21:44.53 | denon | web page figure out what channel your on? |
21:44.58 | denon | that's a pretty loose coupling .. |
21:45.28 | denon | you're going to have to associate location with zap channel *somewhere* |
21:45.31 | denon | or name or whatever |
21:45.38 | citywok | lol yea, callerid tells me what person called into asterisk |
21:45.43 | citywok | and i know who sits at what extension |
21:45.46 | ManxPower | You've been whining about this on the channel for like the last hour. There's a reason nobody responds to you. |
21:46.13 | Katty | do you think a bean bag would make a good doggy bed? |
21:46.15 | citywok | all i want to know how to do is execute a context, or a command, from the api |
21:46.29 | ManxPower | citywok: STOP USING THE TERM API. |
21:46.44 | denon | presumably you're talking about the manager interface |
21:46.45 | denon | so, say that |
21:46.58 | jaytee | depends on the size of the beanbag and the size of the dog. Usually, if the beanbag is bigger than the dog it'll work. |
21:47.26 | ManxPower | denon: I was thinking AGI until a few mins ago. But maybe he wants to code it in the channel, which would be the C API (come to think of it, that's the only thing that might be called an API" |
21:47.31 | citywok | yes, the manager telnet in interface is what i've been using |
21:48.02 | ManxPower | citywok: you pretty much wasted the entire time you've been asking on the channel because you used the wrong term. |
21:48.08 | jeev | jatee, msg |
21:48.16 | outtolunc | grins |
21:48.18 | ManxPower | citywok: the manager interface is called either "manager interface" or AMI |
21:48.33 | citywok | ManxPower, i'm not a god, i dont know everything, and every once in a while we use the wrong term. it doesnt mean you need to treat us like idiots. |
21:48.39 | jaytee | flips jeev back the finger. "Right back at ya, buddy!" ;-) |
21:48.49 | ManxPower | citywok: then you need to do more reading |
21:48.50 | citywok | thank you for explaining that to me |
21:49.16 | citywok | good ol' RTFM, spent a lot of time donig that trying to find a way to do this |
21:49.36 | citywok | and since i couldn't find a way to do it online after a few hours of searching, i figured i would ask those that presumeably might have an answer |
21:49.55 | ManxPower | citywok: manager.txt in the asterisk doc directory and voip-info.org should have many many AMI examples, just remember some of them will be out of date |
21:49.56 | jaytee | I carry my ATFOT copy more often than Buchanan carries a bible. |
21:50.16 | citywok | the voip-info wiki has a lot of info, but almost all of it is out of date unfortunately, and much of it is barely explained |
21:51.12 | ManxPower | citywok: Hint: ".call files", also an example in the asterisk doc dir. |
21:51.26 | *** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net) |
21:51.28 | jaytee | well, there's always Google! or if you prefer to use the command line there's http://goosh.org |
21:51.40 | nny_2 | ManxPower: they brought a BCM nortel in to test for echo |
21:51.53 | nny_2 | ManxPower: mind you this is a low latency system |
21:52.00 | nny_2 | ManxPower: how freaking evil is that |
21:52.04 | ManxPower | nny_2: Um, the BCM has commercial EC built into it./ |
21:52.13 | jaytee | makes gagging noises and starts dry heaving because someone used the N word with BCM after it. |
21:52.14 | nny_2 | ManxPower: yeah |
21:52.28 | ManxPower | nny_2: you'll NEVER get the telco to admit to an echo problem |
21:53.38 | jaytee | Rule #1, the problem is always at the customer premise equipment, never at our end of the circuit. |
21:54.27 | jaytee | Rule #2, if the problem does appear after testing that it is at the CO end of the circuit, fix or stall according to need and then blame it on the CPE side anyways. |
21:55.42 | jaytee | Rule #3, if standard business hours support contract exists, ensure that circuit testing is done after hours and bill customer for off hours support fees. |
21:55.47 | ManxPower | jaytee: one of my customers went with a couple of BCMs (against my advice) and then went back to installing Asterisk at their offices |
21:56.10 | [TK]D-Fender | rule #3, if we care about the customer('s business) enough and fear losing them simply claim it as a mystery |
21:56.49 | jaytee | ManxPower, they've dumped them from their product line or are going to. Their new systems run linux with their version of Pingtel's sipXecs. |
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22:07.38 | km- | anyone watching dayton's presentation at astricon? |
22:07.43 | km- | http://www.ustream.tv/channel/astricon2008 |
22:09.27 | denon | is |
22:09.31 | denon | (just fired it up now) |
22:14.36 | loompek | you farted? |
22:14.43 | loompek | oh.. sry... |
22:14.46 | loompek | fired .D |
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22:22.22 | *** join/#asterisk SpeedDragon (n=SpeedDra@sm4-84-90-136-254.netvisao.pt) |
22:22.57 | SpeedDragon | i can ask questions here ? |
22:23.15 | denon | yes |
22:23.22 | nosbig | SpeedDragon, depends on the question.... ;-) |
22:23.23 | SpeedDragon | ok |
22:23.34 | SpeedDragon | nosbig , is about asterisk :D |
22:23.44 | nosbig | Asterisk stuff, sure... Meaning of life stuff, we might be less helpful... ;-) |
22:24.05 | SpeedDragon | i have soft phones connection to my private server |
22:24.17 | SpeedDragon | it works inside and outside of network |
22:24.22 | SpeedDragon | works realy great |
22:24.47 | SpeedDragon | i have account in voipbuster and netcall (netcall is a SIP Provider in Portugal) |
22:25.09 | SpeedDragon | i want make call throw both providers |
22:25.14 | SpeedDragon | and recive by one |
22:25.26 | SpeedDragon | but i can't find where i need change to do that |
22:25.34 | SpeedDragon | i have asterisk-gui instaled |
22:25.40 | *** join/#asterisk CunningPike (n=arodgers@64.251.77.9) |
22:25.40 | SpeedDragon | and version 1.4 of asterisk |
22:26.24 | SpeedDragon | can anyone help ? :P |
22:27.18 | [TK]D-Fender | SpeedDragon: Look at the channel topic. GUI's are supported in their own channels, not here |
22:27.59 | SpeedDragon | hum ... ok , but is not a relation gui based question, but i post question there |
22:29.38 | AlexTO | Hi [TK]D-Fender, one quick question, i just install a 1.4 version and the user and group is root root insted of asterisk asterisk, my question is with this version uses root and root? because always after the installation leave asterisk and asterisk |
22:30.09 | [TK]D-Fender | SpeedDragon: then where to change it is extensions.conf |
22:30.31 | [TK]D-Fender | AlexTO: no such thing as "version". |
22:30.35 | SpeedDragon | i dont have that file :X |
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22:31.02 | [TK]D-Fender | Speed then your system is either really screwed up or you aren't looking in the right place. |
22:31.23 | SpeedDragon | /etc/asterisk ? |
22:31.24 | baliktad | some precompiled distributions of asterisk (from debian and ubuntu, for example) automatically set up asterisk to run as user asterisk |
22:31.34 | [TK]D-Fender | SpeedDragon: That is where it normall should be |
22:31.36 | baliktad | but if you compile from source asterisk runs as root by default |
22:31.53 | [TK]D-Fender | ~asterisk-non-root |
22:31.54 | jbot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
22:31.57 | [TK]D-Fender | ^^^ |
22:31.57 | SpeedDragon | i used debian |
22:32.01 | SpeedDragon | but debian is 1.2 |
22:32.13 | SpeedDragon | i download from website |
22:32.14 | SpeedDragon | 1.4 |
22:32.14 | [TK]D-Fender | SpeedDragon: then I suggest you use "find" to located your configs |
22:32.27 | SpeedDragon | i know where the configs are |
22:32.29 | SpeedDragon | sip.conf |
22:32.34 | SpeedDragon | rtp.conf |
22:32.35 | SpeedDragon | etc |
22:32.41 | SpeedDragon | but i dont have extension.conf :X |
22:32.44 | [TK]D-Fender | SpeedDragon: If you hve not extensions.conf your system is crippled |
22:32.53 | [TK]D-Fender | SpeedDragon: Go rebuild your configs. |
22:33.04 | AlexTO | Ok thanks, my OS is debian |
22:33.11 | [TK]D-Fender | SpeedDragon: Actually there is the off-chance that EVERYTHING is build in extensions.ael |
22:33.18 | [TK]D-Fender | SpeedDragon: Do check in there |
22:33.34 | [TK]D-Fender | SpeedDragon: Barring all that go read the book as you're likely going to have to start from scratch |
22:34.06 | jaytee | rsync is your friend |
22:34.17 | jaytee | and so is cron |
22:34.35 | [TK]D-Fender | cp, tar, take your pick |
22:35.30 | jaytee | "gee, I've spent 37 hours getting all this stuff working. Wonder if I should make a copy or leave that till later." |
22:36.07 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
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22:37.05 | boolean12 | What do you guys think about ISPCP? |
22:37.54 | boolean12 | A little off topic, but I was curious. |
22:38.02 | SpeedDragon | [TK]D-Fender sorry, i are dumb .. |
22:38.08 | SpeedDragon | i have extension.conf file :P |
22:38.42 | [TK]D-Fender | SpeedDragon: then go read the book and look at what you're doing. |
22:38.44 | sacitec | hello, I'm working with asterisk 1.4.19 and cisco call manager 6.1 with sip trunk. i have these problem with the calls http://bugs.digium.com/view.php?id=9546 . Any idea if 1.4.22 version fix this bug ? |
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22:39.29 | [TK]D-Fender | sacitec: Go look at the dates. It's more than clear |
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22:46.49 | l2cache | I have an issue with asterisk 1.2.24, I go to do 'show application meetme' and it isn't installed, any way to patch it in there? |
22:51.39 | [TK]D-Fender | l2cache: You need to have zaptel compiled prior to Asterisk |
22:52.02 | [TK]D-Fender | l2cache: Go setup Zaptel and ztdummy, and then rebuild * from scratch |
22:52.31 | sacitec | [TK]D-Fender: on the version seems to be fixed, but i still have the issue, what could be the problem ? |
22:53.05 | [TK]D-Fender | sacitec: You say its so but you haven't shown us anything. If you wish our help, provide the backup |
23:01.24 | *** join/#asterisk jplank (n=GBove@240.sub-75-209-88.myvzw.com) |
23:01.45 | jplank | heh, I just met allison at astricon |
23:02.00 | jplank | very weird talking to her |
23:03.42 | baliktad | weird |
23:04.12 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
23:05.27 | EmleyMoor | How can I check for a caller ID beginning with a certain set of four digits and effectively replace them with 1? An unusual call came in today |
23:06.08 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-570d86b7c4963b77) |
23:06.39 | EmleyMoor | I want to change any caller ID that comes in as "0044XXXXXXXXXX" to 0XXXXXXXXXX |
23:07.46 | EmleyMoor | I think I may know but clarification would help |
23:08.18 | l2cache | I did compile zaptel before asterisk |
23:10.20 | *** join/#asterisk robba (n=robert@203.102.118.155) |
23:11.38 | *** join/#asterisk lyroy (n=lyroy@bas1-montreal02-1096727613.dsl.bell.ca) |
23:11.39 | *** join/#asterisk seanmh (n=seanmh@63-255-103-7.ip.mcleodusa.net) |
23:13.21 | lyroy | Does someone here ever run asterisk inside a VMware Guest... i'm experiencing problems with choppy calls... it seems to be a timing issue with the ztdummy module... my question.. is it possible to run asterisk in a vmware environnement? |
23:14.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
23:15.10 | StephenF | lyroy: I've heard of it being done. Also I was experiencing choppy calls once ,and disabling ztdummy fixed that for me. Of course without ztdummy you can't use iax trunking or meetme |
23:15.27 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
23:15.32 | l2cache | I've always had horrible delay when using VMWare with * |
23:15.48 | StephenF | ^ there you go, might just be a side effect of virtualization... |
23:15.51 | drmessano | http://bringvictory.com |
23:17.27 | StephenF | drmessano: wow... |
23:17.36 | lyroy | thats strange because I can run asterisk inside a Xen guest but not inside vmware wich is a mature vistualization architecture |
23:18.54 | l2cache | god i love that song, thanks |
23:19.55 | drmessano | you're welcome |
23:20.06 | drmessano | It's one of the best. ever. |
23:20.22 | l2cache | I just forwarded that to everyone I know |
23:20.30 | EmleyMoor | Am I right in thinking you can use the string slicers (;n:m) on the left hand side in a conditional? |
23:21.56 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
23:32.48 | sacitec | [TK]D-Fender: here is my issue, when i make a call to cisco call manager via sip trunk, connects ok, but when cisco part put on hold and try to pick it up, the call is hanged on MOH on asterisk side(cisco side can recover the call). Here is the debug of that part of the call |
23:32.49 | sacitec | http://pastebin.com/m28757a01 |
23:32.57 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
23:34.37 | [TK]D-Fender | EmleyMoor: "core show application gotoif" <- |
23:35.04 | [TK]D-Fender | sacitec: Complete CLI SIP debug only. |
23:35.29 | EmleyMoor | Doesn't say I can't, so we'll see |
23:37.38 | lyroy | For those of you that are experiencing some timing issue with asterisk in a vmware guest... here's the answer http://www.pbxinaflash.com/vm/ |
23:37.41 | x86 | [TK]D-Fender: what's that 24- or 48-port rackmount SIP ATA you were talking about a while back? |
23:37.53 | x86 | mediatrix? |
23:37.59 | [TK]D-Fender | x86: AudioCodes MP-124 / Mediatrix 1124 |
23:39.08 | *** join/#asterisk rednode (n=admin@dehghany.demon.co.uk) |
23:39.10 | rednode | Hi |
23:39.17 | rednode | What distro do you guys recommend for asterisk? |
23:39.24 | rednode | and can asterisk be used on windows at all? |
23:39.26 | l2cache | CentOS 4.5 |
23:39.38 | seanbright | minix |
23:39.41 | rednode | is that what asterisk recommends or what you recommend? |
23:39.46 | EmleyMoor | rednode; I use Debian etch but it really is a matter of personal choice |
23:39.46 | [TK]D-Fender | ReDWhichever you are most comfortable administering and can install *'s requirements on. |
23:39.56 | seanbright | GNU/Hurd |
23:39.57 | [TK]D-Fender | ReDNAnd FORGET about * under windows. |
23:40.13 | [TK]D-Fender | kicks seanbright in the nads |
23:40.16 | seanbright | [TK]D-Fender: ... for now |
23:40.34 | seanbright | once i make my first million i am porting asterisk to windows |
23:40.40 | rednode | well we are going to be rolling out asterisk to banks and hedge funds in the city |
23:40.49 | rednode | so im not sure which distro to use to be honest |
23:40.56 | rednode | is there a distro which asterisk actually recommend? |
23:40.59 | seanbright | no |
23:41.01 | seanbright | well |
23:41.02 | rednode | Fedora? Suse? Red Hat? |
23:41.02 | seanbright | no |
23:41.10 | rednode | lol thats abit weird... |
23:41.11 | [TK]D-Fender | rednode: Even saying that brings into question your qualifications for a project that serious. |
23:41.22 | seanbright | for banks i would run openbsd |
23:41.34 | seanbright | but it's not trivial getting asterisk to install there, apparently. |
23:41.37 | seanbright | <-- never tried |
23:41.42 | rednode | im not qualified at all for this project but unfortunatley im the most qualified person in the company, so most likeley ill hire a contractor |
23:42.00 | seanbright | rednode: where are you? |
23:42.02 | seanbright | :) |
23:42.11 | [TK]D-Fender | rednode: Then since you're doling that task out, ask them what they would use. |
23:42.26 | [TK]D-Fender | rednode: Otherwise you're telling your contractor how to do their job. |
23:42.39 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
23:42.45 | seanbright | oh, the UK |
23:42.48 | seanbright | nevermind that. |
23:42.54 | rednode | i c ok, was just curious what asterisk recommends thats all |
23:43.17 | seanbright | rednode: a lot of the asterisk devs run ubuntu |
23:43.19 | seanbright | debian |
23:43.22 | *** join/#asterisk jeff_smoker (n=jeff_smo@ip70-162-238-155.ph.ph.cox.net) |
23:43.25 | seanbright | i run centos |
23:43.25 | *** join/#asterisk jks (i=jks@193.189.93.254) |
23:43.38 | rednode | i c ok thanks |
23:44.07 | jeff_smoker | Does anyone here know a good resource on how to run a gsm phone to asterisk via usb? |
23:44.17 | [TK]D-Fender | rednode: So far the vast majority of all-inclusive installs use CentOS which is RHEL effectively. |
23:44.19 | seanbright | if you want to pay to fly me over and pay me obscene amounts of money to do the work, i would happily come |
23:44.42 | [TK]D-Fender | rednode: that means its quirks are well known and support for those in similar positions is easy to find. |
23:44.54 | seanbright | jeff_smoker: step 1, make sure you have a USB port |
23:44.58 | seanbright | jeff_smoker: step 3, profit! |
23:45.05 | [TK]D-Fender | rednode: That is perhaps the strongest reason to choose it IMO |
23:45.29 | seanbright | with centos 5 i have run into *0* quirks |
23:45.35 | citywok | i'm a debian man myself, never had any problems with it -- getting asterisk installed takes 5 minutes |
23:45.37 | seanbright | it's pretty straight forward |
23:45.43 | seanbright | don't install from packages though |
23:45.49 | [TK]D-Fender | jeff_smoker: go find some software that cares that you plugged in a phone at all. |
23:45.49 | seanbright | real men compile from source |
23:45.57 | citywok | go use gentoo then :-D |
23:46.05 | rednode | unfortunatley looksl ike we will be going with fedora or red-hat to be honest |
23:46.15 | rednode | only distros which I know that you can get commercial support |
23:46.19 | rednode | well suse as well |
23:46.35 | seanbright | centos is effectively redhat |
23:46.37 | l2cache | When running 'make config' in zaptel, I get the error "make: D: Command not found \r\n make: [config] Error 127 (ignored)" |
23:46.45 | seanbright | heh |
23:46.49 | l2cache | any insight |
23:46.50 | lyroy | yum install make |
23:46.53 | rednode | CentOS is based on Red Hat? |
23:46.55 | lyroy | apt-get install make |
23:46.59 | rednode | Who owns CentOS? Red Hat? |
23:47.08 | seanbright | rednode: correct. centos is FOSS. |
23:47.12 | *** join/#asterisk krokodilerian (n=vasil@63-255-123-206.ip.mcleodusa.net) |
23:47.15 | seanbright | rednode: owned by no one. |
23:47.20 | l2cache | lol, never had 'make' not on a system before |
23:47.24 | rednode | FOSS? |
23:47.25 | seanbright | rednode: check out centos.org |
23:47.28 | l2cache | how did all the other make commands work??? |
23:47.31 | seanbright | rednode: google FOSS |
23:47.32 | seanbright | :) |
23:47.53 | seanbright | l2cache: `which make` |
23:48.10 | l2cache | libpri, zaptel, asterisk, addons make, make install |
23:48.14 | drmessano | sudo make me a sandwich |
23:48.29 | drmessano | seanbright: He answered you |
23:48.33 | rednode | free and open source softwar :P |
23:48.39 | seanbright | rednode: ding ding ding |
23:48.44 | seanbright | drmessano: did he? |
23:48.55 | drmessano | [19:48] <seanbright> l2cache: `which make` [19:48] <l2cache> libpri, zaptel, asterisk, addons make, make install |
23:49.09 | drmessano | You asked which make, right? |
23:49.15 | seanbright | ah |
23:49.21 | l2cache | everything worked but the make config |
23:49.32 | seanbright | you should include a ;) or :) so i know when things you are saying are funny |
23:49.49 | rednode | Has linux got a similar version to active directory, which is stable as active directory? |
23:49.51 | drmessano | I refuse to use emoticons |
23:50.03 | seanbright | rednode: samba does AD integration |
23:50.07 | drmessano | rednode: Not even close |
23:50.10 | rednode | ahh ok |
23:50.27 | seanbright | rednode: requires voodoo and goat's blood... but you can make it work |
23:50.33 | l2cache | I still get the 'make config' error I provided before |
23:50.34 | rednode | meh someone needs to write something like AD for linux :P if theres any hope of linux becomning widely used in the commercial road :P |
23:50.35 | rednode | lol |
23:50.36 | krokodilerian | is the guy we were supposed to go tormenting the sangoma engineer around?:) |
23:50.37 | drmessano | He didnt ask if it would TALK to AD |
23:50.42 | seanbright | l2cache: where did you get zaptel from? |
23:50.47 | rednode | thanks for your help guys, ill be pestering u more with questions later :P |
23:50.52 | drmessano | He asked if there was a linux drop in replacement |
23:50.52 | l2cache | gnudialer.org |
23:50.54 | drmessano | and there is not |
23:50.56 | rednode | going to try and learn it myself first |
23:51.02 | seanbright | l2cache: pastebin your *entire* zaptel Makefile |
23:51.04 | seanbright | ~pb |
23:51.04 | 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 |
23:51.04 | rednode | go through the ebook which is on asterisk site etc |
23:51.07 | rednode | cheers again lads |
23:51.11 | rednode | and ladies |
23:51.35 | beek | rednode: Linux has Fedora Directory Server |
23:52.11 | rednode | similar to AD or no where near? |
23:52.45 | beek | It's a great LDAP server. You can tie it to Kerberos. AD is basically those two items with Windows proprietary hooks. |
23:53.03 | beek | For a 'doze network it won't provide the full functionality but it may be enough for your purposes. |
23:53.09 | sacitec | [TK]D-Fender: sorry for the past debug, here it is the sip debug on cli http://pastebin.com/m70ce25fa |
23:53.43 | beek | rednode: http://directory.fedoraproject.org/ |
23:53.53 | rednode | thanks bek will look @ it |
23:54.02 | seanbright | his name is beek |
23:54.10 | seanbright | have some respect for christ's sake |
23:54.11 | seanbright | heh |
23:54.37 | rednode | sorry lol :P |
23:54.45 | seanbright | l2cache: where's my pastebin, sucka? |
23:54.46 | rednode | cant type for shit its 00:54 here :( tired |
23:54.49 | rednode | got work in 5 hours |
23:54.54 | seanbright | rednode: go to sleep |
23:55.03 | rednode | cant rebuilding a server :( |
23:55.07 | rednode | fucking arsehole banks |
23:55.08 | rednode | lol |
23:55.37 | beek | rednode: Working on rebullding a server while tired.... now THERE's a recipe for disaster. |
23:55.44 | seanbright | yeah |
23:55.45 | rednode | lool :P yep |
23:55.57 | rednode | can asterisk be implemented with Avya, Nortel or Etrali switches?? |
23:56.08 | rednode | i mean integrated :P |
23:56.21 | Qwell | do they support sip? |
23:56.23 | Qwell | or pri? |
23:56.40 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
23:56.40 | *** mode/#asterisk [+o mog] by ChanServ |
23:56.49 | rednode | they support SIP as far as i knw |
23:57.00 | seanbright | then yes |
23:57.07 | rednode | and cisco? |
23:57.16 | rednode | not sure if cisco switch supports PRI im assuming it does |
23:57.25 | seanbright | asterisk SIP stack is extensive... |
23:57.28 | seanbright | heh |
23:57.36 | Qwell | cisco sip stack is expensive... :p |
23:57.43 | l2cache | seanbright: http://pastebin.com/d5d0c969d |
23:57.56 | seanbright | rednode: lots of the answers you seek are floating upon the interweb |
23:58.07 | seanbright | rednode: voip-info.org for example |
23:58.15 | rednode | yeh i know soz ill stop asking silly questions :P ill come back when iv fucked up the asterisk test install |
23:58.16 | rednode | lol |
23:58.42 | seanbright | rednode: thats why [TK]D-Fender is here |
23:58.57 | seanbright | rednode: ANY question at all... feel free to ask [TK]D-Fender. day or night. |
23:59.04 | seanbright | rednode: i'll PM you his personal e-mail address. |
23:59.32 | rednode | thanks!!! |
23:59.35 | rednode | is he a contractor? |
23:59.38 | seanbright | i was kidding |
23:59.42 | seanbright | just busting balls |
23:59.44 | rednode | lol |