| 00:00.03 | *** join/#asterisk ttwhy (~tekkno@p4FECF5AB.dip.t-dialin.net) |
| 00:01.33 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 00:01.38 | *** join/#asterisk Caplain (~shayne@61.42.100.97.cfl.res.rr.com) |
| 00:03.09 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:29.01 | timeshell | yes |
| 00:29.08 | timeshell | 1 channel per call. |
| 00:34.38 | Katty | hi |
| 00:37.12 | leifmadsen | spenguin[work]: actually, 1 channel carries one part of a call. 1 call is typically made up of 2 channels |
| 00:37.25 | leifmadsen | A ---------call -------------> B |
| 00:37.44 | leifmadsen | A ---- chan1 ---> Asterisk --- chan2 -----> B |
| 00:40.25 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 00:41.58 | *** join/#asterisk zikiti (~zikiti@216.110.126.192) |
| 00:42.11 | zikiti | Good night |
| 00:42.44 | leifmadsen | good eve |
| 00:42.57 | zikiti | I have been tasked with analyzing Asterisk CDRs form the mysql database |
| 00:43.09 | leifmadsen | backs away slowly, turns, and sprints away |
| 00:43.30 | zikiti | One issue I have is identifying a specific call that would have gone to a calling group |
| 00:43.30 | Katty | gives leifmadsen a lift out of town |
| 00:43.38 | zikiti | :) |
| 00:43.50 | zikiti | there seems to be no way to identify a specif call |
| 00:44.03 | Katty | i was thinking about seeing alice in wonderland |
| 00:44.07 | leifmadsen | zikiti: that may be entirely true -- CDRs are really only useful in the most basic of cases |
| 00:44.09 | Katty | or perhaps crazy heart |
| 00:44.16 | zikiti | Understood |
| 00:44.22 | zikiti | But that's all I have to wrk with |
| 00:44.33 | zikiti | They want to use it to see how the customer service reps are doing |
| 00:44.41 | zikiti | They seem to be not answering the calls |
| 00:44.42 | leifmadsen | zikiti: that's why CEL was created (currently in trunk, available in 1.8 once available sometime around Q3 or Q4 of this year?) |
| 00:44.58 | zikiti | CEL? |
| 00:45.04 | *** join/#asterisk sdake (~sdake@cm-24-121-126-215.flagstaff.az.npgco.com) |
| 00:45.25 | leifmadsen | zikiti: I'd suggest looking at some of the existing CDR analyzing programs, perhaps like queue metrics |
| 00:45.27 | leifmadsen | ~cel |
| 00:45.28 | infobot | [cel] Channel Event Logging |
| 00:45.34 | leifmadsen | oh that's useful... |
| 00:45.38 | spenguin[work] | leifmadsen: the skype module supports 1 channel per licence - that would mean I need to have two licences for incomming and outgoing? |
| 00:45.38 | zikiti | ok |
| 00:45.43 | spenguin[work] | or Im just confused |
| 00:45.46 | leifmadsen | spenguin[work]: yes |
| 00:45.52 | leifmadsen | 1 channel per license |
| 00:45.56 | spenguin[work] | hrm |
| 00:46.00 | leifmadsen | incoming == 1 channel |
| 00:46.02 | leifmadsen | outgoing == 1 channel |
| 00:46.05 | leifmadsen | 1 + 1 = 2 :) |
| 00:46.11 | zikiti | leifmadsen: Is it open source? and are there other open source aps you know of? |
| 00:46.23 | zikiti | I'm interested in how they query the DB |
| 00:46.31 | leifmadsen | zikiti: CDRs are a black art, so you're not going to find anything worth beans that doesn't cost at least something |
| 00:46.37 | zikiti | hmmm |
| 00:46.39 | zikiti | ok |
| 00:46.43 | Katty | well. |
| 00:46.44 | leifmadsen | unless you develop it yourself |
| 00:46.47 | spenguin[work] | leifmadsen: that is, if I need to have simultaneously an incomming and outgoing call to be possible? |
| 00:46.47 | Katty | not that it will be much help. |
| 00:46.55 | Katty | but asterisk-stat does offer /some/ very basic querying |
| 00:46.56 | zikiti | Well essentially that's what I'm doing |
| 00:46.56 | Katty | it's php. |
| 00:47.01 | Katty | infobot: asterisk-stat? |
| 00:47.05 | leifmadsen | zikiti: at which point, you need to create a few scenarios, look at what the CDRs are doing, then program your stats around that |
| 00:47.11 | leifmadsen | ~asterisk-stat |
| 00:47.14 | leifmadsen | not sure if that works... |
| 00:47.27 | Katty | infobot: asterisk-stat is http://areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 |
| 00:47.28 | infobot | Katty: okay |
| 00:47.30 | leifmadsen | ya that one might work |
| 00:47.31 | leifmadsen | http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 |
| 00:47.32 | leifmadsen | haha |
| 00:47.34 | leifmadsen | yay google |
| 00:47.38 | leifmadsen | ~google |
| 00:47.39 | infobot | google is probably http://lmgtfy.com/?q=google |
| 00:47.39 | zikiti | :) |
| 00:47.40 | zikiti | thanks |
| 00:47.42 | Katty | I"M TOO QUICK FOR OLD MAN |
| 00:47.46 | *** join/#asterisk trentcreek (~kvirc@129.113.131.65) |
| 00:47.47 | Katty | also, i cant' type |
| 00:47.48 | leifmadsen | Katty: that's what she said |
| 00:47.53 | leifmadsen | literally |
| 00:47.54 | spenguin[work] | hhaha |
| 00:47.55 | Katty | *hee* |
| 00:48.02 | zikiti | Saw this one earlier |
| 00:48.10 | Katty | i have a ferret staring at me |
| 00:48.12 | leifmadsen | heads off to poor a bath |
| 00:48.14 | zikiti | It's queries are BURIED in cryptic code |
| 00:48.21 | leifmadsen | Katty: it wants to bite you, much like my fiancee |
| 00:48.29 | leifmadsen | s/poor/pour/ |
| 00:48.31 | spenguin[work] | Katty: feed it something |
| 00:48.32 | Katty | don't be redonkulus |
| 00:48.35 | Katty | my ferrets don't bite |
| 00:48.37 | zikiti | Was trying to piece together the queries but... I gave up |
| 00:48.50 | zikiti | ferrets? Biting |
| 00:48.52 | zikiti | ? |
| 00:48.52 | leifmadsen | zikiti: yep, you're probably better off just building them yourself at that point |
| 00:49.01 | zikiti | :) |
| 00:49.03 | zikiti | Ok |
| 00:49.15 | zikiti | Thanks guys |
| 00:49.20 | zikiti | And gals |
| 00:50.21 | leifmadsen | zikiti: I ended up just having to program my own functions in func_odbc to write stuff to the database when things happened |
| 00:50.22 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:50.36 | zikiti | I get you |
| 00:50.37 | leifmadsen | zikiti: then creating my own views to abstract the information out of the DB for me for billing purposes |
| 00:50.52 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:50.55 | trentcreek | drmessano: what was the URL to reset the ATAs on a server via http? |
| 00:50.57 | leifmadsen | so I wrote my own queries, then used those to write to the database. It was pretty straight forward. |
| 00:51.11 | leifmadsen | zikiti: I generally avoid the CDRs where at all possible |
| 00:51.18 | zikiti | I have no choice |
| 00:51.25 | leifmadsen | that is unfortunate |
| 00:51.28 | zikiti | Have to look at historical data |
| 00:51.33 | leifmadsen | gotcha |
| 00:52.03 | *** join/#asterisk Z_God (~julius@wlan237121.mobiel.utwente.nl) |
| 00:52.09 | leifmadsen | well, I'd suggest you simplify the scenario a bit by abstracting how the agents COULD handle calls, then reproducing the scenarios on a test machine, and see what your CDRs look like so you can understand what your data is telling you |
| 00:52.26 | zikiti | ok |
| 00:52.30 | leifmadsen | otherwise you'll just end up with something that resembles spaghetti |
| 00:53.09 | spenguin[work] | leifmadsen: Id should just need 1 skype channel if Im am calling/recieving 1 call to/from skype at any given momment? |
| 00:53.30 | spenguin[work] | its a bit confusing :s |
| 00:55.23 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 00:55.31 | leifmadsen | spenguin[work]: not really. You can either accept an incoming call (1 channel) or you can place an outgoing call (1 channel). To accept a call then forward it out to another Skype location would require 2 channels. |
| 00:55.51 | spenguin[work] | ok I get it |
| 00:55.53 | leifmadsen | of course! |
| 00:56.08 | spenguin[work] | :> |
| 00:56.17 | leifmadsen | if you are placing a call FROM a Skype location (phone) to another Skype end point (another phone) you need TWO channels |
| 00:56.40 | leifmadsen | Skype ---> Asterisk ---> Skype That scenario requires 2 channels |
| 00:57.01 | leifmadsen | Skype (jimmy) ----> Asterisk ----> Skype (jackie) |
| 00:57.11 | spenguin[work] | leifmadsen: ok but I wouldnt need asterisk for that .. |
| 00:57.17 | carrar | what about Skype ---> Asterisk ---> Skype ---> Skype ---> Asterisk ---> Skype ---> Skype ---> Asterisk ---> Skype |
| 00:57.47 | leifmadsen | carrar: 2 licenses per asterisk server in that chain |
| 00:57.51 | carrar | heh |
| 00:57.54 | leifmadsen | :D |
| 00:58.05 | leifmadsen | or 6 licenses if on the same server |
| 00:58.09 | leifmadsen | spenguin[work]: right |
| 00:58.26 | carrar | a forever looping echo |
| 00:58.30 | leifmadsen | spenguin[work]: unless of course you had an IVR you wanted to use before the call got to your other client |
| 00:58.36 | leifmadsen | ok, I'm out! |
| 00:58.41 | spenguin[work] | heh |
| 01:07.22 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 01:10.38 | drmessano | What if the call is half duplex? |
| 01:11.00 | drmessano | Can I get away with 4 calls on 2 licenses using proper slotting techniques? |
| 01:13.04 | *** join/#asterisk dptom (~dptom@c-24-6-168-8.hsd1.ca.comcast.net) |
| 01:13.09 | drmessano | What if I use spread spectrum audio and drop every 10th 100hz of audibility staggered across 10 calls? I hardly think the call quality would suffer much, thought the crosstalk is gonna be a bitch |
| 01:15.07 | drmessano | Nevermind the DSP overhead to process every 10th slice from 100hz to 13,000hz |
| 01:17.12 | drmessano | 130 emulated 100hz filters x 10 calls is going to require a little more than my PII 500mhz, I think |
| 01:24.33 | *** join/#asterisk WhIteSidE (~chatzilla@ip01.anthonys1.simplybits.net) |
| 01:34.48 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 01:42.59 | *** join/#asterisk Caplain (~shayne@97.100.42.61) |
| 01:43.33 | trentcreek | I just entered "core show codecs" and g729 shows up. Are they now including it without having to pay? |
| 01:44.07 | ChannelZ | that doesn't mean you have all those codecs |
| 01:44.14 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 01:44.18 | ChannelZ | it's just showing you the ID's of them all.. |
| 01:44.21 | trentcreek | Then what does? |
| 01:44.28 | ChannelZ | See where it says "It does not indicate anything about your configuration." |
| 01:45.36 | trentcreek | yes, but is does not indicate they are installed or not config/install != |
| 01:48.10 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net) |
| 01:48.24 | KingDavidNYC | Hello everybody!! |
| 01:49.05 | ChannelZ | if you have g729 installed you will have some 'g729' commands and a 'g729 show licenses' |
| 01:50.32 | ChannelZ | 'core show translations' can also give you a little bit of a clue as to what is available |
| 01:50.50 | *** join/#asterisk dzup (~alex@unaffiliated/dzup) |
| 01:50.52 | ChannelZ | any codecs you don't have, you will have -'s all the way across the row |
| 01:51.08 | carrar | WHAT |
| 01:51.12 | trentcreek | ChannelZ: thanks |
| 01:51.13 | carrar | I demand free g729!! |
| 01:52.00 | trentcreek | carrar: there is free g729 |
| 01:52.06 | ChannelZ | no there isn't |
| 01:52.08 | carrar | I know |
| 01:52.13 | ChannelZ | there's an illegally free one. |
| 01:52.17 | carrar | but there are free movies and music too |
| 01:52.17 | KingDavidNYC | channelZ: yes there is |
| 01:52.22 | carrar | doesn't mean it's right |
| 01:52.39 | KingDavidNYC | ChannelZ: it is not illegal |
| 01:52.44 | carrar | A theif will be a theif |
| 01:52.48 | ChannelZ | rolls his eyes - here we go |
| 01:53.35 | hardwire | wot? |
| 01:53.56 | coppice | carrar: movies aren't free. Avatar, for example, costs 3 hours of your life you can never get back |
| 01:53.59 | trentcreek | FREE g729! http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 01:54.06 | *** join/#asterisk dzup (~alex@unaffiliated/dzup) |
| 01:54.10 | carrar | haha coppice, I don't plan on seeing it :) |
| 01:54.13 | ChannelZ | Did the g729 patent expire? |
| 01:54.20 | carrar | I have my 3 hours |
| 01:54.25 | trentcreek | nope |
| 01:54.34 | hardwire | please note that use of those binaries will result in poor sexual performance. |
| 01:54.37 | trentcreek | You can't patent it in the EU |
| 01:54.49 | ChannelZ | Does the website you posted say "for every time we see someone download this, we will pay royalities to the patent holder on your behalf"? |
| 01:54.54 | coppice | trentcreek: wrong |
| 01:56.21 | trentcreek | coppice: okay..VERY HARD |
| 01:56.57 | coppice | there are about 15 patents related to G.729, and I don't know of one that isn't valid in the EU |
| 01:57.39 | carrar | the server is on the moon, it's ok if you download it from the moon |
| 01:58.02 | coppice | nah, the US staked claim to the moon in 1969 |
| 01:58.18 | ChannelZ | "Do you have a flag?" |
| 01:58.29 | Nugget | heh |
| 01:58.31 | carrar | trentcreek |
| 01:58.34 | carrar | You arein the US |
| 01:58.35 | coppice | they do. I saw them plant it |
| 01:58.44 | carrar | you need to follow US Law |
| 01:58.58 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 01:59.06 | trentcreek | not if server is not |
| 01:59.15 | carrar | You think so |
| 01:59.30 | carrar | Did you read that on their web site? :) |
| 01:59.34 | Nugget | I wouldn't want to be on that side of the lawsuit. |
| 01:59.43 | ChannelZ | If you can afford the Internet access to be here, you can afford the 8 fucking dollars to buy a g729 license if you actually need it. |
| 02:00.12 | trentcreek | Actually I can't. I don't pay for any internet access |
| 02:00.35 | ChannelZ | Which is worth it for the time you will save in not having to download some jacked up stolen codec code and compile it into your Asterisk |
| 02:00.46 | TJNII | Phrase of the day: "Extruded cookie log" |
| 02:00.56 | ChannelZ | hmmm |
| 02:01.03 | ChannelZ | I just made one of those half an hour ago |
| 02:01.22 | ChannelZ | (Subway for lunch and all..) |
| 02:01.45 | TJNII | Groan... |
| 02:01.57 | ChannelZ | :) |
| 02:02.33 | ChannelZ | New euphamism. "BTB, I gotta go extrude a cookie log.." |
| 02:04.14 | Akiraa | Is there windows software that can interface with digium/sangoma cards which can turn an existing PC into a voip gateway? |
| 02:04.47 | Akiraa | to be used by a remote asterisk instance, for example |
| 02:06.49 | carrar | Whats windows? |
| 02:06.59 | ChannelZ | Those things birds fly into. |
| 02:07.02 | carrar | You X Window? |
| 02:07.06 | carrar | you mean X window? |
| 02:07.14 | carrar | oh |
| 02:07.15 | carrar | yeah |
| 02:07.18 | carrar | GLASS Windows |
| 02:07.23 | coppice | carrar: windows are a source of pane |
| 02:07.26 | hardwire | moon server? |
| 02:07.30 | Akiraa | I was thinking of using a remote low-power yate or 3cx instance running on an existing pc |
| 02:07.31 | hardwire | where do I sign up? |
| 02:08.27 | carrar | As long as the server is not the US, RULES do not apply and you can do anything you want :) |
| 02:08.28 | coppice | Akiraa: yate and freeswitch both run on windows, but with those you probably don't need * at all |
| 02:08.36 | carrar | heh |
| 02:08.37 | spenguin[work] | hey anyone know about the skype module |
| 02:08.40 | spenguin[work] | for asterisk |
| 02:09.00 | spenguin[work] | since Leif isnt around |
| 02:09.27 | trentcreek | Microsoft Response Point |
| 02:09.29 | Akiraa | coppice: just as a gateway for a few trunk lines |
| 02:13.10 | ChannelZ | I have one channel of it |
| 02:15.21 | ChannelZ | Wow, Skype posted the code for SILK |
| 02:18.11 | spenguin[work] | ChannelZ: do I need multiple licenses for inbound skype calls |
| 02:18.23 | spenguin[work] | say I do inbound as skype -> asterisk -> sip |
| 02:20.10 | ChannelZ | for multiple calls yes |
| 02:20.27 | ChannelZ | 1 channel is 1 call, either in or out |
| 02:24.38 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:25.12 | *** join/#asterisk synch (~d2d46c84@gateway/web/freenode/x-pokoenpmntkanvwv) |
| 02:29.31 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 02:48.01 | *** join/#asterisk Dibri (~gavit@190.98.33.38) |
| 02:55.21 | KingDavidNYC | Anyone here knows how to capture a dtmf in the middle of a conversation? |
| 03:05.43 | *** join/#asterisk digitalirony (~digitalir@my.ass.looks.just.like.your-face.info) |
| 03:07.22 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 03:09.10 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 03:09.48 | *** join/#asterisk WhIteSidE (~chatzilla@ip01.anthonys1.simplybits.net) |
| 03:11.42 | *** join/#asterisk rgsteele (~rgsteele@c-71-230-37-172.hsd1.pa.comcast.net) |
| 03:13.50 | *** part/#asterisk fleebailey33 (fleebailey@unaffiliated/fleebailey33) |
| 03:25.58 | *** join/#asterisk hhkahya (~hulusikah@88.247.127.66) |
| 03:25.58 | *** join/#asterisk Corydon76-dig (peuce@c-69-137-80-31.hsd1.tn.comcast.net) |
| 03:25.58 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 03:26.35 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 03:27.38 | *** join/#asterisk WhIteSidE (~chatzilla@ip01.anthonys1.simplybits.net) |
| 03:28.04 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:28.04 | *** join/#asterisk g0z (~gauze@entropy.tmok.com) |
| 03:28.04 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 03:28.04 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 03:28.05 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 03:28.05 | *** join/#asterisk fors1 (~forsen@pat-tdc.opera.com) |
| 03:28.05 | *** join/#asterisk doneir (~cbrunker@appenp.lnk.telstra.net) |
| 03:28.05 | *** join/#asterisk thehar (thehar@thehar.xmission.com) |
| 03:28.09 | *** join/#asterisk jasonwert (~jasonwert@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 03:33.30 | KingDavidNYC | eeqrq |
| 03:40.08 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:42.25 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net) |
| 03:48.08 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 03:51.45 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 03:59.37 | *** join/#asterisk jasonwert (~jasonwert@97-83-97-13.dhcp.trcy.mi.charter.com) |
| 04:02.36 | *** join/#asterisk manxpower (~ewieling@216.186.151.147) |
| 04:02.52 | manxpower | ~answers |
| 04:02.52 | infobot | methinks answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 04:04.39 | KingDavidNYC | anybody knows how to capture a dtmf in the middle of a conversation? |
| 04:24.29 | Kobaz | features.conf |
| 04:36.10 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-hfappgwwtsnnrkep) |
| 04:36.47 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 04:54.30 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:55.15 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 05:05.41 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 05:15.05 | *** join/#asterisk xpot (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 05:15.30 | *** join/#asterisk JayTee52 (~jforde051@unaffiliated/jaytee) |
| 05:17.57 | *** join/#asterisk TelTek (~overlord@cpe-76-188-232-92.neo.res.rr.com) |
| 05:40.31 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 06:02.19 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 06:02.37 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
| 06:12.29 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 06:17.23 | *** join/#asterisk gnufan (~hardev@76.91.81.190) |
| 06:18.20 | gnufan | hi all.. getting the following in /var/log/asterisk.. is this brute force attack? |
| 06:18.21 | gnufan | WARNING[1206] chan_sip.c: Maximum retries exceeded on transmission 238599168882@25.102.114.119 for seqno 2 (Critical Response) |
| 06:20.03 | gnufan | Also... |
| 06:20.04 | gnufan | Maximum retries exceeded on transmission 856858975040@192.168.3.111 for seqno 2 (Critical Response) |
| 06:23.52 | *** part/#asterisk gnufan (~hardev@76.91.81.190) |
| 06:24.48 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 06:55.32 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 07:16.35 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 07:23.02 | *** join/#asterisk xNinja (~3ed7e632@gateway/web/freenode/x-xtiwiokytmwtttpz) |
| 07:23.49 | xNinja | hello...i am seeking a solution for my work building around 50 users or more |
| 07:23.55 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-xyvcaerjktchsxkh) |
| 07:26.04 | xNinja | if we only need the users to have their calls to offices forwarded to their cell phones and to have sip feature to make their calls freely around the country using their office number |
| 07:26.10 | xNinja | which is the best solution for this? |
| 07:27.21 | xNinja | i got a quotation from a big company to use avaya hardware and it costs around 100,000 which i think is very high price! |
| 07:29.45 | kaldemar | an asterisk box won't even cost you a tenth of that. |
| 07:30.13 | xNinja | yeah but i have some questions if you or someone else may answer ? |
| 07:31.01 | kaldemar | go ahead and ask. |
| 07:31.14 | xNinja | we have a normal phone system in the building with extensions |
| 07:33.05 | xNinja | some offices has direct lines which u can call them directly from outside and some no which mean u have to call the main number and put the extension to their office |
| 07:35.09 | xNinja | so what solution and what hardware we need to have these features atleast: calls to offices will be forwarded to cell phone after ex:5 rings and the users can use SIP option to use their office lines to make calls |
| 07:35.19 | xNinja | may you help me with this ? |
| 07:37.28 | xNinja | my specialist in IT so i hope you tellme how things will work and which products i need to go and read about them |
| 07:40.32 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 07:40.47 | *** join/#asterisk xNinja (~3ed7e632@gateway/web/freenode/x-kdoyixtnkpubmwos) |
| 07:40.56 | xNinja | sorry |
| 07:41.31 | xNinja | so what you think ? |
| 07:42.04 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:42.27 | kaldemar | there's many ways to do things, you can get telephony hardware to connect PSTN or use an ITSP. those things are possible and somewhat easy to implement. |
| 07:42.35 | kaldemar | this is a good read on asterisk: |
| 07:42.38 | kaldemar | ~book |
| 07:42.39 | infobot | [~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 at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 07:46.01 | xNinja | first of all i will need an asterisk server...what about a phone hardware do we need other phones than the ones we use ? |
| 07:48.34 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 07:48.36 | ChannelZ | depends on your hardware and the phones you use |
| 07:48.38 | kaldemar | depends on what you use now. |
| 07:49.41 | xNinja | may you tellme what i have to check to know what exactly we need for that ? |
| 07:50.58 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.191.110) |
| 07:52.51 | kaldemar | the technology the phones use. are they regular POTS phones? if VoIP, what protocol do they use? |
| 07:53.51 | tzafrir | just installed Debian on a used G3 iMac. bogomips: 49 |
| 07:54.26 | tzafrir | (I wanted to have a big endian test system) |
| 07:54.42 | *** join/#asterisk MaliutaLap (~biteme@202.124.75.100) |
| 07:54.59 | kaldemar | get 50 more of those and you'll have a usable cluster, maybe. :) |
| 08:06.46 | xNinja | thanks kaldemar i can say its not voip |
| 08:06.51 | xNinja | pots i guess |
| 08:07.40 | xNinja | unless if there is other than pots and voip |
| 08:08.04 | xNinja | i want to ask the guys in the phone department to take more info |
| 08:08.40 | xNinja | is that all i have to ask pots or voip or other ? is there something else which will be useful ? |
| 08:10.29 | kaldemar | there are digital proprietory protocols common in PBX's that asterisk won't do. |
| 08:11.01 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:11.24 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-qkwjabbvlwngeady) |
| 08:14.47 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:16.55 | xNinja | so i will need a converter for that right ? may you point what i may need for our setup ? |
| 08:18.01 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 08:19.25 | *** join/#asterisk eharris_ (eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 08:19.34 | ChannelZ | What country are you in? |
| 08:19.40 | xNinja | kuwait |
| 08:19.43 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 08:19.44 | *** join/#asterisk Tim_Toady (~moi@188.4.62.188.dsl.dyn.forthnet.gr) |
| 08:19.53 | xNinja | the wired to wired phones are free |
| 08:20.00 | xNinja | also wired to cell are free for both |
| 08:21.14 | xNinja | for sure cell to cell or to wired are not free |
| 08:22.00 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 08:23.53 | *** join/#asterisk Polysics (~Luca@host83-67-dynamic.30-79-r.retail.telecomitalia.it) |
| 08:23.54 | Polysics | hello |
| 08:24.10 | Polysics | calls still have no audio, and we are on two different networks today |
| 08:24.28 | Polysics | both normal home ADSLs, which mean no fancy firewalls |
| 08:31.15 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:32.56 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-wzzpltcmjcaaaxkr) |
| 08:33.06 | *** join/#asterisk sulex (~sulex@88-149-154-95.static.ngi.it) |
| 08:35.30 | *** join/#asterisk MaliutaLap (~biteme@202.124.72.137) |
| 08:39.27 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:42.36 | *** join/#asterisk funtoo_nbu (seb@jet.bayhost.net) |
| 08:46.53 | funtoo_nbu | hail |
| 08:47.14 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 08:56.18 | Polysics | i am having repeated DNS failuers when registering to our VoIP provider, yet the server does resolve the host |
| 08:56.43 | Polysics | sip debug shows 3 registers, followed by no answers |
| 08:58.37 | tzafrir | here are also the translation speeds there. I'm done with it: http://pastebin.ca/1831476 |
| 09:01.36 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 09:09.26 | *** join/#asterisk Poincare (~jefffnode@213.219.184.23) |
| 09:13.13 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 09:14.27 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 09:24.09 | *** join/#asterisk icek (~pisg@intrush.pl) |
| 09:25.26 | icek | hi |
| 09:26.26 | icek | someone can help me ? |
| 09:26.46 | ChannelZ | I don't have any spare change. |
| 09:27.09 | kaldemar | ~ask |
| 09:27.09 | infobot | 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. |
| 09:28.14 | *** join/#asterisk styelz (~yoohoo@2001:5c0:1500:800::1) |
| 09:29.05 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:29.45 | *** join/#asterisk Polysics (~Luca@host83-67-dynamic.30-79-r.retail.telecomitalia.it) |
| 09:29.46 | Polysics | hello |
| 09:29.58 | Polysics | firewall woes aside, is it a bad idea to run Asterisk on a VPS? |
| 09:30.04 | Polysics | especially if it is inside a firewall? |
| 09:30.27 | ChannelZ | Possibly maybe |
| 09:30.59 | ChannelZ | It probably has a lot to do with how you sort out the networking on the virtual machine |
| 09:31.11 | Polysics | i would say the problem lies in being on a firewalled machine |
| 09:31.20 | Polysics | the VPS should not be aproblem if properly done |
| 09:31.24 | *** join/#asterisk Z_God (~julius@wlan229239.mobiel.utwente.nl) |
| 09:31.33 | ChannelZ | Of course |
| 09:31.39 | icek | I`m install asterisk using this FAQ : http://www.pcgmarket.com/wiki/index.php/Asterisk#Configurating_database, and allis fine, i create some mysql user in sip-buddies and when i log to asterisk using X-lite, console say : http://pastebin.com/HDEq0u8i |
| 09:33.05 | Polysics | ChannelZ, can firewall woes on the server cause two SIP clients to be able to call each other but not hear audio? |
| 09:33.26 | Polysics | at least i know what to blame |
| 09:33.59 | ChannelZ | Polysics: Definately |
| 09:34.33 | Polysics | i'd swear but i am trying to remain calm |
| 09:34.34 | ChannelZ | The firewall could be blocking the RTP ports, and/or the devices themselves might be giving out phony IP addresses in such a way that they aren't talking to each other |
| 09:35.11 | Polysics | the devices can connect to another properly configured * and talk, so i would say the clients are 90% not a problem |
| 09:35.23 | Polysics | since the SIP accounts have exactly the same options |
| 09:35.36 | Polysics | as in "i copied the sip.conf block over" |
| 09:36.07 | ChannelZ | That could be bad depending on what options are in it, but ok |
| 09:36.22 | ChannelZ | Without knowing your network topology I can't venture any more specific guesses |
| 09:42.00 | *** join/#asterisk pawz (~pawz@ppp118-208-189-193.lns20.bne4.internode.on.net) |
| 09:50.13 | *** join/#asterisk bidossessi (~bidossess@41.221.23.177) |
| 09:50.29 | icek | I`m install asterisk using this FAQ : http://www.pcgmarket.com/wiki/index.php/Asterisk#Configurating_database, and allis fine, i create some mysql user in sip-buddies and when i log to asterisk using X-lite, console say : http://pastebin.com/HDEq0u8i |
| 09:52.07 | *** join/#asterisk bidossessi (~bidossess@41.221.23.177) |
| 09:53.11 | bidossessi | hi all |
| 09:57.05 | kaldemar | icek: looks like you don't have res_config_mysql from asterisk-addons installed or loaded. what does "module load res_config_mysql.so" in asterisk console say? |
| 09:58.14 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 10:01.37 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:03.01 | icek | module load res_config_mysql.so |
| 10:03.01 | icek | Unable to load module res_config_mysql.so |
| 10:03.01 | icek | Command 'module load res_config_mysql.so' failed. |
| 10:03.01 | icek | [Mar 10 10:59:39] WARNING[12344]: loader.c:386 load_dynamic_module: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: cannot open shared object file: No such file or directory |
| 10:03.05 | icek | [Mar 10 10:59:39] WARNING[12344]: loader.c:781 load_resource: Module 'res_config_mysql.so' could not be loaded. |
| 10:03.59 | *** join/#asterisk sourcode (~code@ppp-58-8-236-217.revip2.asianet.co.th) |
| 10:04.45 | *** join/#asterisk sourcode (~code@ppp-58-8-236-217.revip2.asianet.co.th) |
| 10:10.16 | icek | kaldemar: hmm |
| 10:16.53 | kaldemar | you don't have it installed. go to asterisk-addons dir and do a "make menuselect". under ressource modules, if you see XXX next to res_config_mysql, you don't have its dependencies met. if [ ], check it and re-run make and make install. |
| 10:17.31 | *** join/#asterisk fofware (~chatzilla@186.125.121.165) |
| 10:17.52 | kaldemar | in case of XXX, check that you have a libmysqlclient-dev installed and re-run ./configure and check make menuselect again. |
| 10:21.07 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 10:23.06 | icek | kaldemar: http://pastebin.com/U6JJAMQA i have, check make menuselect -> Resource Modules -> [*] res_config_mysql |
| 10:24.42 | kaldemar | have you run make and make install? |
| 10:24.58 | icek | yes now i run |
| 10:25.16 | icek | <PROTECTED> |
| 10:25.51 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 10:28.03 | icek | i copy res_config_mysql.so to /usr/lib/asterisk/modules |
| 10:28.28 | icek | and : |
| 10:28.28 | icek | module load res_config_mysql.so |
| 10:28.28 | icek | Unable to load module res_config_mysql.so |
| 10:28.28 | icek | Command 'module load res_config_mysql.so' failed. |
| 10:28.29 | icek | [Mar 10 11:25:12] WARNING[17492]: loader.c:771 load_resource: Module 'res_config_mysql.so' already exists. |
| 10:30.12 | icek | http://pastebin.com/PYeK5aGG |
| 10:30.15 | icek | now i have this |
| 10:30.29 | *** join/#asterisk hyphenex (~Adium@115-64-56-198.static.tpgi.com.au) |
| 10:30.49 | hyphenex | is there any way to find out why my asterisk install is eating up 100% CPU? |
| 10:32.17 | Chainsaw | hyphenex: Do you have the core set to verbose 10 & debug 10 please? |
| 10:32.36 | Chainsaw | hyphenex: If so, log into the asterisk console and it should admit why. |
| 10:33.07 | icek | kaldemar: thx now its working |
| 10:34.24 | hyphenex | Chainsaw: how do I do that? (I'm kind of new) |
| 10:34.38 | Chainsaw | hyphenex: asterisk -r |
| 10:34.41 | Chainsaw | hyphenex: core set verbose 10 |
| 10:34.44 | Chainsaw | hyphenex: core set debug 10 |
| 10:45.27 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 10:45.57 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 10:47.09 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 10:57.31 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 10:58.14 | bidossessi | hi all. i'm having a dahdi call-out issue that resembles https://issues.asterisk.org/view.php?id=15429, except that in my case an incoming call doesn't reset the hook status. running asterisk 1.6.0.25, dahdi-linux 2.2.0.2 on Centos5. how could i troubleshoot that? |
| 11:04.00 | *** join/#asterisk xishan (~it@58-65-160-128.nayatel.pk) |
| 11:05.00 | *** join/#asterisk Akiraaa (~Akira@92.81.221.18) |
| 11:09.19 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 11:20.25 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 11:20.45 | *** join/#asterisk nightrid3r (kvirc@41.214.195.207) |
| 11:21.04 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 11:27.41 | hyphenex | Chainsaw: if you're still around, it's still eating 100% cpu on idle |
| 11:28.16 | Chainsaw | hyphenex: Unless you look at verbose output and try to infer more from it, or share it with me on pastebin.ca there is nothing I can do. |
| 11:28.42 | hyphenex | Chainsaw: but there is none :( |
| 11:28.47 | *** join/#asterisk carltonb (~rowlando@213.253.145.9) |
| 11:28.54 | hyphenex | gateway*CLI> core set verbose 10 |
| 11:28.55 | hyphenex | Verbosity was 4 and is now 10 |
| 11:28.55 | hyphenex | gateway*CLI> core set debug 10 |
| 11:28.55 | hyphenex | Core debug was 0 and is now 10 |
| 11:29.34 | hyphenex | <PROTECTED> |
| 11:31.42 | carltonb | hi all. i am looking into building a voip server/gateway which can offer tdm, c7 signalling and a billing module. What is the best direction to go in?? |
| 11:42.43 | tzafrir | hyphenex, in top, press 'H' (shift h) to get separate information for each thread |
| 11:43.01 | tzafrir | this will get you the specific thread that uses the CPU |
| 11:43.12 | tzafrir | (wait for a cycle or two to get accurate data) |
| 11:43.21 | tzafrir | now strace it to see what it does |
| 11:46.01 | *** part/#asterisk carltonb (~rowlando@213.253.145.9) |
| 11:54.26 | hyphenex | strace it? |
| 11:55.08 | hyphenex | tzafrir: sorry, I'm still lost |
| 12:06.38 | bidossessi | hi all. i'm having a dahdi call-out issue that resembles https://issues.asterisk.org/view.php?id=15429, except that in my case an incoming call doesn't reset the hook status. running asterisk 1.6.0.25, dahdi-linux 2.2.0.2 on Centos5. how could i troubleshoot that? |
| 12:06.40 | tzafrir | strace -p PID |
| 12:06.45 | tzafrir | hyphenex, ==^ |
| 12:07.39 | hyphenex | ohh cool. Thanks :), i'll give it a go |
| 12:08.37 | tzafrir | bidossessi, technically issue 15429 is a duplicate of https://issues.asterisk.org/view.php?id=14577 . |
| 12:08.45 | bidossessi | i'm hoping it's fixed in dahdi 2.2.1-1. |
| 12:08.54 | tzafrir | however, if an incoming call does not reset it, it's probably a different matter |
| 12:09.07 | tzafrir | bidossessi, what device do you use? |
| 12:10.09 | bidossessi | tzafrir, digium TDM410 with 2 FXO modules |
| 12:12.14 | adnc | the sqlite cdr database which asterisk can handle, is this sqlite or sqlite3? |
| 12:12.23 | hyphenex | tzafrir: I get tons of this: |
| 12:12.24 | hyphenex | octl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xbfffc308) = -1 ENOTTY (Inappropriate ioctl for device) |
| 12:12.24 | hyphenex | write(1, "\0"..., 1)           = 1 |
| 12:12.24 | hyphenex | write(1, "*CLI> "..., 6)         = 6 |
| 12:13.12 | tzafrir | hyphenex, somebody tries to write to the console, but the console is /dev/null ? |
| 12:13.42 | tzafrir | how was the main asterisk process run? What's its full command-line? |
| 12:14.12 | hyphenex | tzafrir: I don't know.. in the /etc/init.d |
| 12:14.35 | tzafrir | ps auxww | grep asterisk |
| 12:16.22 | *** join/#asterisk bidossessi (~bidossess@41.221.23.177) |
| 12:19.15 | hyphenex | root 23501 98.8 4.1 54576 21236 ? Rl 22:35 43:34 /usr/sbin/asterisk -c |
| 12:19.16 | hyphenex | root 23672 0.0 0.1 3116 712 pts/0 S+ 23:19 0:00 grep asterisk |
| 12:19.56 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 12:20.59 | tzafrir | hyphenex, from what terminal was it run? |
| 12:21.16 | tzafrir | Does this terminal exist? |
| 12:21.39 | tzafrir | ls -l /proc/23501/fd/1 |
| 12:22.01 | *** join/#asterisk bidossessi (~bidossess@41.221.23.177) |
| 12:22.28 | hyphenex | tzafrir: lrwx------ 1 root root 64 2010-03-10 23:23 /proc/23501/fd/1 -> /dev/null |
| 12:23.02 | tzafrir | hyphenex, so basically you should avoid using 'asterisk -c' if you don't run asterisk in a terminal |
| 12:23.20 | tzafrir | Even though this is a bug that this causes 100% CPU |
| 12:28.03 | *** join/#asterisk Akiraaa (~Akira@92.81.221.18) |
| 12:33.16 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 12:35.15 | *** join/#asterisk Caplain (~shayne@pool-108-9-0-27.tampfl.fios.verizon.net) |
| 12:36.58 | *** join/#asterisk garymc (~chatzilla@host81-148-109-86.in-addr.btopenworld.com) |
| 12:44.25 | *** join/#asterisk fofware (~chatzilla@190.225.133.208) |
| 12:47.12 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 12:52.11 | *** join/#asterisk mallchin (~mallchin@mail.dataproservices.co.uk) |
| 12:52.20 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 12:52.27 | mallchin | hi guys, is there any way to enable outbound calls when doing a graceful stop? |
| 12:52.44 | mallchin | but reject new inbound calls |
| 12:52.52 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 12:53.12 | mallchin | alternatively, a method to reject inbound calls with busy, I guess this is what graceful does? |
| 12:59.24 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:00.54 | *** join/#asterisk nightrid3r (kvirc@41.214.199.29) |
| 13:02.27 | *** join/#asterisk bidossessi (~bidossess@41.221.23.177) |
| 13:03.06 | bidossessi | only one buggy ISP for this whole country. no wonder... |
| 13:03.30 | bidossessi | tzafrir, i don't know if you remember my intervention earlier |
| 13:04.45 | tzafrir | bidossessi, you simply can't call out? |
| 13:05.01 | bidossessi | yes |
| 13:06.14 | bidossessi | you were asking what device i use: TDM 410P |
| 13:06.52 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:08.07 | *** join/#asterisk hipitihop (~denis@203.132.229.18) |
| 13:08.38 | hipitihop | nods to the residents |
| 13:09.34 | hipitihop | any recommendations for a reliable softphone app for the iphone |
| 13:14.50 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:15.08 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:15.16 | ariel_ | Morning |
| 13:15.21 | fish-bulb | hipitihop: I've used fring before. It seems pretty stable, but not the greatest softphone ever |
| 13:16.33 | hipitihop | fish-bulb, thanks, I guess it would need to always be running for it to be registered with asterisk |
| 13:16.36 | *** join/#asterisk aandrade (~aandrade@189.114.184.18) |
| 13:17.03 | *** join/#asterisk DaBigMac (~jnaumovs@177.106.233.220.static.exetel.com.au) |
| 13:17.13 | DaBigMac | hi all |
| 13:17.38 | DaBigMac | im wondering if someone can help with a problem Im having with asterisk |
| 13:18.15 | hipitihop | DaBigMac, hi.. you're btter off to just ask and if someone knows they will respond. Pretty standard irc |
| 13:18.42 | DaBigMac | I have Debian Linus (Lenny), built Asterisk 1.6, dahdi, and have a std fxo single port in it |
| 13:18.43 | tzafrir | infobot, tell DaBigMac about ask |
| 13:18.58 | DaBigMac | I built asterisk gui |
| 13:19.37 | *** part/#asterisk hyphenex (~Adium@115-64-56-198.static.tpgi.com.au) |
| 13:19.50 | DaBigMac | so far so good, dahdi could see the card, gui was working fine. I started using the gui and made some changes to the hardware section applied them then rebooted |
| 13:19.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:20.13 | DaBigMac | now the server wont boot up past populating /var |
| 13:20.25 | DaBigMac | what files does the hardware section of the gui touch? |
| 13:22.07 | elred_ | well if it's "hardware section" it's probably one of zapata.conf/dahdi.conf |
| 13:22.26 | *** join/#asterisk Terminus- (~justin@112.202.183.221) |
| 13:22.29 | elred_ | or others file dealing with your card |
| 13:22.44 | elred_ | seek info on voip-info.org, it's of some help |
| 13:22.44 | DaBigMac | i noticed it touching modprobe.conf |
| 13:23.06 | elred_ | then it probably just don't even load the right drivers for your card |
| 13:23.17 | elred_ | making asterisk hang |
| 13:23.18 | DaBigMac | where does the gui backup the files befor touching them? |
| 13:23.33 | elred_ | did you look at log file ? Or tried to increase verbosity ? |
| 13:23.45 | elred_ | I have no idee, sorry. Never used a gui |
| 13:24.09 | DaBigMac | its locking the machine up early in the boot cycle |
| 13:24.19 | elred_ | hmmm |
| 13:24.27 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 13:24.30 | *** join/#asterisk _gm (~quassel@203.215.176.22) |
| 13:24.32 | Terminus- | hello. question, how can i find out whether asterisk was built with imap support or not? |
| 13:24.35 | elred_ | if the *system* is blocking it's not asterisk related problem |
| 13:24.41 | _gm | hi |
| 13:25.02 | DaBigMac | agreed but it only started after using hardware change in the gui' |
| 13:25.08 | _gm | i am running asterisk (with realtime mysql) and getting following error when calling another box |
| 13:25.29 | elred_ | Terminus-, "strings $(which asterisk) | grep -i imap" ? ;) |
| 13:25.35 | _gm | <PROTECTED> |
| 13:25.37 | elred_ | Terminus-, or ldd even |
| 13:26.02 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 13:26.03 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 13:26.03 | ManxPower-work | _gm, What is the actual Dial line that shows up on the console? |
| 13:26.12 | Terminus- | elred_: oooh... never though of that. thought there was a more official way. =D |
| 13:26.36 | ManxPower-work | Terminus-, there is no official way, since you should know if you built Asterisk with IMAP support or not. |
| 13:26.40 | _gm | Executing [s@CONFERENCE:6] Dial("SIP/3218435112-00001ab5", "SIP/cyrenity@77.66.16.36") in new stack |
| 13:27.18 | kaldemar | DaBigMac: what gui? |
| 13:27.19 | Terminus- | ManxPower-work: since i didn't build it, i wouldn't know. then again, maybe you'd know if the version in the yum repo has imap support? |
| 13:27.25 | _gm | 77.66.16.36 is a peer defined in sip.conf with qualify=yes but it's always showing UNREACHABLE |
| 13:27.42 | ManxPower-work | _gm, You really should not dial by IP address. I suspect your SIP device 77.66.16.36 is not available. |
| 13:27.53 | elred_ | yup |
| 13:27.57 | ManxPower-work | _gm, NO NOT define peers in sip.conf by IP or hostname. |
| 13:28.23 | ManxPower-work | _gm, do you have NAT or firewall involved. |
| 13:28.35 | ManxPower-work | Terminus-, Packages are not supported here. |
| 13:28.39 | _gm | it's i ran sipp on same machine and it was able to connect the other box which is UNREACHABLE for for asterisk |
| 13:28.51 | ManxPower-work | _gm, answer the question |
| 13:29.10 | _gm | no firewall or nat |
| 13:29.26 | _gm | ManxPower-work: it was working since months |
| 13:29.31 | Terminus- | ManxPower-work: ok. i thought since the repo i'm using is asterisk.org, it would be a valid question. |
| 13:29.38 | ManxPower-work | _gm, Ah, I can't help you since everything is working. |
| 13:29.40 | _gm | we migrated our database and after restarting asterisk it stopped working |
| 13:29.50 | ManxPower-work | Terminus-, I'm sure it's a valid question for the packager. |
| 13:30.14 | Terminus- | ManxPower-work: ok. |
| 13:30.27 | ManxPower-work | This is not Official Digium Support. If you want to contact them, then contact Digium. I bet Digium doesn't support those packages wither. |
| 13:30.44 | *** join/#asterisk steak__ (~alex@fire.perspectix.com) |
| 13:30.45 | steak__ | hello |
| 13:31.04 | _gm | other than this single peer i m able to connect with every peer defined in sip.conf |
| 13:31.24 | steak__ | is anybody here having some experience with the Snom M3 DECT/SIP phones |
| 13:31.26 | steak__ | ? |
| 13:31.55 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:32.27 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 13:32.29 | BCS-Satori | Does asterisk support "sip/simple" hints? |
| 13:33.11 | ManxPower-work | BCS-Satori, I don't think so. I believe Asterisk's hint model is SIP SUBSCRIBE/NOTIFY, rather than SIP SIMPLE (since SIMPLE is a text messaging thingy) |
| 13:34.18 | fish-bulb | hipitihop: yeah, I'm sure there are apps you can use on a jailbroken iPhone that can be backgrounded, but not on legitimate firmware |
| 13:34.26 | [TK]D-Fender | BCS-Satori: No |
| 13:34.54 | BCS-Satori | ManxPower-work & [TK]D-Fender: Thanks |
| 13:35.22 | [TK]D-Fender | _gm: Start by not naming the peer an IP |
| 13:35.35 | Terminus- | ManxPower-work: i wouldn't expect this channel to be digium support, but since it's the channel mentioned on asterisk.org, i can ask here. |
| 13:35.47 | [TK]D-Fender | _gm: then show us its entry, and your attempt with SIP DEBUG enabeld. |
| 13:35.53 | _gm | [TK]D-Fender: okay |
| 13:35.57 | ManxPower-work | [TK]D-Fender, I already told him that. "It was working before 'we upgraded the database'" whatever the hell that means. |
| 13:36.11 | [TK]D-Fender | ManxPower-work: that would be "nothing" |
| 13:36.16 | ManxPower-work | ~packages |
| 13:36.45 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 13:36.58 | hipitihop | fish-bulb, hmm ok, currently that's out of the question .. my gf is resisting getting jailbroken .. I need a seemless setup where if her iphone is at home and on the home net, it is just another peer |
| 13:37.02 | LemensTS | Anyone know of a sip provider that i can do fax over that i can enter into an ATA? |
| 13:37.46 | *** join/#asterisk icek (~pisg@intrush.pl) |
| 13:38.50 | icek | hi, im setup asterisk + mysql and only have in extensions.conf, exten => _X.,1,Dial(DAHDI/g0/${EXTEN}) |
| 13:39.16 | icek | i can call, but i have this : http://pastebin.com/SLVYVC5c |
| 13:39.55 | icek | in chan_dahdi.conf is set up, pridialplan=dynamic |
| 13:40.07 | ManxPower-work | icek, Why do you have dynamic? |
| 13:40.17 | Akiraaa | Has anyone used the Linksys SPA400? According to the seller spec, it only works with the SPA9000 hardware pbx and not a general purpose sip ippbx (like asterisk) |
| 13:40.26 | ManxPower-work | icek, What is the ACTUAL dial line as show in the console that causes that message? |
| 13:40.34 | [TK]D-Fender | ickmund: use "unknown" |
| 13:41.03 | [TK]D-Fender | Akiraaa: it doesn, but its got limitations like not being able to target individual lines, etc |
| 13:41.15 | [TK]D-Fender | Akiraaa: Wouldn't advise... |
| 13:42.24 | Akiraaa | [TK]D-Fender: is there a 4FXO standalone device you would recommend? |
| 13:42.24 | icek | ManxPower-work so i change to unknown |
| 13:42.24 | icek | and this same |
| 13:42.24 | ManxPower-work | icek, answer my question |
| 13:42.37 | [TK]D-Fender | Akiraaa: Mediatrix & Audiocode each have many |
| 13:42.42 | ManxPower-work | remember you should stop and start Asterisk when making PRI changes. |
| 13:42.57 | ManxPower-work | <ManxPower-work> icek, What is the ACTUAL dial line as show in the console that causes that message? |
| 13:44.37 | ManxPower-work | icek, DO NOT MSG ME. |
| 13:44.54 | icek | this is outline |
| 13:45.09 | ManxPower-work | I do not understand. What is an "outline" |
| 13:45.28 | ManxPower-work | I simply asked you to paste that one Dial line in the CLI. |
| 13:45.44 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:45.55 | ManxPower-work | I think I know what the problem is but your lack of providing the information is limiting what I can do. |
| 13:46.54 | icek | ManxPower-work: how to i can check this Dial line ? command ? |
| 13:47.33 | ManxPower-work | icek, connect to the exstiting Asterisk process using "asterisk -rvvv" Make a call. copy the line on the screen that says "Dial" and paste just that one line to the channel. |
| 13:47.53 | ManxPower-work | icek, you should read the Asterisk book. You are asking basic questions everyone should now before coming here. |
| 13:47.54 | ManxPower-work | ~book |
| 13:47.55 | infobot | [~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 at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:48.00 | hipitihop | What is the correct way to setup a divert ... i.e. if I know I will be out of the house and I want calls diverted to my mobile |
| 13:48.14 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:48.28 | ManxPower-work | hipitihop, That depends on the switch. Chances are Asterisk does not support Divert with your switch. |
| 13:48.41 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 13:48.53 | hipitihop | ManxPower-work, pure voip here |
| 13:49.00 | ManxPower-work | Divert is a BRI term, I doubt I'd be much help diagnosing BRI problems. |
| 13:49.21 | ManxPower-work | hipitihop, what SPECIFICALY do you want to do? |
| 13:50.13 | *** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net) |
| 13:50.29 | icek | ManxPower-work: http://pastebin.com/gPCZR92G |
| 13:50.44 | [TK]D-Fender | hipitihop: its your dialplan, do whatever you want |
| 13:51.15 | ManxPower-work | icek, I cannot help you further. |
| 13:52.14 | icek | ManxPower-work: ok, ;/ |
| 13:52.36 | hipitihop | ManxPower-work, [TK]D-Fender ... I would like to be able to simply divert allincoming calls to my mobile... so I guess my incomming dialplan just uses Dial(SIP/040xxxx@voipprovider) |
| 13:53.03 | [TK]D-Fender | hipitihop: You "guess"? This is your dialplan... |
| 13:53.05 | ManxPower-work | hipitihop, correct. Stop using the term "divert" You are not "diverting" you are dialing just like dialing any other destination |
| 13:53.43 | [TK]D-Fender | ManxPower-work: Right... maybe he should have said "route" instead :) |
| 13:53.53 | hipitihop | [TK]D-Fender, yes yes, my dialplan :-) |
| 13:53.56 | ManxPower-work | [TK]D-Fender, "dial" would also be good term. |
| 13:54.11 | [TK]D-Fender | ManxPower-work: </sarcasm> |
| 13:54.24 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:54.31 | [TK]D-Fender | ManxPower-work: c'mon... I didn't jsut telegraph that one, that was snail-mail |
| 13:54.35 | icek | ManxPower-work: http://pastebin.com/7bGk8r3b look only this, pls |
| 13:54.46 | hipitihop | ManxPower-work, ok, sorry in this country mobile providers talk about diverts e.g. when mobile not available, divert to another number |
| 13:55.19 | ManxPower-work | icek, now you are providing the information I asked for. I may be able to help you again. Pastebin your /etc/asterisk/chan_dahdi.conf |
| 13:55.37 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:55.59 | [TK]D-Fender | hipitihop: well you're talking about *. * calls whatever you tell it to call. There is no "divert" concept, because there is no framework for changing where a call goes consistently |
| 13:56.13 | ManxPower-work | hipitihop, they are using "divert" (which I think is called 2BCT), instead of "forward" |
| 13:56.14 | icek | ManxPower-work: http://pastebin.com/By0RhZ2B |
| 13:57.13 | hipitihop | ManxPower & [TK]D-Fender ok, thanks for the clarifications and your patience with my imprecise terminology, I will get better, promise :-) |
| 13:57.54 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 13:58.03 | [TK]D-Fender | icek: pastebin you DAHDI configs |
| 13:58.14 | ManxPower-work | icek, Try this one: http://pastebin.com/p6ybBLiu |
| 13:58.25 | ManxPower-work | remember to STOP asterisk then start asterisk after updating the file. |
| 13:58.29 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 13:58.36 | icek | ok i try |
| 13:59.11 | hipitihop | I thought asmuch just doing Dial but wondered if there was an alternative ... so using the Dial approach, how will that incoming call look like to the provider ? as answered ? or only if the Dial is actualy completed by me answering on my mobile ? |
| 13:59.26 | bidossessi | hi all. i'm having a dahdi call-out issue that resembles https://issues.asterisk.org/view.php?id=15429, except that in my case an incoming call doesn't reset the hook status. running asterisk 1.6.0.25, dahdi-linux 2.2.0.2 on Centos5. how could i troubleshoot that? |
| 13:59.38 | bidossessi | using a TDM410P |
| 14:00.21 | ManxPower-work | hipitihop, if you don't answer the call and don't run any applications that answer the call, then the call should not be answered |
| 14:01.31 | *** part/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 14:01.36 | ManxPower-work | Happy Alexander Graham Bell Day! |
| 14:01.45 | [TK]D-Fender | hipitihop: Incoming call look like to the provider? They called YOU. Whats for them to "look at"? |
| 14:01.55 | icek | ManxPower-work: this same problem, http://pastebin.com/XGHExgTC |
| 14:02.07 | ManxPower-work | icek, what is the output of "core show uptime" |
| 14:02.11 | [TK]D-Fender | icek: pastebin you DAHDI configs |
| 14:02.27 | icek | core show uptime |
| 14:02.27 | icek | System uptime: 1 minute, 1 second |
| 14:02.27 | icek | Last reload: 1 minute, 1 second |
| 14:02.57 | icek | [TK]D-Fender: http://pastebin.com/p6ybBLiu |
| 14:03.03 | ManxPower-work | icek, I have no more ideas. If you are telling the truth, then I have no more ideas. |
| 14:03.11 | hipitihop | ManxPower-work, hmm, makes sense, no different to me dialing my local ATA, if it does not answer, the incoming call is never completed. |
| 14:03.54 | [TK]D-Fender | icek: prilocaldialplan=unknown |
| 14:04.01 | [TK]D-Fender | icek: add this below the other one |
| 14:04.30 | hipitihop | ManxPower-work, so is it posisble to setup a dialplan in such a way that I can enable and disable such a dial without changing the extensions.conf each time ? |
| 14:04.44 | ManxPower-work | hipitihop, yes. |
| 14:05.11 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-1-83.acanac.net) |
| 14:05.15 | [TK]D-Fender | hipitihop: Yes, make an exten that will toggle a consistent value like an AstDB entry. Check for that entry in your inbound extens. |
| 14:05.34 | [TK]D-Fender | hipitihop: "core show application gotoif" , "core show function DB" |
| 14:06.05 | icek | [TK]D-Fender: this same |
| 14:06.16 | icek | ManxPower-work: thx for help |
| 14:06.17 | *** join/#asterisk kartik (~koolkarti@117.199.112.69) |
| 14:06.22 | tzafrir | bidossessi, what exact error message do you get when you try to dial out? |
| 14:06.48 | *** join/#asterisk titter` (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 14:06.53 | bidossessi | tzafrir, Everyone is busy/congested at this time (1:0/0/1) |
| 14:07.03 | *** join/#asterisk neurosys (~neurosys@173.200.203.86) |
| 14:07.18 | neurosys | any polycom gurus in the house? :P |
| 14:07.32 | ManxPower-work | neurosys, I'm sure there are. |
| 14:07.50 | [TK]D-Fender | ~8ball any polycom gurus in the house? |
| 14:07.51 | infobot | Absolutely. |
| 14:08.00 | neurosys | heh |
| 14:08.00 | [TK]D-Fender | infobot: has SPOKEN |
| 14:08.16 | neurosys | TK, I didnt see you on the userlist ;) |
| 14:08.35 | hipitihop | [TK]D-Fender, sounds a little advanced but I guess in the deep end I go .. will tackle with fresh brain tomorrow... |
| 14:08.55 | icek | [TK]D-Fender: any ideas ? |
| 14:09.15 | neurosys | [TK]D-Fender: I have a dozen 501's that try to dial out at DT after the 1st 3 digits. I checked the digitmapping, I cant see any issues. Can i msg you the digitmap and tell me if im missing something? |
| 14:09.31 | icek | ManxPower-work: i can calling, but always see 3 warning ;/ |
| 14:09.32 | [TK]D-Fender | icek: idea = show me you updated configs... don't just say "didn't work" |
| 14:09.32 | ManxPower-work | neurosys, post your digitmap to the channel, it's only one line,. |
| 14:09.36 | neurosys | k |
| 14:09.42 | neurosys | [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT|[1-8]xx |
| 14:09.46 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:09.57 | *** join/#asterisk smooth_penguin (~smoove@59.95.35.59) |
| 14:09.57 | hipitihop | [TK]D-Fender, .. is there any danger since in this proposed setup asterisk will relay (sorry if wrong term) the call to my mobile, that if the mobile drops out, the inbound call stays connected, clocking up call charges ? |
| 14:09.59 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 14:10.11 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 14:10.11 | ManxPower-work | neurosys, which first three digits? |
| 14:10.19 | plundra | Can I "kick out" a logged in manager? |
| 14:10.26 | plundra | (To make it reconnect) |
| 14:10.47 | icek | [TK]D-Fender: http://pastebin.com/xb2FKJek |
| 14:10.48 | [TK]D-Fender | hipitihop: If the mobil drops, wht wouldn't the call to it drop? |
| 14:10.54 | neurosys | Well.. if i try to dial from DT 305-324-8811, It pushs the 305 instantly and gives me an error (since there are no 3 digit exts aside from the 1xx) |
| 14:11.20 | tzafrir | bidossessi, 0 busy, 0 congested, 1 not available |
| 14:11.41 | ManxPower-work | neurosys, since you are not using "9" or similar for outside line and since you are not requiring a 1 for outside calls, you will have to deal with timeouts. |
| 14:11.56 | tzafrir | next thing I would try is to add some debug messages in the function available() in channels/chan_dahdi.c |
| 14:12.01 | [TK]D-Fender | icek: Ok, no idea... |
| 14:12.12 | hipitihop | [TK]D-Fender, in theory yes, in which case * will see it as that phone hanging up and it should hangup too .. I probably need a Hangup() after the Dial() |
| 14:12.28 | [TK]D-Fender | hipitihop: that would be a good idea.... |
| 14:12.49 | neurosys | manxpower: Ok, what if i simple change the [1-8]xx to 1xx. then any 10 digit not starting with 1 should pass correctly? |
| 14:12.54 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:12.56 | hipitihop | [TK]D-Fender, I simply wondered if there were other special precautions one should take in the diaplan for such situations. |
| 14:12.59 | tzafrir | bidossessi, if that function returns 0, the channel is not available. It has various possible places where it can return that value. I wonder which of them applies |
| 14:13.16 | ManxPower-work | neurosys, I would use 1 to dial outside. What ranges are your extensions. |
| 14:13.21 | tzafrir | bidossessi, another sanity check: dahdi show channel NN |
| 14:13.34 | tzafrir | do you have '0' in InAlarm: ? |
| 14:13.39 | [TK]D-Fender | hipitihop: if the call actually drops, you don't need "hangup" |
| 14:13.43 | neurosys | ManxPower-work: 100 thru 110 |
| 14:13.48 | [TK]D-Fender | hipitihop: thats more for if it doesn't answer |
| 14:13.52 | ManxPower-work | neurosys, well that was pretty stupid. |
| 14:14.25 | neurosys | ManxPower-work: lol I came into the situation with the handsets already programmed to 3 digit exten |
| 14:14.37 | ManxPower-work | neurosys, sucks to be them. |
| 14:14.52 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 14:15.15 | neurosys | ManxPower-work: so taking out the [1-8] range and making it 1 only wont work? |
| 14:15.45 | ManxPower-work | try this: [2-9]xx[2-9]xxxxxx|011xxx.T|1xx|911 |
| 14:15.54 | bidossessi | tzafrir, http://pastebin.com/er7ZC6f6 |
| 14:16.05 | hipitihop | ManxPower-work, [TK]D-Fender once again, thanks for letting me pick your brains |
| 14:16.08 | ManxPower-work | then dial all outside calls as area code + phone number. add the 1 if needed by your carrier in your dialplan |
| 14:16.20 | bidossessi | tzafrir, inAlarm: 0 |
| 14:16.28 | tzafrir | bidossessi, InAlarm: 0 , so not that problem |
| 14:16.30 | hipitihop | gnite all |
| 14:16.30 | *** join/#asterisk dymaxion (~dymaxion_@host217-40-240-249.in-addr.btopenworld.com) |
| 14:16.58 | neurosys | ManxPower-work: ok thanks! I dont need the range and X for 10-digit? |
| 14:16.58 | ManxPower-work | hipitihop, you'll need 911T at the end instead because you still have overlapping dial patterns |
| 14:17.12 | ManxPower-work | neurosys, huh? |
| 14:17.26 | neurosys | ManxPower-work: [2-9]xxxxxxxxx? |
| 14:17.39 | *** join/#asterisk mono000333 (~mono00033@host-50.GROUPB.212.5.107.48.0xfffffff0.macomnet.net) |
| 14:17.40 | ManxPower-work | neurosys, that's a crappy pattern for USA PSTN calls. |
| 14:17.47 | ManxPower-work | which is why I replaced it |
| 14:18.37 | ManxPower-work | <PROTECTED> |
| 14:18.46 | neurosys | ManxPower-work: Ahhhh! |
| 14:19.01 | neurosys | ManxPower-work: Gotcha. I didnt notice the missing | |
| 14:19.25 | ManxPower-work | neurosys, some day you'll not notice something that causes a major security issue. |
| 14:19.26 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:19.27 | *** join/#asterisk riksta (~rick@92.63.131.41) |
| 14:19.55 | neurosys | ManxPower-work: I'm quite sure I will. Watch out for nuclear missles ;) |
| 14:20.15 | ManxPower-work | neurosys, not at all, just watch for $10,000 phone bills. |
| 14:20.37 | neurosys | ManxPower-work: Would you like the IP? :) |
| 14:20.59 | ManxPower-work | neurosys, do you have allowguest=no in sip.conf? |
| 14:21.13 | mallchin | hi guys, is there any way to enable outbound calls when doing a graceful stop? |
| 14:21.39 | [TK]D-Fender | mallchin: "stop when convenient" <- |
| 14:21.39 | ManxPower-work | mallchin, Only if you code it in the dialplan. |
| 14:21.59 | neurosys | ManxPower-work: Of course. I dont allow rouge sip connections |
| 14:22.12 | mallchin | [TK]D-Fender: thanks, I notice this can take many hours until server stops, I'll look into it |
| 14:22.32 | ManxPower-work | mallchin, what you want to do is a complex set of dialplan programming. |
| 14:23.04 | mallchin | ManxPower-work: to reject incoming calls? I tried PRI code 44 but it returns busy to the caller |
| 14:23.30 | ManxPower-work | mallchin, what is wrong with that? and why did you pick 44? |
| 14:23.32 | neurosys | ManxPower-work: PS: Thanks for your help |
| 14:23.39 | neurosys | :) |
| 14:23.50 | mallchin | ManxPower-work: 44 seemed the best option, another developer chose it |
| 14:24.16 | ManxPower-work | what do you WANT to happen when you reject the call because of a pending restart? |
| 14:24.36 | mallchin | ManxPower-work: I have 5 circuits as part of a supergroup, rather than returning busy I'd like the calls to flow to another circuit in the supergroup, which they don't if it returns busy |
| 14:24.52 | ManxPower-work | mallchin, 44 is not "BUSY" |
| 14:24.57 | mallchin | ManxPower-work: have it route to another circuit |
| 14:25.16 | mallchin | ManxPower-work: what is 44? |
| 14:25.27 | mallchin | (I know this is a RTFM question) |
| 14:25.42 | ManxPower-work | mallchin, Requested Circuit/Channel not Available. |
| 14:26.02 | ManxPower-work | mallchin, does calls roll over now when one of your PRIs are busy? |
| 14:26.44 | mallchin | ManxPower-work: we've never reached capacity so don't get busy, but they do round robin between all circuits |
| 14:26.50 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:26.50 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:26.54 | mallchin | ManxPower-work: should a circuit go down, calls do not route to it |
| 14:27.26 | mallchin | ManxPower-work: I'ld like to mimick this behaviour for new incoming calls on a graceful restart |
| 14:28.15 | ManxPower-work | mallchin, you can't do that unless your telco is set up to redirect calls to your other PRIs |
| 14:28.42 | *** join/#asterisk Polysics (~Luca@host83-67-dynamic.30-79-r.retail.telecomitalia.it) |
| 14:28.44 | Polysics | hello |
| 14:28.49 | riksta | Hi guys, using asterisk 1.6.1 I have been happily using Set(CDR(mydbfield)=myvalue) for a long time. I just create a varchar field in the cdr table in mysql. I have replicated this for the latest 1.6.2.5 and the value is not set... I cannot find anywhere in the documentation which states that this syntax has change. Has it? |
| 14:29.16 | Polysics | this is tricky: i have clients calling each other, but when one rejects a call, it gets reported as busy which is imho incorrect |
| 14:29.19 | ManxPower-work | mallchin, you could try Hangup(3), but what your carrier does for each code is up to the carrier. |
| 14:29.33 | mallchin | ManxPower-work: the telco is setup to do so, at least they should be |
| 14:29.36 | Polysics | i am using plain Dial extensions with no Answer before or Hangup after |
| 14:29.40 | ManxPower-work | Polysics, most phones send back busy |
| 14:30.05 | Polysics | ManxPower-work, so there is no way to have something else sent? |
| 14:30.13 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 14:30.17 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:30.32 | mallchin | okay thanks for the help guys, I'll do some more research |
| 14:30.37 | mallchin | :) |
| 14:30.55 | ManxPower-work | riksta, all significant changes beween major Asterisk versions (and upgrading 1.6.1 ro 1.6.2 is a major revision change) should be documented in UPGRADE*.txt |
| 14:31.06 | ManxPower-work | mallchin, you can send back any code you want. |
| 14:31.41 | [TK]D-Fender | Polysics: `thats up to the phone |
| 14:31.47 | ManxPower-work | Polysics, I guess you could replace your phone with some other phone. But the PHONE is sending back the busy, not ASTERISK. |
| 14:32.01 | Polysics | ok, understood |
| 14:32.02 | Polysics | thanks |
| 14:32.15 | mallchin | ManxPower-work: it's possible sending the correct code will do the trick, but as you note, it depends how the telco handles the code, and we use several telcos |
| 14:32.24 | ManxPower-work | Polysics, we usually disable phone based call forward, DND, and directories and use server based ones. |
| 14:32.27 | Polysics | i noticed that when i had extensions Answer before Dial, this did not happen |
| 14:32.47 | mallchin | ManxPower-work: I'll have to confirm the configuration there end, and what code to send them to have them re-direct the call to another circuit |
| 14:32.50 | ManxPower-work | Polysics, what does happen then? |
| 14:33.01 | Polysics | ManxPower-work, i have no choice of client as we are using Zoiper Web in a browser based thing |
| 14:33.04 | mallchin | *their |
| 14:33.15 | Polysics | the call gets properly rejected, at least as far as the events I see |
| 14:33.20 | Polysics | not reported as busy |
| 14:33.31 | ManxPower-work | Polysics, that is not a helpful answer. |
| 14:33.52 | ManxPower-work | A helpful answer would be something like DIALSTATUS is ?? or HANGUPCAUSE is ?? or sip debug shows SIP response ?? |
| 14:34.00 | Polysics | it was also horribily phrased :-) |
| 14:34.16 | riksta | ManxPower-work, yeah I cannot see anything relevant in there and want to know if anyone is using Set(CDR.... with a custom field in 1.6.2 before i filed a bug |
| 14:34.45 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.regina.accesscomm.ca) |
| 14:35.10 | Polysics | ManxPower-work, this will make me look bad, but is there any way to debug a Sip call, having a single call logged somewhere? |
| 14:35.19 | Polysics | or do i have to wade through SIP output every time? |
| 14:35.24 | ManxPower-work | no, but you can "sip debug peer XXXX" |
| 14:36.08 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 14:37.10 | Skeeter- | ariel_, with you spectralink, if your customer walk a long way into the boat or buidling and he switchs AP, you lost the signal, is there anyway to bypass that? |
| 14:38.20 | ariel_ | Skeeter-: we use there SVP servers and we have all the AP's configured with the same SSID, we do not have any drops going from ap to ap |
| 14:39.04 | ariel_ | In fact they never know when they move from ap to ap. |
| 14:39.17 | Skeeter- | my AP got the same SSID and encryption, im not sure what the SVP is gonna change |
| 14:39.21 | Polysics | ok, from what I am seeing, nothing is wrong with asterisk |
| 14:40.01 | Polysics | if i use Answer before Dial, it... answers before dialing, thus not sending a "connection is up" signal at the proper time, ie. when the other person picks up the phone |
| 14:40.01 | *** part/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 14:40.12 | ariel_ | Skeeter-: in our case the SVP is the one that keeps the call route so if one ap drops it QoS seamlessly moves to the next available AP |
| 14:40.18 | Polysics | so call rejection works properly as the call is not rejected, but hung up |
| 14:40.41 | Polysics | i would say the proper behaviour is without Answer, ie. if a call is rejected, it is rejected and that's all |
| 14:41.10 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 14:41.11 | Polysics | is Hangup after Dial needed/recommended? |
| 14:41.57 | ariel_ | if a call is rejected there has to be something in the dial plan to account for it. |
| 14:42.02 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:42.11 | *** join/#asterisk sourcode (~code@ppp-58-8-236-217.revip2.asianet.co.th) |
| 14:44.00 | Polysics | ariel_, so Hangup does that? |
| 14:44.18 | ariel_ | what is your next priority in the dial plan? |
| 14:44.37 | Skeeter- | ariel_, so the svp has nothing to do with disconnecting, must be my AP that are not set propely? i talk over with the guy that installed them and he told me that it was normal to lose a ping while switching APs |
| 14:44.44 | ariel_ | you can send it ot vm, hangup, dial another extension what ever you want it to do |
| 14:45.51 | *** join/#asterisk moy (~chatzilla@74.12.129.100) |
| 14:46.00 | ariel_ | Skeeter-: phones are not pinging, there only sending rtp over the net in this case they just switch. No drop nothing they can or should hear. |
| 14:46.31 | ariel_ | svp in our setup gives them a 2nd IP that is used for the rtp |
| 14:47.00 | Skeeter- | ariel_, then switching APs must not be why the phones disconnect, it happens about 1-2 times a day |
| 14:47.14 | ariel_ | diconnect from ? |
| 14:47.39 | Skeeter- | thats ironic |
| 14:48.38 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:48.49 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:49.08 | ariel_ | argh network just took a dump at work. |
| 14:49.26 | Skeeter- | ariel_, they are in the middle of a call and then the call ends on both END and the spectralink states Press End. |
| 14:50.44 | ariel_ | my suggestion is call Polycom support. Only time that happens is if our users go into the freezer or area that there is not coverage |
| 14:51.02 | ariel_ | not/no |
| 14:51.12 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:51.40 | *** join/#asterisk titter (~titter@c-98-208-158-125.hsd1.fl.comcast.net) |
| 14:51.42 | ariel_ | Skeeter-: what ap's are you using? |
| 14:52.04 | Skeeter- | wrt54gl with ddwrt |
| 14:53.55 | ariel_ | do they support WMM |
| 14:54.14 | Skeeter- | ariel_, sure, the phones would even boot/connect |
| 14:54.23 | ariel_ | no not same |
| 14:54.49 | ariel_ | WMM is based on 802.11e Enhanced Distributed Coordination Access (EDCA). Wi-Fi networks that implement WMM optimize the allocation of shared network resources among competing applications by prioritizing media access depending on the traffic type. This approach brings flexibility in networks that have concurrent applications with different latency and bandwidth requirements. |
| 14:55.26 | ariel_ | http://www.polycom.com/global/documents/support/setup_maintenance/products/voice/Best_Practices_Guide_to_Deploying_SpectraLink_8020_8030_Wireless_Telephones.pdf |
| 14:55.45 | Skeeter- | ariel_, ddwrt offers it but the router cant hardwarly do it? |
| 14:56.03 | ariel_ | then you need the SVP |
| 14:57.13 | ariel_ | the SVP Server is responsible for providing packet prioritization and timed release for specific time slot deliveries to all wireless phones. |
| 14:58.32 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 14:59.30 | Skeeter- | ariel_, wow, have you ever check the code of those SVP server? |
| 14:59.43 | ariel_ | yes there based on Linux |
| 14:59.55 | ariel_ | I have one here in my lab |
| 15:00.00 | Skeeter- | ariel_, and it is pretty basic |
| 15:00.04 | ariel_ | and almost every ship has at least 2 of them |
| 15:00.15 | ariel_ | yes actually they are very basic |
| 15:00.28 | Skeeter- | ariel_, and expensive |
| 15:00.36 | ariel_ | but there great and auto failover works |
| 15:00.42 | ariel_ | yes expensive |
| 15:01.01 | Skeeter- | I worked with it abit, and i foudn no use, but it sounds like it provides a bit of stability |
| 15:01.22 | ariel_ | But when you need your phones to stay up and it's used in a commercial use it's well worth the cost |
| 15:01.54 | ariel_ | But WMM if done correctly does the same thing |
| 15:02.00 | Skeeter- | ariel_, i called Speactlink support, they told me over and over to get a SVP Server, i got one, then they told me, check your settings and bla bla bla, they dont have a technical support team, u get always the sales team |
| 15:02.12 | Skeeter- | ariel_, ok |
| 15:02.16 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:02.18 | ariel_ | We have tested Cisco 1200 with there updated firmware and they also work very well |
| 15:02.48 | Skeeter- | ariel_, how much are these? |
| 15:02.55 | Skeeter- | ariel_, and what do they require |
| 15:03.45 | ariel_ | Skeeter-: well since we deploy 80 to 100 phone a month, we are able to get directly to there Speralink support team. |
| 15:04.08 | ariel_ | but I have no idea how much they are |
| 15:04.16 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:04.17 | Skeeter- | ok |
| 15:04.20 | ariel_ | I have not priced anything in over 2 years |
| 15:04.40 | Skeeter- | ariel_, u just play with the stuff they give u |
| 15:07.51 | Skeeter- | ariel_, thanks for the tips BTW |
| 15:08.50 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:11.05 | *** join/#asterisk Jhirley (~Jhirley@mail.mmdlaw.com) |
| 15:13.59 | Skeeter- | ariel_, do you only polycoms wiresless phone or u use sth else? |
| 15:15.56 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:16.56 | ariel_ | we only use the Polycom |
| 15:17.09 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 15:17.22 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:17.29 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 15:17.31 | ariel_ | We tested last year with the Cisco's but there not rugged enough for there cost |
| 15:18.18 | coppice | I think the cisco handsets could be used as hammers. they have the weight |
| 15:18.34 | Jhirley | POLLing question, Trixbox or AsteriskNow ? |
| 15:21.41 | zamba | i want to transport audio at a high quality over the internet with low latency.. which solution/protocol should i look into for this? |
| 15:22.09 | zamba | i'm thinking some voip or rtp based stuff? |
| 15:22.37 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 15:22.40 | zamba | i've been using icecast for streaming, but that introduces latency |
| 15:24.42 | Akiraa | Is anyone making VoIP devices with open source or otherwise programmable firmware? |
| 15:26.44 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:28.34 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:28.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:30.17 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 15:34.27 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:34.27 | ariel_ | Jhirley: I will never ever, no matter what use TrixBox. So my vote would be for AsteriskNow |
| 15:34.59 | fish-bulb | Akiraa: I think Snom's firmware is opensource |
| 15:35.11 | Jhirley | ariel_ why do you feel so strongly about that ? never, ever (even). |
| 15:35.25 | Akiraa | what's the difference between Trixbox and AsteriskNow |
| 15:35.40 | ariel_ | It is no longer a real Open Source product |
| 15:35.45 | Akiraa | as I understood, Trixbox used to be called AsteriskNow, until Digium asked them not to |
| 15:35.51 | ariel_ | there support is really bad |
| 15:35.53 | fish-bulb | Akiraa: AsteriskNOW has a lot less crap in it |
| 15:36.10 | ariel_ | no trixbox came from Asterisk in a Box |
| 15:36.38 | ariel_ | But a commercial company took them over |
| 15:36.50 | Akiraa | or Asterisk@home |
| 15:36.56 | ariel_ | that's it |
| 15:36.58 | [TK]D-Fender | Akiraa: No, Trixbox used to be called Asterisk@Home |
| 15:37.01 | fish-bulb | Akiraa: not quite, Trixbox used to be Asterisk@Home |
| 15:37.03 | Jhirley | Asterisk@home I think. |
| 15:37.56 | ariel_ | AsteriskNOW uses a normal setup of Freepbx, trixbox uses one that is very customized |
| 15:37.58 | [TK]D-Fender | [10:35]<Akiraa>what's the difference between Trixbox and AsteriskNow <_Trixbox uses forked versions of the components, includes a shit--ton of other crap and does evil stuff like phoning home... |
| 15:38.30 | Akiraa | ok, I thought it was merely asterisk with freepbx strapped onto it (and some extra modules) |
| 15:38.46 | Akiraa | using it now for testing, btw |
| 15:39.02 | ariel_ | it's limited in it's basic setup Trixbox CE |
| 15:39.23 | ariel_ | But over all I still stay as far away from it as I can. |
| 15:39.26 | fish-bulb | Akiraa: that is exactly what they want you to believe, but they really mean a ton of extra modules that are not necessary |
| 15:40.31 | [TK]D-Fender | Akiraa: For testing of what? |
| 15:40.35 | Jhirley | who is "they" ? |
| 15:40.48 | Akiraa | [TK]D-Fender: for a 20 terminal system |
| 15:40.57 | Akiraa | and 4 final FXO lines |
| 15:41.06 | Akiraa | currently at 10 terminals and 2 FXO |
| 15:41.53 | [TK]D-Fender | Akiraa: What contitutes "testing"? |
| 15:42.27 | Akiraa | using in an office environment, with a fallback to standard telephony in case things go awry |
| 15:42.39 | [TK]D-Fender | Akiraa: that isn't "tesing", that's "using" |
| 15:42.50 | [TK]D-Fender | Akiraa: you are in production <- |
| 15:42.52 | Akiraa | a temporary setup, though |
| 15:43.13 | Skeeter- | [TK]D-Fender, do you have any scholarship or cerfitication, no offense but u always seems to use the perfect words and the best questions |
| 15:43.31 | [TK]D-Fender | Skeeter-: Common sense is very rare. I have that. |
| 15:43.59 | Skeeter- | [TK]D-Fender, You would make a good Jedi i guess... uhh |
| 15:44.13 | [TK]D-Fender | Skeeter-: This is not the channel you are loking for... |
| 15:44.17 | [TK]D-Fender | waves his hand |
| 15:44.35 | Skeeter- | hahaha |
| 15:44.55 | Jhirley | uhm, play nice you all . in my bestest ingles |
| 15:46.23 | ariel_ | argh, I hate it when people feel that just adding things to a dhcp3.conf and not putting comments as to what they added and checking on what it does. |
| 15:49.18 | manxpower | [TK]D-Fender: How many years have you been involved in working with Asterisk? |
| 15:49.54 | Nugget | ariel_: we've started tracking /etc and /usr/local/etc in git on production machines for that reason. |
| 15:49.56 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 15:50.02 | Nugget | it's a great solution |
| 15:50.47 | ariel_ | yes I just looked at the history file and found the person, now to kill him, many times over.... just for fun |
| 15:50.49 | jameswf | hai ho |
| 15:51.12 | Nugget | you want to borrow my LART? |
| 15:51.18 | ariel_ | lol |
| 15:51.32 | ariel_ | I am going to use a spoon due to it will hurt more |
| 15:51.40 | Nugget | I just got it back from the shop, it's all balanced and hydroclaved. |
| 15:51.58 | jameswf | remembers an episode of OZ involving a spoon |
| 15:52.15 | manxpower | ariel_: a simple informative message to everyone in the company explaining the cause of the network problems might go a long way. Let the user's peers beat them senseless. |
| 15:53.13 | ariel_ | that would be a normal response but this is on a ship and they should not have done anything on it or even tried to help them. |
| 15:53.20 | *** join/#asterisk mechbangirc (~mechbangi@115.167.1.226) |
| 15:53.21 | *** join/#asterisk vettehead (~mvarner@205.127.233.211) |
| 15:53.46 | manxpower | [TK]D-Fender: one of my new polycom idle messages: "When taking a report from a customer don't run in circles, don't scream and shout! Write it down, just like in school! Write it down so you don't look like a fool!" |
| 15:53.47 | ariel_ | goes and changes all passwd's and user access |
| 15:54.46 | jameswf | remember kids your not that smare hire a consultant and go back to coloring elephants purple and watching VinDiesel movies |
| 15:54.46 | mechbangirc | hi can someone tell me which response is responsible for billing OR is it rtp stream which is used for billing? |
| 15:54.57 | jameswf | s/smare/smart/ |
| 15:55.54 | tzafrir | Nugget, I guess etckeeper is for you |
| 15:56.08 | mechbangirc | i meant SIP response |
| 15:56.25 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:56.29 | Nugget | spiffy, thanks, I'll take a look |
| 15:56.42 | manxpower | mechbangirc: none of that has anything to do with billing. You should bill based on your CDRs |
| 15:57.42 | [TK]D-Fender | manxpower: about 6 |
| 15:58.20 | mechbangirc | manxpower: i have contract with someone and have problems of FAS it is actually the other party billing i am concerned about |
| 15:58.49 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:59.18 | mechbangirc | manxpower: my carrier is providing fake FAS i am trying to implement some technique in asterisk to tackle this issue |
| 16:00.02 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 16:00.20 | *** join/#asterisk vettehead (~mvarner@205.127.233.211) |
| 16:00.56 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 16:02.11 | mechbangirc | so i need to know how the other party knows that callee has picked up the phone on my side? |
| 16:05.25 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:07.11 | *** join/#asterisk nightrid3r (kvirc@41.214.205.63) |
| 16:08.30 | *** join/#asterisk thehar_ (thehar@thehar.xmission.com) |
| 16:09.11 | *** join/#asterisk nightrid3r (kvirc@41.214.205.63) |
| 16:10.54 | *** join/#asterisk mvarner_ (~mvarner@205.127.233.211) |
| 16:12.14 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:13.11 | *** join/#asterisk vettehead (~mvarner@205.127.233.211) |
| 16:14.37 | manxpower | mechbangirc: You can't fake Facilities Associated Signaling. |
| 16:15.08 | manxpower | mechbangirc: look at SIP debug to see what is sent when call is answered. |
| 16:15.12 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 16:15.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:15.41 | mechbangirc | manxpower: i want to avoid fake False Answer Supervision. it is my carrier who is doing it |
| 16:16.17 | manxpower | mechbangirc: you will fail. |
| 16:16.36 | mechbangirc | manxpower: 200 OK is sent but I read somewhere that it is actually rtp stream which is used for billing. i am confused |
| 16:16.37 | manxpower | mechbangirc: they are not faking the answer supervision. They are ACTUALLY answering the call. |
| 16:16.58 | manxpower | mechbangirc: RTP is audio, you don't "use it for billing" |
| 16:17.18 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:17.20 | mechbangirc | manxpower: yes they answer it as soon i make it even before trying to connect it to callee |
| 16:17.42 | coppice | why do people love cooking up new acronyms like FAS which conflict with perfectly well established acronyms like FAS? |
| 16:17.42 | manxpower | mechbangirc: you will fail at what you are trying to do. Either change carriers or convince your existing carrier to STOP DEFRAUDING you by answering all calls. |
| 16:18.41 | manxpower | coppice: I think because * users don't know much about TCA (Telecom Common Acronyms) or FDA or RCTP |
| 16:18.51 | manxpower | comes up with a few more random acronyms |
| 16:19.11 | coppice | you mean a FMRA |
| 16:19.21 | Jhirley | Those aren't really telcom, try fxo or 1fb or even centrex |
| 16:19.30 | mechbangirc | manxpower: i almost implemented it. i wait the incoming leg of call and originate a new call to the callee i then waitforsilence and waitfornoise combinations. later when it is assured that the callee picked up i bridge two legs |
| 16:19.46 | *** join/#asterisk Havokmon (~rick@mx1.vfemail.net) |
| 16:19.59 | manxpower | mechbangirc: I wish you the BEST of luck. |
| 16:20.13 | ariel_ | I like bets, PETA = People Eating Tasty Animals....;-) |
| 16:20.24 | *** join/#asterisk ttwhy (~tekkno@p4FECF5AB.dip.t-dialin.net) |
| 16:20.49 | Jhirley | they guy from the price is right just gave them a few million dollars. |
| 16:20.58 | leifmadsen | Bob Barker |
| 16:20.59 | mechbangirc | manxpower: thanks |
| 16:21.00 | manxpower | mechbangirc: your carrier is siill defrauding you. |
| 16:21.13 | coppice | False Acronym Substitution |
| 16:21.26 | mechbangirc | manxpower: i dont care i have some bulk package from them it does not cost me extra |
| 16:21.37 | manxpower | mechbangirc: then why do you care? |
| 16:21.51 | manxpower | Your carrier's answer will cause you all sorts of issues. |
| 16:22.18 | mechbangirc | manxposwer: it is the other party with whom i have voip contract i need to send them right signalling for their billing otherwise contract will be blocked |
| 16:23.56 | mechbangirc | i dont care about my carrier and they dont care about themselves |
| 16:27.31 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 16:29.32 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 16:31.06 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.227.3.dsl.dyn.forthnet.gr) |
| 16:31.36 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 16:33.20 | *** part/#asterisk mechbangirc (~mechbangi@115.167.1.226) |
| 16:44.25 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:45.09 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 16:46.46 | Havokmon | I'd like to provide a button on a web page that will allow the user to connect to a predefined conference number on * - without the user needing a login. Does anyone know if that's possible? Are there 'registration-less' clients? |
| 16:46.47 | Kobaz | ~daylightsavings |
| 16:46.52 | Kobaz | mm |
| 16:46.56 | Kobaz | when does it start this year |
| 16:46.58 | *** join/#asterisk smanek (~smanek@rrcs-24-213-164-136.nyc.biz.rr.com) |
| 16:48.55 | Kobaz | looks like the 14th |
| 16:49.05 | *** join/#asterisk boch (~fran@200.61.191.9) |
| 16:49.13 | [TK]D-Fender | Havokmon: Who said you need to register? |
| 16:49.22 | boch | is it possible to send DTMF during a bridged call ? |
| 16:49.28 | nightrid3r | last saturday of march |
| 16:49.47 | dddh | hm |
| 16:49.51 | dddh | morning |
| 16:50.15 | [TK]D-Fender | boch: yes |
| 16:50.40 | freezey | how can i check to see if meetme is an installed application with my asterisk install? |
| 16:50.55 | [TK]D-Fender | freezey: "CORE SHOW APPLIACTION MEETME" |
| 16:51.01 | [TK]D-Fender | -typos & caps |
| 16:51.32 | freezey | ok sweet |
| 16:51.33 | freezey | its there |
| 16:51.35 | freezey | just not working lol |
| 16:51.38 | boch | [TK]D-Fender, i know D option to Dial() is useful, with it will send DTMF just after bridge, i need to wait a few seconds |
| 16:52.02 | [TK]D-Fender | boch: boch then add some delay |
| 16:52.30 | boch | [TK]D-Fender, how is the question |
| 16:52.42 | [TK]D-Fender | boch: D()wwwwwwwwwwwwwwwwww1234567890) |
| 16:53.07 | freezey | [TK]D-Fender: so this is what i have so far. http://pastebin.ca/1832083 and it errors out with http://pastebin.ca/1832084 |
| 16:53.13 | boch | [TK]D-Fender, what is the delay for a w ? a second ? |
| 16:53.14 | sbrath | Is it possible to create a Hint that will respond as ONHOLD when a specificed Queue has callers waiting? |
| 16:54.03 | [TK]D-Fender | freezey: that dialplan has jack shit to do with the CLI output |
| 16:54.15 | freezey | yeah realized that |
| 16:54.18 | [TK]D-Fender | freezey: any more apples & oranges to share & compare? |
| 16:54.23 | freezey | meh nope |
| 16:54.24 | freezey | lol |
| 16:54.34 | freezey | thats just what happens when i dial the conf number |
| 16:54.59 | [TK]D-Fender | sbrath: yes. make a custom hint and a monitoring app to toggle it as their status changes |
| 16:55.09 | [TK]D-Fender | freezey: you have a dialplan problem |
| 16:55.36 | freezey | k |
| 16:55.57 | freezey | anyway possible to point in a direction where i could possibly try and figure this out? |
| 16:56.34 | freezey | like i see the setup where it calls that office-iguanas in my extensions.conf |
| 16:59.25 | manxpower | boch: are you using an analog fxo port? |
| 17:00.03 | boch | manxpower, depends, could be fxo as a sip or iax peer |
| 17:00.13 | [TK]D-Fender | freezey: you don't seem to have a clue what your own dialplan is doing and put an extension where it needs to be... |
| 17:00.19 | [TK]D-Fender | freezey: want a direction?... |
| 17:00.22 | [TK]D-Fender | ~book |
| 17:00.23 | infobot | [~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 at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:00.24 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 17:00.29 | [TK]D-Fender | freezey: Head that way |
| 17:00.56 | freezey | yeah figured that much |
| 17:00.58 | freezey | thanks |
| 17:01.19 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 17:03.03 | sbrath | [TK]D-Fender: monitoring App, is that something outside asterisk? |
| 17:03.20 | [TK]D-Fender | sbrath: yes |
| 17:04.09 | sbrath | any sugestions on one to look at? VracBazar came up as a option, |
| 17:04.16 | [TK]D-Fender | sbrath: Get coding <- |
| 17:04.52 | sbrath | So basicly a endless loop perl script checking the AMI for queue members and toggle the lights... |
| 17:06.34 | manxpower | boch: analog fxo acts differently from SIP and IAX when it comes to considering the called answered. |
| 17:08.33 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 17:08.39 | *** join/#asterisk Polysics (~Luca@host83-67-dynamic.30-79-r.retail.telecomitalia.it) |
| 17:08.42 | Polysics | hello |
| 17:08.47 | Polysics | ok, one last thing before i call it a day |
| 17:08.59 | Polysics | right now i can call a SIP user even if he is already in a call |
| 17:09.15 | *** join/#asterisk dzup (dzup@unaffiliated/dzup) |
| 17:09.18 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:09.29 | Polysics | is it possibile to have someone calling a user that is already talking to be put on hold somehow |
| 17:09.45 | Polysics | with a message saying "you are waiting to be connected"? |
| 17:10.20 | Polysics | which probably has a better name |
| 17:12.16 | *** join/#asterisk netpro25_ (~mmanning@64-238-176-105.ksg.apt.gru.net) |
| 17:14.07 | leifmadsen | Polysics: sounds like a queue :) |
| 17:14.08 | *** join/#asterisk TimeRider (steve@5ad08e49.bb.sky.com) |
| 17:14.34 | Polysics | leifmadsen, would that mean i need a queue with one agent for each SIP account? |
| 17:14.50 | Polysics | i do have queus up in the DB, so it would not be that bad |
| 17:17.06 | Polysics | at worst, i need a SIP account to ring busy if it is already on a call |
| 17:17.12 | Polysics | is that what call-limit is for? |
| 17:17.31 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 17:19.35 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 17:25.57 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:26.48 | Polysics | is a queue the only way to do that? |
| 17:27.04 | [TK]D-Fender | Polysics: Or you could try.... DIALPLAN |
| 17:27.14 | *** join/#asterisk Geminizer (~whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 17:27.17 | *** join/#asterisk outtolunc (~me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:27.31 | Geminizer | Hello all. Is there a way to get "Elapsed Time" of a channel using AMI ? |
| 17:27.48 | Polysics | [TK]D-Fender, i don't get waht you mean - obviously stuff is done using dialplans :-) |
| 17:27.50 | [TK]D-Fender | Geminizer: Yes |
| 17:28.14 | [TK]D-Fender | Polysics: Check phone. If buy = wait. Play message. Try again. |
| 17:28.27 | Geminizer | [TK]D-Fender: is "Elapsed Time" considered to be a variable? |
| 17:28.27 | Polysics | would that require AEL? |
| 17:28.50 | [TK]D-Fender | Polysics: Of course not |
| 17:29.11 | [TK]D-Fender | Geminizer: Go lok at the list of available vars |
| 17:29.18 | [TK]D-Fender | Geminizer: but that's not how I'd check |
| 17:29.25 | Polysics | i am missing the "check phone" part |
| 17:29.40 | [TK]D-Fender | Polysics: "core show application chanisavail" <- |
| 17:31.42 | *** join/#asterisk Acro (~omgponies@unaffiliated/acro) |
| 17:33.32 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 17:34.32 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:36.20 | *** join/#asterisk maximo (~maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 17:37.47 | Polysics | [TK]D-Fender, the wiki says ChanIsAvail is not supposed to be used to detect if a line is in use or not |
| 17:38.27 | Qwell | The wiki is wrong. |
| 17:38.44 | Polysics | Qwell, i am happier that way :-) |
| 17:38.47 | Qwell | That's kinda the entire point of ChanIsAvail |
| 17:38.59 | [TK]D-Fender | Qwell: Notice : Incoming non-personal attack with course language. |
| 17:39.08 | [TK]D-Fender | Polysics: FUCK THE FUCKING WIKI |
| 17:39.11 | [TK]D-Fender | :D |
| 17:39.15 | [TK]D-Fender | exhales |
| 17:39.25 | Polysics | can i come out from under my desk now? |
| 17:39.31 | [TK]D-Fender | Polysics: GO BACK!!!!!! |
| 17:39.42 | [TK]D-Fender | </bark> |
| 17:39.48 | Polysics | it puts the lotion on its skin or it gets the hose again? |
| 17:40.17 | Polysics | ok, so ChanIsAvail it is |
| 17:41.35 | roe | can someone clarify for me how a T-1 that has been split to do half phones (12 channels) and half data looks at layer 1? Is it two hand-offs or is some kind of IVAD router needed? |
| 17:42.29 | Polysics | thre is DEVICE_STATE too |
| 17:42.55 | *** join/#asterisk niekvlessert (~niekvless@ip80-101-235-201.hotspotsvankpn.com) |
| 17:43.23 | Polysics | at this point i should take a look at AEL, i suppose |
| 17:43.33 | Polysics | would probably make syntax easier |
| 17:43.45 | Polysics | too bad Adhearsion does not work with 1.6... |
| 17:44.15 | *** join/#asterisk hipitihop (~denis@203.132.229.18) |
| 17:45.46 | [TK]D-Fender | Polysics: total waste |
| 17:45.59 | [TK]D-Fender | Polysics: this less than half a dozen lines of dialplan. |
| 17:46.36 | Polysics | does AEL add overhead? |
| 17:47.00 | [TK]D-Fender | Polysics: Yes. AEL does nothing you can't do yourself in extensions.conf directly, and even les |
| 17:47.37 | Polysics | i guess the step after extensions.conf would be AGI then |
| 17:47.50 | Polysics | not wanting to look at it right now, just wondering |
| 17:48.45 | carrar | HELLO and Happy morning to everyone :) |
| 17:50.48 | codefreeze-lap | Polysics: well, yes and no, AEL inserts some comments for debug; a very small amt of overhead. Otherwise, it's about as tight as hand-written dialplans. |
| 17:51.34 | Polysics | i think i will stay with dialplans until i need AGI |
| 17:51.44 | Polysics | i was liking Adhearsion a lot, really |
| 17:52.00 | *** join/#asterisk vettehead (~mvarner@205.127.233.211) |
| 17:52.12 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 17:56.52 | Polysics | and am still missing for a simple system to interact with aMi based on events, which Adhearsion was providing |
| 17:59.35 | bmoraca_work | roe: at layer 1, it's four wires. |
| 18:00.04 | *** join/#asterisk Alagar (~Administr@122.164.89.242) |
| 18:00.34 | [TK]D-Fender | roe: the same as one that isn't split :) |
| 18:00.40 | [TK]D-Fender | bmoraca_work: Sometimes :) |
| 18:00.49 | roe | bmoraca_work, and the T-! card in the asterisk box just sees the 12 channels instead of 24? |
| 18:01.14 | [TK]D-Fender | roe: No... there is no "see. You need to configure it to match or "bad things" happen |
| 18:01.25 | bmoraca_work | roe: the T1 card in asterisk uses whatever channels you tell it to |
| 18:01.54 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 18:02.01 | bmoraca_work | roe: a T1 that is configured like this will have a MUX on either end. a T1 is simply 24 channels that you can do whatever you want with them. are you trying to get Asterisk to see both parts of the T1? |
| 18:02.23 | Geminizer | is it possible to access CDR variables using AMI? |
| 18:04.11 | *** join/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 18:05.22 | seanbright | yes. use GetVar |
| 18:07.09 | Cobraz | Hello! |
| 18:07.33 | Cobraz | I am tring to setup an SIP server with Asterisk, i keep getting this error: "...Not a local domain" |
| 18:07.44 | seanbright | no you don't |
| 18:09.07 | Cobraz | How do i fix this problem? I am using Bria in my office now, and it keeps telling me that the service is unavailable. Can't register... |
| 18:10.23 | *** join/#asterisk SomethingISODD (~Dan@d75-152-164-253.abhsia.telus.net) |
| 18:10.48 | SomethingISODD | Hello everyone does anyone know of any Java SIP or IAX phone that will work on a WIFI TV Mobile phone. |
| 18:14.22 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:14.40 | roe | bmoraca_work, no, we are adding a T-1 for 12 lines and the T-1 provider offered to bond the other half of that T-1 to our existing data T-1. The details of how that actually works eludes me a bit |
| 18:14.50 | roe | (practically, not theoretically) |
| 18:15.50 | Cobraz | Getting error when trying to register SIP client with Asterisk.. (Error msg: ".... Not a local domain").. Can you help? :-) |
| 18:16.02 | bmoraca_work | roe: they'll put a MUX on your side that will split the single T1 in from the provider into two 12 channel T1s, one for your PBX and one that will be part of an MLPPP group with your other data T1. if they're not providing the DSU for your existing data T1, you'll need to upgrade your router to take advantage of it |
| 18:16.32 | bmoraca_work | (probably) |
| 18:16.37 | roe | I believe they are providing the DSU |
| 18:16.53 | bmoraca_work | then they'll need to upgrade it themselves, which usually isn't an issue |
| 18:17.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.08 | bmoraca_work | i've never done MLPPP with varying speed T1s...could be interesting |
| 18:17.14 | [TK]D-Fender | bmoraca_work: Or you can jsut let your T1 card do it in * |
| 18:17.33 | bmoraca_work | [TK]D-Fender: good luck bonding that with an existing data T1 |
| 18:17.48 | [TK]D-Fender | bmoraca_work: that is much rarer. |
| 18:18.10 | [TK]D-Fender | bmoraca_work: I've seen that with IP-PRI's, but not anything CBR |
| 18:18.40 | bmoraca_work | roe: "integrated T1s", as these used to be known, are usually implemented where the voice is done over SIP to the provider now. |
| 18:19.47 | *** part/#asterisk sneaker (~sneaker@90.b160.bendtel.net) |
| 18:19.57 | roe | we are reluctant to SIP to the outside world as we don't trust their SLA and QoS |
| 18:20.35 | Cobraz | Please help? I'm googling my ass of here :P Could be nice with a chat about it ^^, |
| 18:20.51 | bmoraca_work | roe: the difference between these types of services and an ITSP is that these types of services are entirely ON-NET |
| 18:21.11 | roe | !ON-NET |
| 18:21.23 | roe | isn't there a bot in here? |
| 18:21.36 | bmoraca_work | roe: so, you're not relying on anyone else's network. therefore, you have end-to-end QoS and it is possible for the telco to offer and enforce an SLA |
| 18:21.56 | bmoraca_work | roe: basically, your voice traffic, even though it is SIP, never actually leaves the telco's network via IP |
| 18:22.42 | roe | so if I were confident in their abilities to manage our data T-1 line with a high availability and high QoS I should trust my SIP traffic to them as well? |
| 18:24.41 | *** join/#asterisk cweagans (~432aa645@gateway/web/freenode/x-jtvmvjxqfwvjyixu) |
| 18:24.49 | *** join/#asterisk afo0l (~afo0l@85.114.131.108) |
| 18:24.58 | cweagans | anybody in here have experience with the Cisco 7940 phones? I cannot get my handsets to register with Asterisk....looking at the sip debug messages right now (they're pasted at http://pastebin.com/Lwakk56F) |
| 18:25.18 | *** join/#asterisk bakermd (~bakermd@38.104.0.102) |
| 18:25.30 | bakermd | Does this look invalid to anyone? Getting an error... exten => 30,n,Set(savestatus=${CURL(http://callblast.stryden.net/listener.php?function=save&call_id=${callblastid}&response=${RESPONSE})}) |
| 18:25.58 | [TK]D-Fender | cweagans: nowhere in there do we see a register attempt |
| 18:26.18 | cweagans | [TK]D-Fender: heh, whoops. I'll go track that one down real quick |
| 18:26.21 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:26.27 | cweagans | I thought I grabbed it.. =/ |
| 18:27.07 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:28.21 | afo0l | hi, i setup an ISDN bound asterisk server, using a hfc card and zaphfc, now asterisk creates the channels etc, but dialing out is impossible |
| 18:28.36 | afo0l | Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 18:28.52 | afo0l | i tried finding some info on the web, but documentation is rare |
| 18:29.10 | afo0l | what is a good way to check if the isdn card itself works? |
| 18:30.16 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:31.49 | Cobraz | Does anyone know of a nice guide how to setup SIP accounts in Asterisk? Since noone can help me :P |
| 18:31.59 | tzafrir | afo0l, what version of asterisk? |
| 18:32.22 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:32.48 | afo0l | 1.6.2 |
| 18:32.52 | afo0l | ubuntu |
| 18:33.17 | LemensTS | anyone know of a sip provider that i can put on an ata to do faxing? |
| 18:33.21 | tzafrir | roe, you meant: ~ON-NET . But the bot does not know this either |
| 18:33.38 | roe | tzafrir, thanks |
| 18:34.12 | tzafrir | afo0l, do incoming calls work? |
| 18:34.20 | afo0l | i'm just trying that |
| 18:36.31 | cweagans | [TK]D-Fender: http://pastebin.com/R4abfREt <--registration is at the top. It looks like the phone tries to register, and asterisk tries to respond with a 200, but for some reason, that device is unreachable? |
| 18:37.59 | [TK]D-Fender | cweagans: looks fine. Cisco isn't answering the OPTIONS packetsm so turn off qualify <- |
| 18:38.10 | [TK]D-Fender | cweagans: * OK's the register |
| 18:38.15 | *** join/#asterisk Skeeter- (~skeeter@190-141.cgocable.ca) |
| 18:38.26 | cweagans | [TK]D-Fender: turn off qualify in sip.conf? |
| 18:38.36 | [TK]D-Fender | cweagans: yes |
| 18:38.43 | cweagans | okay, I'll try that |
| 18:39.32 | afo0l | tzafrir: no, incoming calls dont work either, and won't show up in the log |
| 18:39.49 | afo0l | so it must be about the config of the isdn card itself ? |
| 18:39.58 | tzafrir | afo0l, what's the output of: lsdahdi |
| 18:40.44 | afo0l | it lists 3 spans, stating each spans channels are (In use) |
| 18:40.53 | afo0l | 2 isdn cards, one dummy |
| 18:41.34 | afo0l | should i query you the whole output? |
| 18:41.40 | afo0l | thanks for helping me out |
| 18:42.03 | tzafrir | ~pb |
| 18:42.03 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:42.46 | afo0l | http://pastebin.org/108778 |
| 18:44.28 | *** join/#asterisk magicblaze007 (~piyush@fl-67-235-215-192.dhcp.embarqhsd.net) |
| 18:44.50 | cweagans | [TK]D-Fender: hmm. the phones are registering now (I had to set qualify=no and nat=no), but I'm still not able to call to/from the phones, or out through my VoIP provider. Watching the sip debug messages go by, I see a lot of Restransmitting messages. A set of them about every 10 seconds (they look very similar to the ones in my first paste) |
| 18:45.12 | [TK]D-Fender | cweagans: Know what I see? |
| 18:45.33 | magicblaze007 | voicepulse just increased my dues from $11 a month to $14 a month...have others also seen dues increases in recent times? (sip/pbx service) |
| 18:45.34 | cweagans | [TK]D-Fender: hard to say. enlighten me? |
| 18:45.35 | cweagans | :) |
| 18:45.44 | [TK]D-Fender | cweagans: NOTHING |
| 18:46.00 | cweagans | ROFLMAO |
| 18:47.00 | tzafrir | afo0l, what's the output of 'dahdi show channels' in the command line of asterisk? (rasterisk) |
| 18:47.37 | cweagans | okay. there's a lot of messages that fly by when I do 'sip set debug ip 192.168.0.5'. I'm not really sure what to paste...what would be of use? I'm pretty sure these phones are trying to kill me :) |
| 18:47.56 | afo0l | http://pastebin.com/jphEfmvF |
| 18:47.57 | afo0l | this |
| 18:47.59 | magicblaze007 | ITEM/ACCOUNT/REGULATORYCOMPLIANCEFEE: at $2.95 / which voip provider is recommended here? I need 4 incoming lines and i am willing to pay for outgoing calls... |
| 18:48.21 | afo0l | omg |
| 18:48.41 | afo0l | i think i may have messed up the contexts |
| 18:49.24 | cweagans | magicblaze007: not sure what you're asking, but speakeasy has been pretty good to me :) $35/trunk/month, so you'd be looking at $140/month, unlimited calling. |
| 18:49.43 | *** join/#asterisk rob (~wrab@uce.mx) |
| 18:50.17 | magicblaze007 | cweagans: I pay $11 for a phone number + 4 parallel incoming lines. They just increased it to $14. I am looking for something cheaper than that, if not, i'll pay $14 :) $140 sounds excessive to me for now. |
| 18:50.40 | Geminizer | Do you you know when you run 'core show channel [CHAN]' you get a whole dump of info? How can I use AMI to get values under the "-- General --" section ? |
| 18:51.00 | cweagans | magicblaze007: oh wow, who's that through? |
| 18:52.00 | Geminizer | in other words, is there another way to get those values (without running 'core show channel')? |
| 18:52.13 | magicblaze007 | cweagans: voicepulse.com |
| 18:52.49 | magicblaze007 | cweagans: if you sign up for them, please let me give you a referral. that way they give me $5 credit. |
| 18:53.10 | cweagans | okay. send me whatever info I need to do that. my nick @gmail.com |
| 18:53.20 | cweagans | I may or may not do that in the near-ish future :) |
| 18:53.42 | magicblaze007 | i understand, its a big pain to change providers :) |
| 18:54.00 | afo0l | tzafrir: http://pastebin.com/jphEfmvF |
| 18:54.45 | tzafrir | afo0l, do those contexts really exists? from-pstn and fr-pstn? |
| 18:54.50 | tzafrir | (in the dialplan) |
| 18:54.55 | magicblaze007 | cweagans: am sending you a referral. |
| 18:55.17 | magicblaze007 | its $15 for me if you sign up, 1st month is free :) |
| 18:55.48 | afo0l | tzafrir: fr-pstn does exist, the other doesnt, but channels 4 and 5 arent in use |
| 18:55.54 | afo0l | does that matter? |
| 18:56.44 | tzafrir | afo0l, if you want incoming calls: yes |
| 18:56.46 | *** join/#asterisk fors1 (~forsen@pat-tdc.opera.com) |
| 18:57.05 | afo0l | both have to exist even it only one is used? |
| 18:57.09 | afo0l | ok i'll fix it up then |
| 18:57.15 | tzafrir | hmm... sorry, I didn't read the last line. it doesn't matter |
| 18:57.15 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:57.54 | afo0l | do you know of any linux tool i could use to test the isdn card itself? |
| 18:58.19 | afo0l | and is it normal that if i plug the "landline" into the NT set card i get a kernel panic? |
| 18:58.26 | Kobaz | no |
| 18:58.37 | Kobaz | contact tech support for your card manufacturer |
| 18:59.00 | tzafrir | afo0l, in the CLI, run: core set verbose 3 |
| 18:59.14 | tzafrir | then try an incoming call and show the trace of it |
| 19:01.26 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 19:07.39 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 19:08.02 | timholum | hello I am wondering if there is any way for asterisk to detect studer dial tone on an incoming line? |
| 19:09.34 | afo0l | tzafrir: nothing at all, no log entry and no ringtone |
| 19:10.10 | tzafrir | afo0l, pri set debug 1 span 2 |
| 19:11.07 | afo0l | span for "fr-pstn" is 1 |
| 19:11.21 | afo0l | ah |
| 19:11.31 | afo0l | i set debugging there as well |
| 19:11.49 | afo0l | its "Sending set Asynchronous Balanced Mode Extended" |
| 19:13.29 | afo0l | perpetually |
| 19:14.59 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:17.03 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:20.02 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 19:21.38 | afo0l | tzafrir: i got 2 identical cards in there, one NT one TE, if i plug the cable in the wrong one i instantly get a kernel panic |
| 19:21.47 | afo0l | and i'm not 100% sure which is wich |
| 19:21.57 | afo0l | might be i need to go back to start |
| 19:22.04 | afo0l | and try a different driver |
| 19:22.07 | afo0l | or kerneö |
| 19:26.43 | Polysics | are tehre that many AGI and AMi differences between a.4 and 1.6? |
| 19:26.57 | Polysics | not in the number of commands, more fundamental ones |
| 19:27.06 | Polysics | so far i have noticed . intead of | |
| 19:27.13 | Polysics | *, |
| 19:28.26 | timholum | Does asterisk have any modules or addon's that can detect stutter dial tone? |
| 19:28.33 | timholum | I have looked an I cant find any |
| 19:30.49 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 19:37.38 | manxpower | timholum: no. It's not the function of a PBX |
| 19:37.51 | *** join/#asterisk ttwhy (~tekkno@p4FECF5BD.dip.t-dialin.net) |
| 19:38.17 | manxpower | Polysics: you mean other than the ones listed in the UPGRADE*.txt files included in the Asterisk tarball? |
| 19:38.26 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 19:38.56 | manxpower | | .vs. , is not an AMI or AGI think. It's an ASTERISK thing. Go re-read those UPGRADE*.txt files. |
| 19:39.14 | spenguin[work] | heh w00t https://developer.skype.com/silk |
| 19:39.41 | timholum | manxpower, thanks for the info |
| 19:40.46 | Polysics | manxpower, is that the reason why most of the third party stuff for 1.4 simply states "this doesn't work with 1.6, period"? |
| 19:43.32 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:45.59 | *** join/#asterisk Vrtigo1 (~vrtigo1@vpn.lpga.com) |
| 19:49.07 | Vrtigo1 | How can I set asterisk to answer a DID and initiate a monitoring session and not hang up until the calling party hangs up? |
| 19:52.02 | manxpower | Vrtigo1: Depends on what you mean by "monitor". |
| 19:52.20 | Vrtigo1 | The command "monitor" |
| 19:52.20 | manxpower | In Asterisk app_monitor records calls. |
| 19:53.25 | manxpower | Vrtigo1: Answer the DID, run monitor. |
| 19:53.32 | manxpower | No real way to prevent the caller from hanging up. |
| 19:53.40 | *** part/#asterisk magicblaze007 (~piyush@fl-67-235-215-192.dhcp.embarqhsd.net) |
| 19:53.45 | Vrtigo1 | Yes, thanks. That works beautifully for all of about 1 second, then asterisk hangs up. |
| 19:54.01 | manxpower | Vrtigo1: then you are doing something wrong |
| 19:54.33 | *** join/#asterisk sulex (~sulex@host-78-14-170-90.cust-adsl.tiscali.it) |
| 19:54.40 | Vrtigo1 | Any insight as to what that might be? Kind of hard to go wrong with answer, then monitor...\ |
| 19:54.48 | *** join/#asterisk netpro25_ (~mmanning@64-238-176-105.ksg.apt.gru.net) |
| 19:54.52 | manxpower | Vrtigo1: pastebin the cli output |
| 19:54.53 | manxpower | ~pb |
| 19:54.54 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:54.56 | Vrtigo1 | k |
| 19:56.09 | Vrtigo1 | http://pastebin.ca/1832359 |
| 19:56.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 19:56.39 | manxpower | Executing [3868680470@inbound:3] Set("SIP/vitel-inbound-00000154", "CALLFILENAME=20100310-145829-inbound-to-868680470-from-"LDIES PROF GLF" <3862744925>") in new stack |
| 19:56.47 | manxpower | That does not look much like a valid filename to me. |
| 19:56.57 | Vrtigo1 | It works fine in other contexts. |
| 19:58.11 | manxpower | so, if you touch CALLFILENAME=20100310-145829-inbound-to-868680470-from-"LDIES PROF GLF" <3862744925> I strongly doubt if it will work |
| 20:01.14 | manxpower | the extra quotes, the < > shell redirection, your filename is a mess. |
| 20:03.17 | *** join/#asterisk Z_God (~julius@wlan238202.mobiel.utwente.nl) |
| 20:03.48 | leifmadsen | huzzah! |
| 20:05.28 | manxpower | leifmadsen: you joined the Marines? |
| 20:05.32 | leifmadsen | never |
| 20:05.48 | Vrtigo1 | manxpower: changed the filename, same result http://pastebin.ca/1832369 |
| 20:06.17 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 20:06.47 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 20:07.44 | Vrtigo1 | basically, i need some way to tell asterisk not to hang up when it runs out of things to do |
| 20:08.17 | Vrtigo1 | if i turn autofallthrough off, are there any negative implications other than needing to make sure i clean up after calls properly? |
| 20:10.26 | [TK]D-Fender | Vrtigo1: what is * supposed to do at that point? |
| 20:10.51 | [TK]D-Fender | Vrtigo1: When you run out of things to do why the hell would you just sit around forever? |
| 20:11.17 | Vrtigo1 | [TK]D-Fender: continue recording the call |
| 20:11.26 | Vrtigo1 | perhaps I should clarify what I am trying to do... |
| 20:11.33 | [TK]D-Fender | Vrtigo1: What call? Who's talking there? |
| 20:11.50 | *** join/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 20:12.08 | [TK]D-Fender | Vrtigo1: So far there is jsut the inbound call sitting round not being prompted for anything. |
| 20:12.28 | *** join/#asterisk heliosj (~jeff@i216-58-41-253.cybersurf.com) |
| 20:12.33 | [TK]D-Fender | Vrtigo1: You may as well use Record() at that point. |
| 20:12.48 | Vrtigo1 | I want to be able to conference in a DID that goes to my * server when I want to record a call. For example, i'm in my car and I want to make a call that I want to have recorded, i dial my * DID, then conference in the 2nd party, and I want * to record the conversation. |
| 20:13.50 | Vrtigo1 | Essentially, I just want * to answer the call, then record the conversation until the call is disconnected. |
| 20:13.55 | manxpower | Vrtigo1: double check your format of the monitor command for your version of Asterisk? |
| 20:14.18 | manxpower | "core show application monitor" I hope you didn't use voip-info.org for find out the valid options for Monitor |
| 20:14.20 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 20:14.47 | Vrtigo1 | manxpower: is there a version specific command reference available? I did use voip-info.org. |
| 20:14.50 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 20:14.52 | manxpower | Vrtigo1: then stiop using the wrong application |
| 20:15.05 | [TK]D-Fender | Vrtigo1: then use Record() < |
| 20:15.09 | manxpower | Vrtigo1: YES! IN THE CLI "core show application X" where X is the application |
| 20:15.34 | manxpower | Vrtigo1: that is your ONLY source of official docs for the specific version of Asterisk you are using. |
| 20:16.26 | Vrtigo1 | manxpower: ok, thanks. according to those docs, monitor should do exactly what I want. |
| 20:16.53 | Vrtigo1 | "The channel's input and output voice packets are logged to files until the channel hangs up" |
| 20:17.11 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.35 | [TK]D-Fender | Vrtigo1: there is no outbound <- Just use Record |
| 20:18.33 | *** join/#asterisk Tim_Toady (~moi@188.4.62.188.dsl.dyn.forthnet.gr) |
| 20:18.52 | Vrtigo1 | [TK]D-Fender: right, i don't care if there's no outbound because everything i want recorded is inbound. I'm looking at the record docs and it says that the recording will be cancelled in the event the channel hangs up, which is exactly what I want to avoid. |
| 20:19.15 | Vrtigo1 | [TK]D-Fender: also with Monitor, I can tell it to record only inbound |
| 20:20.27 | [TK]D-Fender | ... |
| 20:20.33 | [TK]D-Fender | Vrtigo1: there IS ONLY INBOUND |
| 20:20.44 | [TK]D-Fender | Vrtigo1: You call INTO IT. There is no 3rd party! |
| 20:21.12 | Vrtigo1 | [TK]D-Fender: I see what you're saying, but doesn't outbound refer to the connection between * and the caller? |
| 20:21.27 | leifmadsen | [TK]D-Fender: chill |
| 20:21.38 | [TK]D-Fender | Vrtigo1: And DUH... if you hang up it'd BETTER stop recording or you'll be recording non-stop 24/7 with nothing on the other end. Wnat to see your HD free space disappear for no good reason? There's a start |
| 20:22.08 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 20:22.20 | [TK]D-Fender | Vrtigo1: You only use Monitor when you're about to do something like a Dial following it. |
| 20:22.44 | [TK]D-Fender | Vrtigo1: or a MeetMe, etc... but that one has its own options to isolate the entire conference |
| 20:23.03 | [TK]D-Fender | Vrtigo1: But it is not at all appropriate to use it for a 3-way merged-in call |
| 20:23.13 | Vrtigo1 | [TK]D-Fender: ok |
| 20:23.48 | [TK]D-Fender | Vrtigo1: Your 3-way only has audio coming from the device that called in. The local channel that you merge in isn't contributing audio. |
| 20:24.50 | Vrtigo1 | [TK]D-Fender: i'm a bit confused, i thought the "local channel that I merge in" and the "device that called in" were the same. |
| 20:25.15 | Vrtigo1 | [TK]D-Fender: or are you referring to the monitor command as a channel that i'm merging with the inbound call? |
| 20:26.09 | [TK]D-Fender | Vrtigo1: Your phone is the device. it was already on a call. That's parties A & B. A then calls in to that exten and it needs to record A & B. Not itself <--- A passes on B's audio in the call to that exten. |
| 20:26.51 | leifmadsen | Vrtigo1: When you place a call with A (to B) and the A channel triggers the Monitor(), the Monitor()'ing (recording) is going to follow the channel that was created by phone A |
| 20:26.53 | [TK]D-Fender | Vrtigo1: so audio from both is all "inbound" to that exten you 3-way. |
| 20:27.14 | leifmadsen | Vrtigo1: whatever Phone A hears, is what the recording is going to contain |
| 20:27.51 | Vrtigo1 | leifmadsen: whatever Phone A hears, plus any audio coming from Phone A, I assume? |
| 20:27.57 | leifmadsen | of course |
| 20:28.21 | leifmadsen | whatever the original channel that Phone A created which launched the Monitor() application |
| 20:28.30 | leifmadsen | if that channel goes away, then the recording stops |
| 20:28.32 | [TK]D-Fender | Still don't use monitor, because that call has nothing to do sitting around empty... |
| 20:28.33 | Vrtigo1 | leifmadsen: right, i get that. so it seems to me that I should be able to Monitor() that and just tell monitor to only record the inbound |
| 20:28.58 | Vrtigo1 | [TK]D-Fender: yes, i think that's the root issue i'm running into, the timeout being hit |
| 20:29.03 | leifmadsen | Vrtigo1: in that case, you'd only hear what Phone A said |
| 20:29.06 | [TK]D-Fender | Vrtigo1: Monitor = both directions. but the way you call that exten, * isn't playing any audio. There is nothing to record. |
| 20:29.24 | [TK]D-Fender | Vrtigo1:vrtSo just use Record() |
| 20:29.37 | leifmadsen | [TK]D-Fender: huh? even I have no idea what you're saying |
| 20:29.56 | leifmadsen | [TK]D-Fender: he wants to conference in someone into a conversation, but he only wants to record the audio from one direction (is my understanding) |
| 20:30.12 | [TK]D-Fender | leifmadsen: he's using a 3-way call to bring the recording exten into the mix. Not used as a precusor to a Dial <- |
| 20:30.26 | Vrtigo1 | leifmadsen: i think i've got everyone sufficiently confused. |
| 20:30.33 | leifmadsen | Vrtigo1: fuck ya :) |
| 20:30.39 | Vrtigo1 | leifmadsen: :P |
| 20:30.40 | SomethingISODD | Hello everyone does anyone know of any Java SIP or IAX phone that will work on a WIFI TV Mobile phone. |
| 20:30.43 | leifmadsen | plus I was working on something and only caught the last half of it |
| 20:30.54 | leifmadsen | SomethingISODD: nope |
| 20:31.17 | leifmadsen | Vrtigo1: I'd start again |
| 20:31.23 | Vrtigo1 | leifmadsen: here's the high level...i want to be able to initiate a call from my cell to *, then conference another party in on the existing call from my cell |
| 20:31.36 | SomethingISODD | ok ty leifmadsen |
| 20:31.54 | Vrtigo1 | leifmadsen: so, a 3-way call with my cell talking to *, and another party |
| 20:31.58 | leifmadsen | Vrtigo1: ok, then I'd use a feature code to do that (features.conf) -- DTMF based |
| 20:32.14 | bmoraca_work | is anyone here familiar with reselling AT&T copper or know of any CLECs that will do that on a wholesale basis? |
| 20:32.26 | [TK]D-Fender | Vrtigo1: So far your description fails to include 3 people <- |
| 20:32.31 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 20:32.52 | Vrtigo1 | [TK]D-Fender: i never said anything about 3 people, i said a 3-way call with * being one of the dialed parties |
| 20:32.53 | [TK]D-Fender | Vrtigo1: Try again. Where does the first call originate from? Where does the 2nd call come from? |
| 20:32.56 | leifmadsen | Vrtigo1: assuming the first leg of the call was a direct call to your cell phone -- then that is not possible |
| 20:33.08 | leifmadsen | Vrtigo1: oh I think I get it |
| 20:33.18 | *** join/#asterisk Alagar (~Administr@122.164.89.242) |
| 20:33.33 | Vrtigo1 | leifmadsen: the root objective is to be able to record a conversation between myself and one other party |
| 20:33.40 | leifmadsen | Vrtigo1: Mom calls your cell. You then call your Asterisk box to record the call. You then hit "conference" on your cell. |
| 20:33.43 | [TK]D-Fender | leifmadsen: Vrtigo1 So you want the call to look like its coming from your cell (because it is), and bring in just for recording, right? |
| 20:33.52 | [TK]D-Fender | Vrtigo1: ^ |
| 20:33.55 | leifmadsen | Vrtigo1: then yes, just call Asterisk which answers with Record() |
| 20:34.01 | Vrtigo1 | leifmadsen: exactly, although i would probably call * first, and call the second party after |
| 20:34.08 | leifmadsen | Vrtigo1: ya, order doesn't matter |
| 20:34.13 | [TK]D-Fender | Vrtigo1: Which leaves us precisely where I told you before. RECORD(), not MONITOR() |
| 20:34.21 | leifmadsen | Vrtigo1: just call Asterisk, have it do a Record(), then conference in your other party |
| 20:34.34 | leifmadsen | Vrtigo1: you will end up recording both people in the call -- there is no way around that |
| 20:35.03 | [TK]D-Fender | Vrtigo1: No way around... because its your CELL that passes both audio as its own |
| 20:35.11 | Vrtigo1 | leifmadsen: thanks, the only other question I have is how can I make * finish recording when I hang up, the docs say it will cancel the recording when that happens |
| 20:35.41 | [TK]D-Fender | Vrtigo1: It won't "canel" anything. It will simple end recording. |
| 20:35.46 | leifmadsen | Vrtigo1: option 'k' |
| 20:35.50 | [TK]D-Fender | Vrtigo1: Did you think you'd lose it somehow? |
| 20:35.52 | leifmadsen | [TK]D-Fender: not true |
| 20:36.00 | [TK]D-Fender | leifmadsen: Oh? |
| 20:36.03 | Vrtigo1 | [TK]D-Fender: "If the user hangs up during a recording, all data will be lost and the application will terminate." |
| 20:36.04 | leifmadsen | options: k: Keep recording if channel hangs up. |
| 20:36.21 | leifmadsen | Vrtigo1: option 'k' (may not be available on 1.4... not sure when that was added?) |
| 20:36.24 | Vrtigo1 | leifmadsen: perfect, that's exactly what I needed, just didnt read down far enough |
| 20:36.29 | leifmadsen | Vrtigo1: :) |
| 20:36.33 | leifmadsen | Vrtigo1: hope that helped |
| 20:36.44 | leifmadsen | [TK]D-Fender: see how much more efficient that was? |
| 20:36.48 | Vrtigo1 | leifmadsen: yep, i think i should have everything i need thank you both |
| 20:36.56 | [TK]D-Fender | leifmadsen: Who is there to record if they hang up? |
| 20:37.03 | [TK]D-Fender | leifmadsen: that makes no sense whatsoever |
| 20:37.12 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 20:37.18 | [TK]D-Fender | leifmadsen: Record indefinately and chew up HD? |
| 20:37.21 | leifmadsen | [TK]D-Fender: Who is there to record if they hang up? I don't understand your question |
| 20:37.23 | Vrtigo1 | [TK]D-Fender: there's no one there to hang up, but it gives me the opportunity to do something with the recording after they hang up |
| 20:37.24 | leifmadsen | [TK]D-Fender: NO! |
| 20:37.37 | Vrtigo1 | [TK]D-Fender: err, to record i mean |
| 20:37.40 | leifmadsen | [TK]D-Fender: if you hang up, by default the Record() application throws away the recording |
| 20:38.02 | Vrtigo1 | leifmadsen: so then i'd just stop the recording upon hangup correct? |
| 20:38.05 | leifmadsen | [TK]D-Fender: unless you end it with a DTMF to signal you're done, or use the 'k' option, which ends the recording, and keeps it |
| 20:38.12 | [TK]D-Fender | leifmadsen: I think I see the linguistic snafu <- |
| 20:38.15 | leifmadsen | Vrtigo1: when you hangup, the recording will stop |
| 20:38.36 | [TK]D-Fender | leifmadsen: Keep the file recorded to that poitn, not "keep it alive and recording indefinately" :) |
| 20:38.41 | leifmadsen | [TK]D-Fender: I don't see how you could... this is very clear to me: "If the user hangs up during |
| 20:38.41 | leifmadsen | a recording, all data will be lost and the application will terminate. |
| 20:38.41 | leifmadsen | " |
| 20:39.02 | leifmadsen | "Keep *the* recording..." might be better |
| 20:39.06 | leifmadsen | in fact I'll fix it now |
| 20:39.21 | leifmadsen | it's not "Keep recording indefinitely" |
| 20:39.27 | [TK]D-Fender | Vrtigo1: I believe I agree with leifmadsen that "k" is what you'll want. |
| 20:39.28 | leifmadsen | because that would make no sense |
| 20:39.43 | Vrtigo1 | leifmadsen: i had the same thought - i thought that's what it meant as well, and that i would have to initiate some process to stop the recording upon hangup |
| 20:39.55 | leifmadsen | Vrtigo1: heh no, I guess I never read it like that :) |
| 20:39.59 | [TK]D-Fender | leifmadsen: Ok, we're all good with this... it had 2 distinct and entirely valid interpretations.... I of course chose the WRONG one ;) |
| 20:40.00 | leifmadsen | I will update the documentation |
| 20:41.02 | [TK]D-Fender | leifmadsen: Thanks... |
| 20:41.27 | [TK]D-Fender | Vrtigo1: Ok, so crisis aborted. Record() w/ "k" |
| 20:45.25 | *** join/#asterisk jshriver (~jshriver@72.240.39.37) |
| 20:45.28 | jshriver | Greetings everyone |
| 20:45.51 | jshriver | when editing the extension.conf is there an extern command to dial if the previous dial attempt failed? |
| 20:46.01 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 20:46.10 | jshriver | Basically doing this now: |
| 20:46.13 | jshriver | exten => 0,1,Playback(transfer,skip) ; "Please hold while..." |
| 20:46.13 | jshriver | exten => 0,2,Dial(SIP/300&SIP/301&SIP/312,15,rt) |
| 20:46.13 | jshriver | exten => 0,n,Voicemail(u300) ; Voicemail (unavailable) |
| 20:46.35 | jshriver | But want to add another Dial() after the 2nd line so if noone at those 3 extensions pick up, it'll try dialing another set of extensions |
| 20:46.52 | leifmadsen | [TK]D-Fender, Vrtigo1: of course this is only a problem on 1.6.2 and trunk -- no idea why the information is different |
| 20:47.18 | Vrtigo1 | leifmadsen: i'm running 1.6, figures |
| 20:47.19 | leifmadsen | [TK]D-Fender, Vrtigo1: in 1.6.0, and 1.6.1, it says, "Keep recorded file upon hangup" |
| 20:47.29 | [TK]D-Fender | leifmadsen: I won't ask.. you're working on fixing it. I don't care about problems you're already taking care of :) |
| 20:47.31 | leifmadsen | Vrtigo1: 1.6 doesn't mean anything, as there is no such thing as 1.6 |
| 20:47.34 | Vrtigo1 | leifmadsen: 1.6.2.2 specifically |
| 20:47.42 | leifmadsen | Vrtigo1: 1.6.2 != 1.6 |
| 20:47.45 | jshriver | also what is the 3rd field in extern mean? |
| 20:47.59 | leifmadsen | 1.6.0, 1.6.1, and 1.6.2 are all separate branches much like 1.2 and 1.4 are separate |
| 20:48.05 | leifmadsen | ~asterisk-versioning |
| 20:48.06 | [TK]D-Fender | jshriver: huh? |
| 20:48.15 | leifmadsen | ~asteriskversioning |
| 20:48.16 | infobot | asteriskversioning is, like, http://www.asterisk.org/asterisk-versions |
| 20:48.20 | Vrtigo1 | leifmadsen: i didn't realize that |
| 20:48.26 | leifmadsen | Vrtigo1: please read the above |
| 20:48.35 | jshriver | in the extensions.conf files what is the 2nd field mean ? extern => 0,X,Dial() what does X mean? |
| 20:48.52 | Vrtigo1 | jshriver: the priority |
| 20:49.02 | leifmadsen | jshriver: http://astbook.asteriskdocs.org |
| 20:49.07 | leifmadsen | jshriver: see chapter 5 |
| 20:49.10 | jshriver | ok |
| 20:49.35 | jshriver | if you have multiple extern => dial() entries will it try dialing if noone answers the previous dial() attempt? or do I have to add something else |
| 20:49.55 | leifmadsen | jshriver: it will continue one if there is a busy, or timeout happens |
| 20:49.58 | leifmadsen | s/one/on/ |
| 20:50.17 | leifmadsen | jshriver: again, answers obtained via documentation, that's why it's there :) |
| 20:50.24 | jshriver | ok |
| 20:50.43 | jshriver | sorry I really loath having to take care of this system not my field was dumped on me a year ago, so learning just enough to get it working |
| 20:50.49 | *** part/#asterisk manxpower (~ewieling@216.186.151.147) |
| 20:50.52 | jshriver | er keep it working lol |
| 20:50.56 | jshriver | thanks for the book will help |
| 20:52.12 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:53.37 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 20:58.10 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 21:06.15 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 21:12.11 | *** part/#asterisk Vrtigo1 (~vrtigo1@vpn.lpga.com) |
| 21:21.19 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 21:23.23 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-457f22ad.dyn.optonline.net) |
| 21:28.48 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 21:34.58 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 21:34.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:36.35 | *** part/#asterisk FuriousGeorge (~chatzilla@ool-457f22ad.dyn.optonline.net) |
| 21:38.46 | redax | hi. |
| 21:40.28 | redax | I have an extension where a gate opener operates, I have to call the extension, and enter a PIN code like: Dial(SIP/900,,D(ww1234)) |
| 21:41.04 | redax | I'd like to make the same application, just without picking up the caller. |
| 21:41.45 | redax | how one could call an extension and play some DTMF without accepting the incoming call? |
| 21:43.16 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 21:46.36 | nightrid3r | how is the gateopener going to read the dtmf whitout accepting the call? |
| 21:53.37 | [TK]D-Fender | redax: Use M() and don't accept the call |
| 21:58.24 | Kobaz | i need a nap |
| 22:00.08 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:06.22 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 22:07.08 | hackeron | I'm trying to configure a polycom 501 phone with asterisk and I'm getting: "WARNING[2243]: chan_sip.c:12673 check_auth: username mismatch, have <reception1>, digest has <###4######" -- what does that mean? |
| 22:08.19 | hackeron | where in the polycom configuration do I change this "digest"? |
| 22:09.29 | Kobaz | your sip username on your polycom is set wrong, very weong |
| 22:09.52 | hackeron | it's set to reception1 |
| 22:10.00 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 22:10.02 | Kobaz | the digest is in the sip packet it's getting for registration/calling |
| 22:10.04 | hackeron | well, I don't have username, I hace auth user id and address |
| 22:10.08 | Kobaz | hackeron: apparently not |
| 22:10.50 | hackeron | Kobaz: when I open the configuration it says Address: reception1 - Auth User ID: reception1 - what should I change? |
| 22:11.02 | Kobaz | you're using the web interface? |
| 22:11.15 | Kobaz | that doesn't look like a valid address |
| 22:11.41 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 22:11.44 | hackeron | Kobaz: yeah, the web interface |
| 22:11.57 | Kobaz | you should provision using ftp. it's much easier |
| 22:13.27 | hackeron | Kobaz: it's set up to get the bootrom and app using http and my config file is empty - let me try putting the auth user id in the config file |
| 22:14.15 | *** join/#asterisk Jhirley (~Jhirley@mail.mmdlaw.com) |
| 22:14.19 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 22:14.19 | Kobaz | reg.1.displayName reg.1.auth.userId reg.1.auth.password reg.1.server.1.address reg.1.server.1.port |
| 22:14.23 | Kobaz | you'll need hose |
| 22:14.24 | Kobaz | those |
| 22:16.02 | hackeron | Kobaz: so I just put that 1 line in my empty file? |
| 22:16.37 | hackeron | Kobaz: oh wait, so each on separate line, like say reg.1.displayName Reception\n etc? |
| 22:16.44 | Jhirley | what do folks in irc land recomend for telecom billing management ? |
| 22:17.40 | hackeron | Kobaz: would you mind pastebinning a simple sample config or do you have a link that shows the syntax and config options? |
| 22:19.18 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 22:22.20 | adnc | my system-time is ok, but still the value of strftime gives wrong time back. for example this variable holds a time that differs by two hours CALLFILENAME=${STRFTIME(${EPOCH},GMT+1,%d%m%Y-%H%M)}.${CALLERID(num)}.${MACRO_EXTEN} |
| 22:22.35 | adnc | although my system time is gmt+1 |
| 22:22.42 | adnc | what could be wrong? |
| 22:22.55 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.3) |
| 22:24.12 | LemensTS | I can call my cell phone from sip phone and see cid, but i can call an att landline and it says unknown. Any thoughts on that? |
| 22:25.03 | LemensTS | I mean just the number, not passing the actual caller name |
| 22:26.15 | Kobaz | hackeron: http://pastebin.ca/1832610 |
| 22:26.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 22:27.01 | Kobaz | LemensTS: your land line provider is not pulling name from the callerid dip service it's using |
| 22:27.53 | hackeron | Kobaz: thanks! |
| 22:27.54 | *** join/#asterisk clintc (~clintc@n128-227-15-193.xlate.ufl.edu) |
| 22:28.54 | Kobaz | make sure you adjust linekeys and etc for your specific phone |
| 22:29.09 | bjhaid | i am trying to reload the diaplan after editing my extensions.conf file, but the diaplan reload command does not work |
| 22:29.27 | Kobaz | ~details |
| 22:29.28 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 22:30.31 | hackeron | hmm, when I try to dial out, I see: "[Mar 10 22:30:07] WARNING[3681]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)" |
| 22:30.47 | Kobaz | hackeron: your dahdi config is broken. or your line is down |
| 22:31.08 | hackeron | Kobaz: this is my dahdi show status and channels: http://pastie.org/864091 |
| 22:31.38 | hackeron | and dahdi_cfg -vvv looks fine |
| 22:32.29 | Kobaz | ls -al /dev/dahdi |
| 22:32.38 | Kobaz | as it readable by asterisk? |
| 22:33.11 | hackeron | Kobaz: seems to be: http://pastie.org/864096 |
| 22:34.23 | Kobaz | paste the rest of your configs too |
| 22:34.51 | *** join/#asterisk doneir (~cbrunker@appenp.lnk.telstra.net) |
| 22:35.08 | hackeron | Kobaz: http://pastie.org/864101 |
| 22:35.25 | doneir | does anyone have a link to a decent .ael vim syntax file, the one on voip-info.org is a bit under developed |
| 22:36.48 | doneir | http://vim.sourceforge.net/scripts/script.php?script_id=1900 seems to be the best bet |
| 22:37.47 | Kobaz | too bad it's not a sangoma card, you could have also gotten voltage levels to see if the card is working/connected right |
| 22:39.07 | Deeewayne | does anyone know of an AudioCodes vendor that doesn't suck when it comes to support ? |
| 22:39.40 | Kobaz | i've been getting my stuff from e4 |
| 22:39.42 | hackeron | Kobaz: hmm, well, if I do dahdi_monitor I can see every ring when I dial the number |
| 22:40.40 | hackeron | Kobaz: but the call is not being picked up by asterisk, I just see CID errors and I get that "unknown error" when I dial out -- any ideas at all what I could try? |
| 22:40.52 | Kobaz | oh right you were here the other day too |
| 22:40.57 | Kobaz | did you talk to digium support? |
| 22:41.14 | Kobaz | configs look fine, dahdi output looks fine... check your syslog/dmesg for errors |
| 22:41.28 | hackeron | Kobaz: no, had to disconnect asterisk and plug in a normal phone temporarily |
| 22:41.39 | hackeron | Kobaz: just plugged the lines back in as it's out of hours now :) |
| 22:42.06 | Kobaz | i've never worked with uk stuff... it looks like a card/driver issue |
| 22:42.21 | Kobaz | but in the us... all that would work |
| 22:42.28 | hackeron | Kobaz: ok, I'll give them a ring now (if they're still open) |
| 22:42.39 | [TK]D-Fender | hackeron: Where do we see the failed call? |
| 22:42.53 | *** join/#asterisk wpbrown (~wpbrown@wh-gtw-0001.woolfharris.com) |
| 22:43.01 | [TK]D-Fender | hackeron: You seem to be showing us everything but the dead body. |
| 22:43.08 | Kobaz | yeah... one more bit |
| 22:43.09 | Deeewayne | [TK]D-Fender, its right there ---> . |
| 22:43.12 | Kobaz | hackeron: what are you dialing |
| 22:43.15 | [TK]D-Fender | hackeron: Don't ask for an autopsy until you do |
| 22:45.13 | wpbrown | I have a general question: Have you guys ever had a issue with Asterisk/Sangoma PRI card where cell phones can't dial extentions very well? The IVR says "Invalid number" If so where should I begin looking to tweak this issue? |
| 22:45.34 | Kobaz | wpbrown: asterisknow? freepbx? |
| 22:45.56 | wpbrown | Asterisk Open Source package.. |
| 22:46.06 | Kobaz | so what's this ivr you speak of |
| 22:46.07 | wpbrown | Never installed from the 123 cd's before. |
| 22:46.36 | Kobaz | wpbrown: did you download the build the asterisk source from the website? is that what you're saying? |
| 22:46.40 | drmessano | [TK]D-Fender: Except if the box bursts into flames. Then we can assume it's an OD, skip the autopsy and go straight to the condemnation. |
| 22:46.42 | wpbrown | Correct. |
| 22:46.58 | Kobaz | okay so... you'll need to give more details |
| 22:47.00 | wpbrown | This version has been in service since July. |
| 22:47.06 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.1) |
| 22:47.39 | wpbrown | Say for instance if you were to call my office.. from a cell you would dial a ext 6016 for instance.. asterisk will tell you that you dialed a invalid ext |
| 22:47.49 | Kobaz | dtmf reading issues |
| 22:47.50 | [TK]D-Fender | wpbrown: How about actually providing version #['s of things? |
| 22:48.03 | wpbrown | sure one second .. let me log into the box |
| 22:48.17 | hackeron | Kobaz: anything I dial shows that error instantly |
| 22:48.29 | hackeron | [TK]D-Fender: in asterisk -Rvd |
| 22:48.29 | Kobaz | but what specifically are you dialing |
| 22:48.38 | hackeron | [TK]D-Fender: dead body? |
| 22:48.41 | Kobaz | if you are dialing DAHDI/foobar, it's obviously not going to work |
| 22:48.54 | Kobaz | you can x out the phone number, but show the first part |
| 22:49.03 | hackeron | Kobaz: dialing DAHDI/131 |
| 22:49.06 | Deeewayne | hackeron, if you pastebin something there is a chance a couple people might actually look at what is happening |
| 22:49.18 | [TK]D-Fender | hackeron: You seem to have some comprehension issues here.... |
| 22:49.23 | Kobaz | hackeron: your dial string is wrong... you need to specify a group |
| 22:49.30 | hackeron | <PROTECTED> |
| 22:49.31 | [TK]D-Fender | hackeron: SHOW US THE ENTIRE DAMN CALL |
| 22:49.33 | hackeron | <PROTECTED> |
| 22:49.36 | hackeron | [Mar 10 22:49:10] WARNING[3700]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 22:49.40 | Kobaz | okay that's better |
| 22:49.41 | Deeewayne | eep |
| 22:49.56 | *** join/#asterisk manxpower (~ewieling@216.186.151.147) |
| 22:49.57 | bjhaid | I am a newbie having problem reloading the diaplan after editing the extensions.conf file, diaplan reload command does not work |
| 22:50.23 | hackeron | [TK]D-Fender: heh, sorry, I did earlier - forgot not everyone seen it :) |
| 22:50.39 | manxpower | Whoo! Digium just sent me an e-mail thanking me for registering my product! Since I've not registered a Digium product in at least 4 years....... |
| 22:50.51 | Deeewayne | manxpower, congratulations |
| 22:50.53 | Deeewayne | :-) |
| 22:51.04 | drmessano | Its never too late to say "Thank you" |
| 22:51.07 | Kobaz | hackeron: and you've tried other channels... and you've confirmed you have dialtone on all channels? |
| 22:51.08 | manxpower | I hope their other departments move faster than product registration |
| 22:51.16 | [TK]D-Fender | hackeron: "do it again, and include "core show channels concise" and "dahdi show channel 2" just prior |
| 22:51.59 | [TK]D-Fender | bjhaid: who is the owner of the file currently? |
| 22:52.29 | Kobaz | bjhaid: like i said before... you'll have to provide more details... a console output paste would be a good starter |
| 22:52.31 | hackeron | Kobaz: a normal phone has dialtone and I tried 2 channels (there are only 2 lines) |
| 22:52.32 | bjhaid | i have the roots right, its in etc/asterisk folder |
| 22:53.14 | hackeron | [TK]D-Fender: http://pastie.org/864127 |
| 22:53.30 | wpbrown | [TK]D-Fender, Kobaz Asterisk 1.6.0.9 |
| 22:53.44 | bjhaid | kobaz: my problem is that my modem does not have drivers for linux so i cannot paste console output |
| 22:54.25 | adnc | my system-time is ok, but still the value of strftime gives wrong time back. for example this variable holds a time that differs by two hours CALLFILENAME=${STRFTIME(${EPOCH},GMT+1,%d%m%Y-%H%M)}.${CALLERID(num)}.${MACRO_EXTEN} |
| 22:54.43 | adnc | what could be wrong= |
| 22:55.25 | [TK]D-Fender | bjhaid: MODEM? huh? |
| 22:55.50 | [TK]D-Fender | bjhaid: And what rights on on the other files? Who is * running as? |
| 22:56.07 | [TK]D-Fender | adnc: You aren't showing enough |
| 22:56.18 | adnc | [TK]D-Fender, what else could i show? |
| 22:56.40 | [TK]D-Fender | adnc: That isn't a complete valid syntax. There is no app.. no exten. we do not see the complete attempt to see what DID match, etc |
| 22:56.51 | adnc | i see |
| 22:57.07 | manxpower | adnc: you don't know what's important for [TK]D-Fender to see. |
| 22:57.08 | hackeron | [TK]D-Fender: any ideas? |
| 22:57.18 | manxpower | so pastebin everything |
| 22:57.22 | bjhaid | Fender: asterisk is running as root |
| 22:57.22 | [TK]D-Fender | adnc: Don't show me a catalog picture of your car and then ask what's wrong with the one you actually own. |
| 22:57.30 | adnc | hehe |
| 22:57.31 | [TK]D-Fender | hackeron: Not sure... |
| 22:57.35 | KavanS | anyone use voicepulse and have occasional DTMF issues? |
| 22:57.43 | Kobaz | hackeron: how's digium tech support coming along? |
| 22:57.57 | hackeron | Kobaz: called them, they said to register the card |
| 22:58.08 | Kobaz | did you register it? |
| 22:58.09 | hackeron | Kobaz: just about to take apart the PBX to get the serial number |
| 22:58.13 | Kobaz | ah |
| 22:58.26 | Kobaz | that's why i like dealing with sangoma |
| 22:58.28 | adnc | [TK]D-Fender, http://pastebin.com/0VD1EvJX |
| 22:58.30 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:58.40 | adnc | here i use this variable |
| 22:58.58 | manxpower | shouldn't the serial number be on the paperwork that came with the card? |
| 22:59.03 | [TK]D-Fender | hackeron: Signalling Type: FXS Kewlstart <--- is this legit where you are? |
| 22:59.25 | Kobaz | hackeron: the other day he told me he's got another box with the same settings hooked up to the same telco |
| 22:59.30 | hackeron | [TK]D-Fender: I tried loopstart too and the card won't let me set groundstart |
| 22:59.37 | [TK]D-Fender | adnc: Where is the complete failed call? Where are the extens that CALL that macro? |
| 22:59.39 | hackeron | Kobaz: yeah, but a few miles away |
| 22:59.53 | [TK]D-Fender | hackAnd difference on loopstart? |
| 23:00.02 | [TK]D-Fender | hackeron: Any difference on loopstart? |
| 23:00.07 | hackeron | [TK]D-Fender: nope, same |
| 23:00.21 | [TK]D-Fender | hackeron: I don't see a reason off-hand yet |
| 23:01.06 | adnc | [TK]D-Fender, the exten that calls the macro is there. i'll copy the output from the cli when a call gets in |
| 23:01.20 | hackeron | [TK]D-Fender: any ideas what else I can do to diagnose? -- the lines seem to be OK, a normal cheap analog phone works and I see the rings when I do dahdi_monitor on the line and dial the number |
| 23:01.47 | [TK]D-Fender | hackeron: REALLY not sure why this isn't working. go get Digium support in on this... |
| 23:02.05 | hackeron | [TK]D-Fender: ok, just got the serial number, going to file a support ticket, thanks |
| 23:02.05 | Kobaz | hackeron: seems like a card/driver issue |
| 23:02.39 | hackeron | Kobaz: the official driver didn't work at all, I had to compile from svn trunk not to get kernel oops errors - so yeah maybe |
| 23:02.48 | [TK]D-Fender | Kobaz: chan_dahdi is looking fine... the channel dumps OK. AFAIK * won't wait on dialtone, and this is dumb analog. |
| 23:02.53 | Kobaz | i should have asked this forever ago... what kernel version? |
| 23:02.59 | adnc | does someone know where STRFTIME() gets his values from? from the operating systems environment? |
| 23:03.20 | manxpower | adnc: what specific values? |
| 23:03.24 | [TK]D-Fender | adnc: You GIVE it the values. |
| 23:03.44 | hackeron | Kobaz: 2.6.31-19-server |
| 23:03.46 | Kobaz | hackeron: uname -a |
| 23:03.49 | [TK]D-Fender | adnc: Go look in the nearest mirror |
| 23:03.50 | Kobaz | hackeron: bad |
| 23:03.55 | Kobaz | hackeron: too new |
| 23:03.58 | hackeron | Kobaz: lol, what? |
| 23:04.13 | adnc | [TK]D-Fender, whats wrong? |
| 23:04.14 | Kobaz | dahdi has issues > 2.6.30 |
| 23:04.21 | hackeron | Kobaz: even svn trunk? |
| 23:04.30 | Kobaz | i wouldn't trust svn trunk for production |
| 23:04.58 | Kobaz | get the latest 2.6.27 kernel and rebuild dahdi |
| 23:05.16 | hackeron | Kobaz: don't think that's available anymore for my distribution (ubuntu server) |
| 23:05.20 | Kobaz | The latest stable 2.6.27 version of the Linux kernel is: 2.6.27.45 |
| 23:05.37 | manxpower | He has a kernel from the future! |
| 23:05.47 | Kobaz | well no... heh |
| 23:05.51 | Kobaz | The latest stable 2.6.31 version of the Linux kernel is: 2.6.31.12 |
| 23:05.53 | adnc | [TK]D-Fender, what are you talking about looking into a mirror? |
| 23:06.10 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 23:06.11 | Kobaz | continues to finger kernel.org |
| 23:06.23 | [TK]D-Fender | adnc: YOU pass it the value. It doesn't get it from "just anywhere" it was a PARAMETER. |
| 23:06.43 | adnc | [TK]D-Fender, yes right, the ${EPOCH} is it, but where does this value come from |
| 23:06.47 | manxpower | [TK]D-Fender: I think it defaults to ${EPOCH} as the time. |
| 23:07.00 | [TK]D-Fender | manxpower: No... he passed it. |
| 23:07.01 | manxpower | adnc: epoch comes from the system. |
| 23:07.05 | adnc | i see, i should have asked more precize |
| 23:07.05 | [TK]D-Fender | ^^ |
| 23:07.19 | Corydon76-dig | EPOCH comes from all around you. |
| 23:07.29 | Kobaz | haha |
| 23:07.34 | manxpower | EPOCH has no tomezones. |
| 23:07.39 | Corydon76-dig | You cannot see it, taste it, or smell it, but EPOCH is there all the same |
| 23:07.50 | manxpower | like god! |
| 23:07.50 | Kobaz | i can see epoch |
| 23:08.01 | Kobaz | 1268262478 |
| 23:08.01 | manxpower | wait, sorry. EPOCH exists. |
| 23:08.30 | Corydon76-dig | Kobaz: you only see a representation of a snapshot of EPOCH |
| 23:08.50 | Corydon76-dig | As soon as you store EPOCH, it is no longer the EPOCH |
| 23:08.58 | Qwell | How long did EPOCH last? |
| 23:09.00 | Kobaz | heh |
| 23:09.05 | Qwell | Was it a full second? A millisecond? |
| 23:09.25 | Corydon76-dig | Qwell: that depends on whether you're using 1.4 or 1.6 |
| 23:09.30 | Qwell | heh |
| 23:09.34 | manxpower | Qwell: 57 years, by mt rough off the top of my head calculations |
| 23:10.00 | Kobaz | i think we have 27 years left |
| 23:10.01 | Corydon76-dig | manxpower: 68 years, if you only count the positive part |
| 23:10.12 | Kobaz | it's signed right? |
| 23:10.22 | Corydon76-dig | Yes |
| 23:10.27 | Kobaz | silly coders |
| 23:10.41 | manxpower | I was using jan 1 1970 thru 2037, which is the date I vaguely recall it blowing up |
| 23:10.53 | Corydon76-dig | 2038 |
| 23:11.04 | Kobaz | could have doubled the span using unsigned |
| 23:11.19 | Corydon76-dig | Kobaz: then you couldn't have represented 1969 |
| 23:11.22 | hackeron | so wait, dahdi doesn't work on linux 2.6.31? |
| 23:11.26 | Kobaz | "they'll come up with a fix by then" |
| 23:11.31 | Kobaz | hmm |
| 23:11.34 | Kobaz | true |
| 23:11.51 | Corydon76-dig | and with 30 year mortgages... that was important |
| 23:11.55 | Kobaz | hehe |
| 23:12.30 | Corydon76-dig | (Note that banks had to solve this problem 2 years ago, for the same reason) |
| 23:12.42 | Kobaz | make time_t bigger? |
| 23:13.39 | Kobaz | although that would completely break anything that blindly uses int's instead of time_t |
| 23:13.59 | Kobaz | we need a time64_t |
| 23:14.12 | Corydon76-dig | No, we don't. time_t is fine |
| 23:15.04 | Corydon76-dig | Remember that int also grows on 64-bit platforms |
| 23:15.29 | Kobaz | but what if you're not on a 64bit platform |
| 23:15.45 | Corydon76-dig | Then you shouldn't be managing mortgages |
| 23:15.49 | Kobaz | heh |
| 23:16.02 | Qwell | Corydon76-dig: Banks won't be using 64-bit systems for another 20 years or so. |
| 23:16.06 | Qwell | at least |
| 23:16.18 | Kobaz | what if it's running on an 8 bit machine with 4 bit registers written by a 2 bit programmer |
| 23:16.29 | Corydon76-dig | Qwell: UltraSPARC is already 64-bit |
| 23:16.32 | drmessano | I cant wait for banks to upgrade to 32 bit |
| 23:16.45 | [TK]D-Fender | Kobaz: for a company that can't stand 1 bit of competition... |
| 23:16.52 | Kobaz | hehe |
| 23:18.07 | *** join/#asterisk bjhaid (~herbayjha@41.206.15.1) |
| 23:18.25 | Kobaz | the problem with switching to a 64bit time, is in the year 584942415385 they'll run into this same problem all over again |
| 23:18.28 | Kobaz | and have to move to 128bit |
| 23:20.15 | leifmadsen | lol |
| 23:20.27 | drmessano | If you don't take the Y5B problem into account |
| 23:20.40 | Nugget | lol |
| 23:20.58 | drmessano | The sun has enough energy to only last another 5 billion years. We'll be toast by then.... cold toast. |
| 23:21.20 | drmessano | Nobody likes cold toast. |
| 23:21.27 | Kobaz | nope |
| 23:22.02 | leifmadsen | I do! |
| 23:22.09 | drmessano | Canuck |
| 23:22.09 | leifmadsen | it put it on top of my cold pizza |
| 23:22.09 | Corydon76-dig | If I'm still around when that problem occurs, you're more than welcome to call me up to fix it |
| 23:22.19 | leifmadsen | Corydon76-dig: I'll hold you to that |
| 23:22.36 | drmessano | Corydon76-dig: I'll submit a bug report a week prior |
| 23:22.45 | leifmadsen | ok, reboot time because my alt+tab combo seems to have stopped working.... *groan* |
| 23:23.00 | drmessano | He should have restarted the alttab daemon |
| 23:23.01 | drmessano | Duh |
| 23:23.17 | drmessano | Linux: There's a daemon for that |
| 23:23.45 | Qwell | I'd like to know what console-polkit-daemon is, and why it likes taking 100% CPU |
| 23:23.50 | Kobaz | haha |
| 23:23.52 | Kobaz | just kill it |
| 23:23.55 | Qwell | I do |
| 23:24.03 | Qwell | and you know what happens? |
| 23:24.04 | drmessano | cron job |
| 23:24.05 | Qwell | NOTHING |
| 23:24.09 | Kobaz | it's some sort of auditing tool |
| 23:24.15 | Corydon76-dig | Qwell: it plays polka music? |
| 23:24.17 | Kobaz | you dont need it |
| 23:24.27 | Kobaz | i always have to kill/disable mine |
| 23:24.30 | Kobaz | i just delete the binary |
| 23:24.33 | drmessano | kill -somereallybadswitch ? |
| 23:24.34 | Deeewayne | Corydon76-dig, polkit music |
| 23:24.46 | Kobaz | it spawns, and all it does is suck up 100% cpu and half a gig of memory |
| 23:24.46 | Qwell | Kobaz: heh, good plan |
| 23:24.50 | Qwell | yep |
| 23:24.58 | Kobaz | Qwell: i have yet to find the startup rc script where it's spawned |
| 23:25.01 | Qwell | there's another one that acts stupid sometimes too.. |
| 23:25.03 | drmessano | polkit music to the console.. Shit, someone will want to use that with originate in Asterisk, trust me |
| 23:25.04 | Qwell | some gnome thing |
| 23:25.22 | Qwell | something to do with auth |
| 23:25.55 | Corydon76-dig | Qwell: A gnome buttplug? http://www.exotic-erotics.com/store/images/products/gnome-new.jpg semi-SFW |
| 23:26.13 | Deeewayne | lol |
| 23:26.19 | Qwell | polkit-gnome-authentication-agent-1 |
| 23:26.42 | drmessano | Safe for Liberal workplaces or barely existent govt jobs? |
| 23:26.57 | drmessano | Oh nice |
| 23:27.08 | drmessano | One gnome you don't want to lose |
| 23:27.14 | Corydon76-dig | drmessano: it's not explicitly NSFW, except when you consider what it's for |
| 23:27.37 | Deeewayne | he looks happy |
| 23:27.44 | Kobaz | Corydon76-dig: that's a nice gnome |
| 23:27.54 | Qwell | Kobaz: man polkitd |
| 23:28.06 | drmessano | Corydon76-dig: Sorta like the ink blot test "Pervert? You're the one showing me all this porn!!" |
| 23:28.22 | Kobaz | No manual entry for polkitd |
| 23:28.34 | Kobaz | # man female |
| 23:28.35 | Kobaz | No manual entry for female |
| 23:28.43 | Qwell | polkitd provides the org.freedesktop.PolicyKit1 D-Bus service on the system message bus. Users or administrators should never need to start this daemon as it will be automatically started by dbus-daemon(1) whenever an application calls into the service. |
| 23:29.21 | drmessano | I'd file that as an Asterisk bug, just sayin |
| 23:31.25 | Katty | hi |
| 23:31.48 | drmessano | "My system regularly hangs with 100% CPU usage. I am also using Asterisk. I have attached a copy of my sip.conf" |
| 23:32.07 | Kobaz | haha |
| 23:32.14 | Katty | funny. |
| 23:32.34 | jaytee | hehe |
| 23:32.38 | Katty | oh |
| 23:32.40 | Katty | hi there mister tee |
| 23:32.58 | Katty | any news? (= |
| 23:33.28 | manxpower | funny, but Asterisk recently had a 100% CPU usage issue |
| 23:35.06 | drmessano | 1.6 will/would get DoS'ed by a slightly older version of Flash Operator Panel |
| 23:35.26 | drmessano | That was a cute one |
| 23:36.30 | drmessano | I think that's the only 100% CPU issue I have ever had with Asterisk, and it wasn't really an Asterisk issue |
| 23:37.02 | Kobaz | i've had several 100% cpu issues with asterisk in 1.4 |
| 23:37.39 | drmessano | will now replace his opinion on 1.4 with "1.6.x releases are nice" |
| 23:37.49 | drmessano | 1.6.x releases are nice |
| 23:37.55 | Katty | can't say i've ever had one |
| 23:38.28 | Kobaz | 1.6 has been pretty good |
| 23:38.36 | Kobaz | a few regression here and there, but my testing picks them up |
| 23:38.46 | drmessano | 1.6.x has been light years better than 1.4 |
| 23:39.01 | Kobaz | at least it finds the regressions i check for |
| 23:40.49 | hluesea | Hii |
| 23:41.01 | Katty | asl |
| 23:41.08 | drmessano | HAHAH |
| 23:41.47 | hluesea | :D |
| 23:42.00 | drmessano | m/f? |
| 23:42.02 | hluesea | m |
| 23:42.08 | hluesea | 24 m |
| 23:42.09 | hluesea | :P |
| 23:42.17 | manxpower | Katty: Older than you, as often as possible, cyberspace |
| 23:42.17 | drmessano | *click* |
| 23:42.25 | hluesea | it is very nice irc is the old irc simple :) |
| 23:42.55 | jaytee | older than dirt, Clinton was President, in exile in Indiana |
| 23:43.08 | Katty | :< |
| 23:43.11 | Kobaz | that's not old |
| 23:43.12 | Katty | i still love you. |
| 23:43.15 | Katty | hugs jaytee |
| 23:43.18 | jaytee | aww thanks |
| 23:43.22 | jaytee | hugs Katty |
| 23:43.36 | Kobaz | anyone remember regan? |
| 23:43.37 | manxpower | jaytee: must be married, eh? |
| 23:43.46 | jaytee | nope |
| 23:43.51 | Kobaz | me neither |
| 23:43.57 | drmessano | Middle Paleolithic, Yes, Waiting to acquire signal |
| 23:44.39 | drmessano | Reagan or Regan? |
| 23:44.41 | Katty | well i need to go run soon. before it startsg etting dark out |
| 23:44.47 | jaytee | I remember Regan, Donald Regan? he was Sec of Treasury under REAGAN |
| 23:44.55 | drmessano | Indeed |
| 23:44.57 | Kobaz | heh |
| 23:45.14 | Katty | cheerio (= |
| 23:45.20 | drmessano | trix |
| 23:46.09 | [TK]D-Fender | Katty: No... clearly Froot Loops |
| 23:47.30 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:50.46 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:51.22 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:51.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
