| 00:02.32 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:03.42 | n00dle | Ciao! |
| 00:05.45 | toerkeium | JT: dIDN'T imagine it. Should I install FoIP within asterisk? |
| 00:09.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:09.49 | *** part/#asterisk galeras (n=root@200.31.204.42) |
| 00:11.07 | JT | toerkeium: asterisk does not support T.38 endpoint, only passthrough in 1.4 |
| 00:11.19 | JT | toerkeium: it's something you really should have researched and checked first |
| 00:11.37 | JT | toerkeium: voip codecs are designed to move voice information, not modem signals |
| 00:13.51 | toerkeium | I did never think about a fax signal bein different than the channel where the voice goes (if you understand what I ment :) |
| 00:14.27 | toerkeium | I just think about it like different thing, going over the same channel |
| 00:14.55 | toerkeium | anyway.. I was worried about my * install.. I can rest now :) |
| 00:15.13 | JT | toerkeium: what codec were you using to your voip provider? |
| 00:15.45 | toerkeium | licensed g729 |
| 00:15.51 | toerkeium | from digium |
| 00:16.05 | toerkeium | 10 bucks to the trash! :;) |
| 00:16.15 | JT | yeah absolutely no way it will work for fax |
| 00:16.22 | JT | it's good for compressing voice |
| 00:16.33 | toerkeium | yeah, voice sounds just perfect |
| 00:16.51 | toerkeium | had some problems, but nothing that asterisk updated cound't fix |
| 00:17.06 | toerkeium | updates* |
| 00:17.48 | toerkeium | now I am trying to connect 2 endpoints using vpn with ipsec to speak h323 |
| 00:18.05 | toerkeium | strange behaviors |
| 00:18.32 | toerkeium | I can hear the person who calls, but this person can't hear me |
| 00:18.42 | toerkeium | I guess it has to be something related to rtp |
| 00:18.48 | JT | h.323 is pretty flakey in asterisk |
| 00:19.00 | toerkeium | but well.. reading now what the hell stunt and and TURN |
| 00:19.18 | toerkeium | not using * this way, just some cisco devices |
| 00:20.01 | toerkeium | do you think that behavior is related to rtp ? |
| 00:20.13 | *** join/#asterisk wotcha (n=jim@cust4716.qld01.aanet.com.au) |
| 00:20.14 | *** join/#asterisk nickmannick (n=Ownerdsa@S0106001346face5f.ed.shawcable.net) |
| 00:21.13 | nickmannick | Does anybody have a working sample of a extension were it ask the user to press one and then if they do then it will go to the next thing or do something if not then it will hangup |
| 00:22.05 | toerkeium | I don't know how to do it nickmannick, but I know it's pretty easy |
| 00:22.31 | nickmannick | somebody that does know would be helpfull |
| 00:24.21 | _charly_ | nickmannick: http://www.das-asterisk-buch.de/stable/einfache-ivr.html and http://www.das-asterisk-buch.de/stable/mehrstufige-ivr.html |
| 00:24.24 | toerkeium | search for asterisk commands |
| 00:25.05 | _charly_ | nickmannick: sorry, it's in german, but i think the dialplan is easy to understand |
| 00:25.54 | nickmannick | i'll give it a shot |
| 00:26.46 | toerkeium | it only needs exten => s,4,ResponseTimeout,10 ; Response Timeout to 10 seconds |
| 00:26.48 | toerkeium | I guess |
| 00:27.10 | toerkeium | to setup a hangup timeout |
| 00:31.50 | *** join/#asterisk saftsack (n=saftsack@pD9E055AA.dip.t-dialin.net) |
| 00:32.23 | *** join/#asterisk saftsack (n=saftsack@pD9E055AA.dip.t-dialin.net) |
| 00:34.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:35.17 | boch | anyone using Asterisk::Manager ? |
| 00:36.18 | toerkeium | I used the API to access the asterisk manager, I guess it's called like that, or is that a perl module? |
| 00:36.56 | boch | a perl module, but docs are weak |
| 00:37.46 | toerkeium | use antoher languague, like php, it's pretty well documented, at least for basic things I guess |
| 00:38.00 | boch | is there a class or anything ? |
| 00:38.11 | toerkeium | there is a lot of information in php |
| 00:40.01 | Juggie | http://eder.us/projects/phpagi/ |
| 00:40.30 | toerkeium | http://www.voip-info.org/wiki-Asterisk+manager+API |
| 00:40.44 | boch | Juggie, thats for AGI not Manager |
| 00:41.07 | JT | eww php |
| 00:41.14 | Juggie | i know i'm old and all, being 26 |
| 00:41.20 | Juggie | but i dont think my eyes are going yet |
| 00:41.24 | Juggie | look at the link carefully |
| 00:41.42 | Juggie | AGI_AsteriskManager |
| 00:41.43 | Juggie | Description |
| 00:41.43 | Juggie | Description | Vars (details) | Methods (details) |
| 00:41.43 | Juggie | Asterisk Manager class |
| 00:42.30 | boch | :$ sorry, my mistake |
| 00:42.47 | toerkeium | explain the meaning if "eww" JT :) |
| 00:44.05 | JT | php is a yucky programming language, hence the "eww" |
| 00:44.59 | toerkeium | lol |
| 00:45.15 | toerkeium | what would be your programming languague choise? |
| 00:45.26 | Nugget | INTERCAL |
| 00:45.43 | Nugget | even it is more appropriate than php for general programming. |
| 00:46.14 | JT | for scripting, python or perl |
| 00:47.09 | toerkeium | never heard of it "intercal" |
| 00:48.24 | JT | php has let every newbie with a text editor release awful code |
| 00:49.03 | Nugget | php originally stood for "personal home page" |
| 00:49.15 | Nugget | it was, and still is to a large degree, nothing more than an uppity web markup language. |
| 00:49.33 | Nugget | people who mistake it for a general-purpose programming language are just aiming the shotgun at their foot |
| 00:49.36 | toerkeium | JT, as all newbie codes! :) |
| 00:50.02 | JT | toerkeium: some languages are much more crap than others :) |
| 00:50.27 | Nugget | http://en.wikipedia.org/wiki/Intercal > PHP |
| 00:50.31 | *** join/#asterisk sandorp (n=sandor@firewall2.wsi.net) |
| 00:50.33 | toerkeium | yeah, but a newbie will manage it to make awful code anyway (like me, for example) :) |
| 00:50.54 | sandorp | is there a boot (rc) script in the asterisk 1.2 source tree somewhere? |
| 00:51.46 | toerkeium | from wikipedia: INTERCAL es un lenguaje de programación esotérico diseñado para ser extremadamente difícil de entender |
| 00:51.56 | toerkeium | it's not translated in the ENglish version |
| 00:52.30 | toerkeium | means: INTERCAL is a programming language designed to be extremely difficult to understand (or similiar) |
| 00:53.05 | toerkeium | it's fun the part "it was designed to be" heh |
| 00:53.19 | s0ck | anyone running asterisk on the newer core2duo based xeons? |
| 00:53.45 | Nugget | I run it on a mac pro, but not in production. I think that's the kind of xeon you mean. |
| 00:54.00 | JT | s0ck: i don't think they're called "core2duo" for xeons |
| 00:54.06 | s0ck | they aint |
| 00:54.09 | s0ck | but you know what i mean :D |
| 00:54.20 | JT | toerkeium: no, seriously, some programming languages really are much worse |
| 00:54.23 | JT | php is awful |
| 00:54.30 | s0ck | i've been running * on an old dual p3 |
| 00:54.35 | JT | i don't know many experienced *nix users who like it |
| 00:54.36 | s0ck | about to put it on a 'proper' box |
| 00:54.45 | s0ck | and i bet im gonna have issues with the raid card/chipsets etc |
| 00:54.56 | s0ck | just wondering if anyone else already running something like that |
| 00:55.32 | toerkeium | JT, I don't say you are wrong, I cound't, I just have basic skills of programmings, if I could say that.. |
| 00:55.50 | JT | toerkeium: python is quite easy to learn imho and far cleaner |
| 00:56.05 | s0ck | is python similar to perl |
| 00:56.13 | JT | s0ck: depends what the proper box is... |
| 00:56.17 | JT | s0ck: not really |
| 00:56.18 | Nugget | python is more structured than perl |
| 00:56.23 | JT | phython is much more OO |
| 00:56.29 | JT | and uses whitespace for formatting |
| 00:57.00 | toerkeium | JT, but you have to admit, that php grows fast and have impressive improvements |
| 00:57.12 | s0ck | xeon 3060/15k sas raid1 on a serveraid 8k card prolly |
| 00:57.35 | JT | toerkeium: you mean it changes massively with every release and code you wrote for one release doesn't work on the next? yes |
| 00:57.44 | s0ck | lol |
| 00:57.45 | toerkeium | lol |
| 00:57.47 | Nugget | I've heard good things about those serveraid cards, but I'm still buying 3ware for my garage-build machines. |
| 00:57.50 | JT | s0ck: just check driver availability first i guess |
| 00:57.52 | s0ck | that's so true loll |
| 00:58.03 | JT | s0ck: is that ibm? if so, i've had luck |
| 00:58.05 | JT | on an X260 |
| 00:58.06 | s0ck | JT: it purports to support linux to im gonna have to wait and see |
| 00:58.08 | JT | X360 |
| 00:58.15 | s0ck | it is indeed an ibm rackmount :D |
| 00:58.21 | JT | love ibm |
| 00:58.29 | toerkeium | JT: you're not wrong with that |
| 00:58.31 | s0ck | tis all we do really |
| 00:58.37 | s0ck | you uk/us? |
| 00:58.46 | JT | .au |
| 00:58.51 | s0ck | ah right |
| 00:59.06 | JT | s0ck: ibm are massively behind linux, so yeah |
| 00:59.15 | s0ck | cool |
| 00:59.21 | JT | dell still make the cheapest brand name servers though :/ |
| 00:59.32 | s0ck | i have a feeling serveguide runs in some kind of linux so it would make sense |
| 00:59.58 | s0ck | quite happy with the ibms |
| 01:00.01 | s0ck | customers like them |
| 01:00.07 | s0ck | 3 year 4 hour response |
| 01:00.12 | s0ck | stick it in and forget about the hardware |
| 01:00.40 | JT | yeah, the build quality is quite nice |
| 01:00.57 | s0ck | had a nightmare trying to get a vxa320 drive to work though ;/ |
| 01:01.01 | voipnet-tech | $350 -> 1.9Ghz AMD X2 64 (Dual Core) 1Mb Cache, 2Gb DDR2 800Mhz Dual Channel SDRAM, 200Gb SATA II HD, 24x DVD-RAM w/ LightScribe, KB/Mouse |
| 01:01.17 | JT | voipnet-tech: is that an ad? |
| 01:01.36 | voipnet-tech | i just built two of these buying parts |
| 01:01.39 | voipnet-tech | no |
| 01:01.50 | JT | okay :) |
| 01:01.52 | voipnet-tech | these scream... would work great as a server too |
| 01:02.07 | voipnet-tech | $350 for a server is cheap too |
| 01:02.12 | JT | yeah, buying parts is so much cheaper than brand names for servers |
| 01:02.14 | s0ck | the core2s are miles ahead in terms of performance tho |
| 01:02.20 | JT | unfortunately you don't get the warranty etc |
| 01:02.25 | JT | and the refinement of a brand name |
| 01:03.07 | voipnet-tech | if you're using linux... why get a brand name PC if you aren't running a brand-name OS? |
| 01:03.12 | voipnet-tech | lol |
| 01:03.14 | toerkeium | well.. I have opened some brand name servers and found that have generic hardware, like some compaq servers |
| 01:04.14 | s0ck | indeed |
| 01:04.27 | s0ck | freecome dat72 tapedrive on the market = 200 quid |
| 01:04.43 | s0ck | EXACT same drive sold by ibm = 450 |
| 01:04.43 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:04.44 | s0ck | freecom fs |
| 01:04.50 | s0ck | free come lol |
| 01:04.57 | nickmannick | i am trying to use the mysql in the extension and getting this error No application 'MYSQL' for extension |
| 01:05.04 | nickmannick | I did install the asterisk-addons |
| 01:05.36 | toerkeium | see? I'll never buy a brand name server ... ... ... again |
| 01:05.46 | JT | voipnet-tech: are you joking? have you ever owned a brand name server? |
| 01:06.59 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 01:08.41 | toerkeium | nickmannick: that means that there is no application called MYSQL |
| 01:09.11 | nickmannick | so how do i get the mysql stuff working all i want to do is query a table |
| 01:09.15 | nickmannick | for my applicatiopn |
| 01:09.38 | nickmannick | exten => _X.,1,MYSQL(Connect connid localhost dbuser dbpass dbname) |
| 01:09.55 | Hmmhesays | there is an example on the wiki |
| 01:10.13 | nickmannick | thats what i am reading |
| 01:10.16 | s0ck | voipnet-tech: didn't see the comment above |
| 01:10.23 | Hmmhesays | connect get your info, disconnect |
| 01:10.29 | Hmmhesays | so you don't have any zombie mysql connections |
| 01:10.32 | s0ck | regardless of the os, when you build a server you want it to work, forever, hassle free, right? |
| 01:11.16 | nickmannick | well i get mysql application etc |
| 01:11.18 | s0ck | linux isn't a second grade os :P |
| 01:13.04 | voipnet-tech | s0ck, actually in my experience... there's no such thing as a hassle free server that works forever, so i'd rather have one that's easier to fix when it does break. So I'd rather have one with standard parts. also OS is important because it needs to be highly compatible with the hardware to work properly... doesn't matter what kind of hardware you've got... if the software doesn't work properly you're screwed anyway |
| 01:14.46 | toerkeium | I have about 45 servers running, for various purposes, and what I keep always in mind when buying a server is the chipset, same brand, always.. that will let you change harward without screwing your system |
| 01:15.16 | toerkeium | no matter what motherboard brand I buy, I make sure its chipset is intel |
| 01:15.29 | nickmannick | So what is it that i am missing to be available to use mysql |
| 01:16.15 | JT | voipnet-tech: i take it you have very little corporate/enterprise IT experience then |
| 01:16.31 | JT | voipnet-tech: brand name servers have these things called warranties |
| 01:16.43 | JT | you don't need to futz about finding a clone part |
| 01:17.15 | toerkeium | JT, in my case, I could say I have not enterprise experience, since I run my own business, and it's pretty small |
| 01:17.49 | JT | toerkeium: it's something you just have to weigh up, i guess |
| 01:17.52 | toerkeium | but all we have different situations, and enterprise hardware or equipments is only available for enterprises, not for everyone |
| 01:18.06 | JT | but i often prefer to use out of warranty second hand brand name servers to clone servers |
| 01:18.17 | JT | toerkeium: brand name servers can be bought by anyone |
| 01:18.38 | toerkeium | but are known to be very expensive, compared to clone machines |
| 01:19.11 | JT | the level of refinement is massively different |
| 01:19.22 | JT | a lot of stuff cannot be hotswapped in a clone |
| 01:19.26 | JT | and a lot of things need use of scredrivers |
| 01:19.26 | toerkeium | but the ones which are cheap, have mostly generic hardware |
| 01:19.26 | JT | screwdrivers |
| 01:19.28 | JT | yeah, they're a waste of time |
| 01:19.28 | Qwell | ~cheap |
| 01:19.29 | jbot | somebody said cheap was when microsoft designs softhardware, or nasty |
| 01:19.30 | JT | usually |
| 01:20.14 | JT | also, clone power supplies are not as powerful or redundant often, and there are less LOM options |
| 01:20.47 | JT | that said, clones are sometimes the right choice, depends on the situation |
| 01:21.11 | toerkeium | yeah, thats true |
| 01:21.31 | toerkeium | my customers are not enterprise .. probably tht's why I don't need to buy enterprise solutions |
| 01:22.00 | JT | i treat everything like an enterprise |
| 01:22.03 | Qwell | JT: Clone servers - for when downtime IS an option. |
| 01:22.17 | JT | my home network runs on mostly enterprise grade hardware |
| 01:22.28 | toerkeium | hehehe |
| 01:22.38 | JT | switch has hotswap modules and redundant power supply, redundant power supplies on servers |
| 01:23.04 | br4k3r | is it that important really? |
| 01:23.04 | Qwell | Your...SWITCH...is hotswappable? |
| 01:23.06 | toerkeium | <JT> i treat everything like an enterprise > what do you mean by that? |
| 01:23.30 | br4k3r | i think he means THE enterprise |
| 01:23.35 | JT | Qwell: yes, all the 8 port 10/100 port modules can be hotswapped, and the power supply, only the supervisory card cannot be hotswap |
| 01:23.41 | Qwell | heh |
| 01:23.44 | JT | Hp ProCurve |
| 01:23.45 | Qwell | excessive |
| 01:23.46 | [TK]D-Fender | qwell : Yes, the electrons flow freely through the air while you seach for your backup ;) |
| 01:23.48 | Qwell | must've been cheap |
| 01:23.58 | JT | yeah, it was, i love auctions |
| 01:24.02 | Qwell | heh |
| 01:24.20 | ltdwk | HP provide 4000M |
| 01:24.21 | JT | 48 port switch, got it for $400 about 6 years ago when they were still worth $4k |
| 01:24.23 | ltdwk | procurve even |
| 01:24.28 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 01:24.40 | JT | AUD that is |
| 01:24.42 | *** join/#asterisk rollinnco (n=dj@c-71-57-138-68.hsd1.fl.comcast.net) |
| 01:24.49 | ltdwk | i use hp gear... it really is great value for money |
| 01:24.57 | JT | hp procurve stuff is worth it |
| 01:25.01 | JT | lifetime warranty |
| 01:25.08 | JT | on almost everything |
| 01:25.22 | JT | except a couple of models of enterprise firewalls or something |
| 01:25.34 | Iamnacho | awsome product for the money. and cant beat the lifetime warrenty |
| 01:25.40 | ltdwk | exactly |
| 01:25.45 | ltdwk | i run it in the core and have never had any issues |
| 01:25.58 | JT | toerkeium: their range of networking gear |
| 01:26.14 | ltdwk | one got destroyed by a lightning strike but that's all |
| 01:26.24 | JT | wish they had bigger PoE switches though, think the biggest is 48 port |
| 01:26.32 | JT | ltdwk: did you claim warranty? |
| 01:26.40 | toerkeium | oh oh |
| 01:26.46 | ltdwk | ltdwk: yep, claimed it under insurance and got warranty so ended up with two new ones |
| 01:26.55 | JT | dodge :P |
| 01:27.05 | ltdwk | no, smart :P |
| 01:27.09 | JT | heh |
| 01:27.26 | JT | did they need much paperwork to process warranty? |
| 01:27.50 | ltdwk | can't remember, too long ago now |
| 01:27.50 | Hmmhesays | fraud! |
| 01:27.56 | toerkeium | yeah |
| 01:28.07 | toerkeium | this conversation is taking a non-legal subject |
| 01:28.20 | toerkeium | I am going to log everything! |
| 01:28.27 | ltdwk | you do that :P |
| 01:28.40 | toerkeium | lol |
| 01:28.55 | br4k3r | jeez u can heat ur house on that switch d00d |
| 01:29.00 | br4k3r | 535 btu/h |
| 01:29.06 | br4k3r | on the procurve 4000m |
| 01:29.17 | JT | but yeah, if money was not the primary concern, i would only buy IBM and Sun servers |
| 01:29.25 | br4k3r | hehehe |
| 01:29.25 | JT | br4k3r: no, it can't |
| 01:29.27 | JT | seriously |
| 01:29.31 | JT | i wish it was warmer |
| 01:29.36 | JT | it's freezing at the moment |
| 01:29.39 | br4k3r | u need some supersparc |
| 01:29.42 | JT | and it did not warm enough |
| 01:29.54 | JT | sparcs are risc, don't make that much heat |
| 01:30.00 | br4k3r | hehehe |
| 01:30.01 | br4k3r | mine do |
| 01:30.07 | rollinnco | I am not trying to hijack the subject, however, I was wondering if anybody has had a problem passing through g729 calls from provider to provider using asterisk 1.2.17 |
| 01:30.07 | JT | i could get the Sun Enterprise 4000 out of the garage though |
| 01:30.20 | Iamnacho | the 5400zl will do more than 48 ports w/ PoE |
| 01:30.21 | br4k3r | nice |
| 01:30.29 | br4k3r | i'd strangle someone for an e4k |
| 01:30.41 | JT | Iamnacho: is that the only model, and is that only a poe + gigabit option? |
| 01:30.44 | br4k3r | nice big scsi controller |
| 01:31.40 | JT | br4k3r: they're easy to get on ebay |
| 01:31.40 | br4k3r | i just got a new e480 |
| 01:31.40 | Iamnacho | JT: it is. but its a modular switch like the 4000 |
| 01:31.40 | br4k3r | sunfire |
| 01:31.40 | toerkeium | wahh ebay sucks |
| 01:31.40 | br4k3r | i'm setting it up to werk with a couple of my blades |
| 01:31.50 | br4k3r | got about 9 sunblade attachments around here |
| 01:31.55 | br4k3r | wanna put them all on pbx :) |
| 01:31.57 | toerkeium | I buy a keyboard for my old and lovely compaq M700 and never came to my hands |
| 01:32.16 | br4k3r | :( |
| 01:32.18 | br4k3r | that sucks |
| 01:32.24 | JT | toerkeium: i'd say my transaction success rate on ebay is about 98% |
| 01:32.47 | br4k3r | i've never had an issue, buying or selling, since 03 |
| 01:32.52 | br4k3r | or 04 |
| 01:32.53 | JT | br4k3r: i have a sun ray 150 sitting around doing nothing too :P |
| 01:32.59 | br4k3r | hehe |
| 01:33.04 | br4k3r | they're fun little rigs |
| 01:33.08 | Qwell | I have a T2000 sitting around doing nothing :D |
| 01:33.11 | br4k3r | check out the sun cobalt servers |
| 01:33.18 | JT | Iamnacho: sounds excessive, i only want 100Mbits on PoE ports |
| 01:33.19 | br4k3r | t2000? |
| 01:33.22 | toerkeium | well, I cound't trust them anymore, even the keyboard cost me a few 30 bucks, how to make sure¡? I cound't even ask for a charge back |
| 01:33.26 | JT | br4k3r: T2000 is awesome |
| 01:33.27 | Qwell | I was trying to get zaptel working on it, in linux |
| 01:33.29 | JT | sun coolthreads |
| 01:33.32 | Iamnacho | :) |
| 01:33.43 | br4k3r | what brand is t20000 |
| 01:33.44 | Qwell | failed though, due to user land/kernel space bitedness differences |
| 01:33.45 | *** join/#asterisk froguz (n=froguz@pc-6-103-104-200.cm.vtr.net) |
| 01:33.47 | Qwell | br4k3r: Sun |
| 01:33.47 | JT | toerkeium: why wouldn't you ask for a chargeback? |
| 01:33.53 | br4k3r | sun |
| 01:33.54 | br4k3r | ok |
| 01:33.56 | br4k3r | :) |
| 01:34.06 | JT | Iamnacho: unfortunately it is a financial issue that may exclude the procurve from selection |
| 01:34.12 | br4k3r | oh this is the really new stuff |
| 01:34.21 | toerkeium | JT: I did, but passing 30 days the wash their hands, something to keep in mind |
| 01:34.22 | br4k3r | i haven't seen any of this yet |
| 01:34.29 | toerkeium | the = they |
| 01:34.30 | *** join/#asterisk V3rM3 (n=verme@201.79.169.244) |
| 01:34.30 | JT | toerkeium: credit card? |
| 01:34.30 | Iamnacho | jt: true. they are expensive. but well priced. |
| 01:34.33 | br4k3r | the newest we have at werk is the sunfire 490 |
| 01:34.37 | rollinnco | anybody? voice quality problems passing calls from provider to provider using asterisk 1.2.17? |
| 01:34.39 | *** part/#asterisk V3rM3 (n=verme@201.79.169.244) |
| 01:34.45 | JT | br4k3r: multihread computing, it's great |
| 01:34.47 | *** join/#asterisk tomcontr3 (n=tomcontr@51-79-246-201.adsl.terra.cl) |
| 01:34.59 | toerkeium | JT: they ask for the bill, otherwise they nothing can do |
| 01:35.00 | tomcontr3 | does any one knows a good TTS system for Asterisk |
| 01:35.02 | JT | Iamnacho: but i don't WANT to pay for gigabit on PoE ports, imho that's stupid |
| 01:35.13 | JT | toerkeium: credit card or not? |
| 01:35.22 | toerkeium | yes, I paid by credit card |
| 01:35.30 | froguz | is it possible to call from a PRI, though an asterisk connected to a GSM gateway, sending the PRI telephone number (caller ID) instead of the SIM card number? |
| 01:35.34 | toerkeium | credit card is who ask for the bill |
| 01:35.38 | JT | usually you can reverse payment on credit cards for 90 days |
| 01:35.56 | JT | froguz: no. |
| 01:36.05 | toerkeium | not here, apparently :/ |
| 01:36.21 | JT | toerkeium: that's silly |
| 01:36.28 | br4k3r | The Sun Fire T2000 Server supports Ubuntu 6.10 (Edgy Eft) |
| 01:36.30 | JT | get a new CC |
| 01:36.30 | br4k3r | :):) |
| 01:36.37 | toerkeium | yeap |
| 01:36.55 | Qwell | eh, mine runs Gentoo |
| 01:37.00 | froguz | JT, not even using libss7? |
| 01:37.01 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 01:37.17 | Qwell | MAKEOPTS="-j64" |
| 01:37.23 | JT | froguz: gsm doesn't signal ssh over the air interface |
| 01:37.25 | Qwell | You have no idea just how freaking awesome that is |
| 01:37.28 | JT | s/ssh/ss7/ |
| 01:39.16 | *** join/#asterisk dlynes (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 01:42.41 | froguz | JT, thank you. so, no possibility at all =( |
| 01:42.41 | JT | froguz: i hope you weren't that serious about it |
| 01:42.41 | JT | the only reason people usually use gsm gateways is to save money |
| 01:42.41 | JT | and hope their customers and the telcos don't notice the use of gateways |
| 01:42.41 | JT | they're not what telcos use for interconnect |
| 01:44.18 | froguz | what they use? |
| 01:45.11 | JT | T1, E1, DS-3, E3, and up, and seperate SS7 serial links |
| 01:45.21 | JT | the standard depends on the country mainly |
| 01:46.15 | JT | it would be dumb for telcos interconnecting wasting radio spectrum with gsm, with poor audio quality, instead of using cabled links |
| 01:49.16 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 01:49.17 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 01:49.26 | froguz | there's a "small" telco, here in Chile, wich is having problems sending caller id to cell phones. i've connected an asterisk (with GSM gateway) to one of this telco's E1 trunk to test for a solution |
| 01:50.05 | JT | i see |
| 01:50.27 | toerkeium | froguz, bring me a laptop from chile, and I will buy you some beers! |
| 01:51.08 | froguz | they are urged to find a solution to avoid paging fines |
| 01:51.53 | froguz | toerkeium, a laptop? why? are laptops cheapers here than some place? |
| 01:52.15 | toerkeium | yeah, cheaper than in argentine |
| 01:52.31 | toerkeium | is it true you hate argentinians? be honest! |
| 01:52.46 | froguz | i can bring you very good an cheap wines, maybe some coper cathods, but laptop? |
| 01:53.10 | froguz | hahahahaha i personally like argentinian people, really |
| 01:53.26 | toerkeium | ok, wine is fine :) |
| 01:53.36 | JT | froguz: sounds like they need to employ someone who has a clue :) |
| 01:53.44 | froguz | specially argentinian musicians |
| 01:54.17 | toerkeium | ehh.. that's a nice subject to talk about! |
| 01:54.42 | toerkeium | I don't know why! |
| 01:54.44 | froguz | JT, you're right... but my boss insist in throwing me to that river |
| 01:55.40 | froguz | toerkeium, you 'stupid country' has the beauties girl i've ever seen |
| 01:55.46 | JT | froguz: it must be a really small telco |
| 01:56.00 | JT | you sure it's not just an ITSP or something? |
| 01:56.19 | toerkeium | hey yeah, I said stupid country, not stupid womans! and that's another very good subject to talk abou!" ;) |
| 01:56.40 | froguz | JT. 200.000 minutes a month (just mobile traffic) |
| 01:56.53 | JT | ok that's miiscule |
| 01:57.02 | JT | definitely doesn't sound like a tier 1 carrier |
| 01:57.19 | JT | does this "telco" own any infrastructure in the ground? |
| 01:57.21 | JT | like cables |
| 01:57.47 | killfill | i need an E1 card.. wich is best?.. digium, rhino or sangoma?.. |
| 01:57.50 | froguz | JT i'm pretty sure is not as little as you are imaging |
| 01:58.13 | JT | froguz: 200000 minutes is only ~4 Erlangs of traffic |
| 01:58.32 | froguz | the are a monopoly in a 7 millions region of the country |
| 01:58.33 | toerkeium | froguz, is it clandestine termination? |
| 01:58.38 | JT | i see |
| 01:58.53 | froguz | 7 million people |
| 01:58.58 | killfill | froguz: your from chile?... |
| 01:59.05 | JT | clearly they don't all use phones... to call mobiles |
| 01:59.09 | froguz | killfill, yes |
| 01:59.23 | JT | 200000 erlangs is ITSP/calling card provider quantities of traffic |
| 01:59.23 | killfill | froguz: hello mate!.. (me too :P |
| 01:59.28 | JT | hell, even callcentre |
| 01:59.40 | JT | err $ erlangs/200k minutes i mean |
| 01:59.45 | JT | 4 |
| 01:59.47 | JT | damnit :P |
| 01:59.52 | toerkeium | JT: not sure where you are, cell phone comunications are expensive |
| 01:59.55 | froguz | JT, maybe i'm wrong. surely is 200000 a day or week, i just heard that amount today |
| 02:00.01 | toerkeium | in latin america |
| 02:00.17 | JT | froguz: 200k min/mo is an average of 4 constantly utilised circuits |
| 02:00.45 | froguz | hi killfill, were are you? santiago? |
| 02:00.59 | killfill | exactly.. |
| 02:01.22 | froguz | JT, then is definetly not 200k a month |
| 02:02.11 | froguz | killfill maybe you can help me showing JT that Telefonica del Sur is not a so little telco |
| 02:02.24 | toerkeium | killfill: you can say hello to me too, we are at a USD120 distance ! |
| 02:02.46 | killfill | heh... |
| 02:03.15 | killfill | froguz: yeah.. well i dont know numbers.. :P |
| 02:03.41 | killfill | froguz: i use "TelefonicaCTC".. no problems with showing numbers |
| 02:03.51 | JT | froguz: the company should exmploy some telecommunications engineers :P |
| 02:03.59 | froguz | do you work at ctc? |
| 02:04.05 | JT | and i have NO IDEA why a legit big telco would use a gsm gateway for termination |
| 02:04.24 | killfill | froguz: thank god not.. :P |
| 02:04.28 | froguz | JT they don't... i'm just trying to find a solution |
| 02:04.44 | killfill | froguz: tour searching for a GSM gateway?.. |
| 02:04.51 | killfill | s/tour/your |
| 02:05.06 | JT | froguz: they must already have a network to terminate to gsm |
| 02:05.13 | froguz | killfill no. |
| 02:05.17 | tomcontr3 | hi killfill, are you from Chike? |
| 02:05.20 | tomcontr3 | CL |
| 02:06.09 | killfill | tomcontr3: yup |
| 02:06.16 | froguz | tomcontr3, you too? |
| 02:06.42 | tomcontr3 | yep |
| 02:07.17 | tomcontr3 | anyone of you uses any kind of TTS system? |
| 02:07.20 | froguz | JT, i can't understand why are they geting this mayor problem either |
| 02:07.58 | killfill | wow.. too much of us .. :P |
| 02:10.32 | blitzrage | ManxPower: zup! |
| 02:11.53 | *** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net) |
| 02:14.08 | toerkeium | I've never met people so ... so... so... than this freenode mantainers always telling you what they are going to do with that notices |
| 02:15.40 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:21.36 | toerkeium | someone knows how to explain in easy words what STUNT servers are and do? |
| 02:21.42 | toerkeium | can't understand very well from wikipedia |
| 02:22.32 | [TK]D-Fender | toerkeium, the jump through rings of FIRE and swallow swords WHOLE! |
| 02:22.38 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 02:24.15 | toerkeium | heh |
| 02:25.47 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 02:26.28 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 02:28.44 | *** join/#asterisk JunK-Y (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 02:31.09 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 02:34.30 | L|NUX | Hello every one |
| 02:34.35 | Hmmhesays | behringer is pissing me off |
| 02:34.40 | Hmmhesays | even more than L|NUX |
| 02:34.45 | L|NUX | can some one help me with getting call on IAX Client |
| 02:35.00 | *** part/#asterisk elg (n=fugalh@216.31.27.110) |
| 02:35.08 | *** join/#asterisk notoriousrab1982 (n=notoriou@76.195.14.206) |
| 02:35.46 | notoriousrab1982 | can anyone help me with a SIP dialling issue between * boxes? or at least give me thoughts on where to troubleshoot |
| 02:36.26 | Hmmhesays | what fantastically vague questions |
| 02:36.55 | L|NUX | May 31 21:36:14 NOTICE[30608]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 02:37.10 | L|NUX | getting this every time i am trying to get call on IAX Softphone |
| 02:38.21 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 02:39.02 | JT | make sure it's connected to the network? |
| 02:39.59 | JunK-Y | you trying to call an IAX2 softphone and using SIP... |
| 02:40.00 | notoriousrab1982 | im getting a similar problem - [May 31 19:39:12] WARNING[4472]: chan_sip.c:2738 create_addr: No such host: 192.168.2.52/b |
| 02:40.00 | notoriousrab1982 | [May 31 19:39:12] WARNING[4472]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 02:40.00 | notoriousrab1982 | <PROTECTED> |
| 02:40.23 | notoriousrab1982 | if you do sip show peers on both * machines, it shows they as online |
| 02:40.26 | JT | JunK-Y: rofl |
| 02:40.47 | JunK-Y | JT: ive to jet, i will let you help him :) |
| 02:40.56 | Hmmhesays | just because they are registered doesn't mean they are ready to accept a call |
| 02:40.59 | JT | JunK-Y: piker! |
| 02:41.28 | notoriousrab1982 | how can you test whether they are ready to accept a call? |
| 02:41.51 | *** join/#asterisk drcode (n=chatzill@87.69.35.234.cable.012.net.il) |
| 02:42.00 | Hmmhesays | No such host: 192.168.2.52/b <-- what is that 'b'? |
| 02:42.09 | JT | notoriousrab1982: clearly you have a typo in your config rile |
| 02:42.17 | drcode | hi all |
| 02:42.29 | notoriousrab1982 | it is an extension in the context which call goes into |
| 02:42.32 | drcode | dose Asterisks have video confrence? |
| 02:42.36 | drcode | like mcu? |
| 02:42.41 | JT | "Asterisk" |
| 02:42.54 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 02:43.17 | drcode | it has sip |
| 02:43.24 | drcode | is it support also in h323? |
| 02:43.31 | JT | not very well |
| 02:43.32 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 02:43.45 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 02:43.57 | drcode | k |
| 02:44.24 | drcode | is there some project that support h323 and sip like openmcu? |
| 02:44.54 | JT | i have no ide what openmcu is |
| 02:44.56 | JT | idea |
| 02:45.28 | drcode | k |
| 02:45.38 | L|NUX | JT : its connected |
| 02:45.51 | L|NUX | when i do iax2 debug its showing |
| 02:46.45 | JT | L|NUX: clearly you are doing it wrong, either choose IAX or SIP |
| 02:47.01 | L|NUX | JT : i have choose SIP |
| 02:47.09 | L|NUX | well see |
| 02:47.19 | L|NUX | i am registered on IAX Client |
| 02:47.26 | L|NUX | and when call comes on DID |
| 02:47.32 | L|NUX | it will give me this error |
| 02:47.49 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 02:48.05 | BSD_Tech | ok in 1.4 how to enable the jitterbuffer |
| 02:48.19 | BSD_Tech | I have almost 35ms of jitter |
| 02:48.24 | JT | L|NUX: you are not making sense |
| 02:48.29 | BSD_Tech | at points |
| 02:48.35 | JT | L|NUX: you must use SIP OR IAX to connect to the softphone |
| 02:48.39 | JT | not a combination of both |
| 02:48.52 | L|NUX | JT : i have choose IAX |
| 02:48.57 | L|NUX | JT : in softphone |
| 02:49.05 | JT | L|NUX: then why on earth are you dialling sip? |
| 02:49.08 | L|NUX | JT : but when i call DID it will not work |
| 02:49.15 | JT | because you are dialling sip |
| 02:49.16 | JT | ffs |
| 02:49.19 | JT | READ the error message |
| 02:49.39 | L|NUX | JT : so i should use Dial(IAX2/DID) ? |
| 02:49.42 | russellb | BSD_Tech: i have a blog post on asterisk.org that explains the how the 1.4 jitterbuffer works |
| 02:49.47 | blitzrage | russellb: !!! |
| 02:49.50 | JT | L|NUX: that would be simply logical |
| 02:49.59 | L|NUX | ok |
| 02:50.01 | L|NUX | brb |
| 02:50.07 | JT | ~thebook |
| 02:50.20 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:50.23 | JT | L|NUX: please read that ^ |
| 02:50.24 | [TK]D-Fender | JT : Screw that.... |
| 02:50.28 | [TK]D-Fender | ~osmosis |
| 02:50.36 | jbot | rumour has it, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 02:50.37 | [TK]D-Fender | ~ |
| 02:50.38 | JT | heh |
| 02:50.39 | [TK]D-Fender | !!! |
| 02:50.48 | L|NUX | JT : ok |
| 02:51.14 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 02:53.20 | L|NUX | JT: thanks |
| 02:53.26 | JT | np |
| 02:54.59 | JT | a thanks should suffice :P |
| 02:57.31 | killfill | how do you guys implement reduncandy?.. i.e. i have 1 E1 line, and want ot have 2 asterisk server. So one can take over the line when the other fails |
| 02:57.58 | JT | you will need an L1 PRI failover hardware box, and each server will need a PRI card |
| 02:58.57 | killfill | hmm failover hardware box.. |
| 02:59.05 | [TK]D-Fender | killfill, Or a PRI > VoIP gateway (anythin remotely decent will have a failover server. |
| 02:59.25 | JT | either way, it's not that cheap :P |
| 02:59.47 | killfill | [TK]D-Fender: what do you mean by a gateway.. an asterisk with pri is a gateway.. isnit? |
| 02:59.58 | JT | killfill: a hardware embedded gateway |
| 03:00.07 | killfill | hm.. |
| 03:00.08 | JT | pri to sip |
| 03:00.30 | [TK]D-Fender | killfill, Yes, I'm talking like an AudioCodes Mediant 2000 |
| 03:00.47 | killfill | oh.. like pstn----> [pri,sip] <--- asterisk pbx's?.. i.e. make asterisks conect via SIP to this box? |
| 03:00.57 | IOscanner | Anyone using 4port FXO cards. I have boxes with 5 ports - and another with 12 ports. I am hearing echo. Only the phone (Cisco 7940) on asterisk hears the echo not the caller from external. Any ideas? |
| 03:01.06 | IOscanner | We have tested the lines. |
| 03:01.17 | [TK]D-Fender | killfill, http://www.voipsupply.com/product_info.php?products_id=1039 |
| 03:01.27 | [TK]D-Fender | killfill, Yes |
| 03:01.49 | blitzrage | [TK]D-Fender: I don't want to meet your mom |
| 03:01.57 | IOscanner | Installed octware that helped some, but still have echo on our side. The caller from external is clean. |
| 03:02.14 | [TK]D-Fender | IOscanner, Zaptel EC and/or your cards suck. |
| 03:02.24 | [TK]D-Fender | blitzrage, I just want... |
| 03:02.31 | IOscanner | Here is what I have in zaptel: http://paste.uni.cc/15964 |
| 03:02.34 | blitzrage | ! ! ! |
| 03:02.48 | IOscanner | ZaptelEC? |
| 03:02.51 | blitzrage | [TK]D-Fender: have you ever wondered why no one has said, "that joke is sooooo old" |
| 03:03.11 | *** join/#asterisk bugzee (n=rickw@ppp-70-128-124-134.dsl.tulsok.swbell.net) |
| 03:03.21 | [TK]D-Fender | blitzrage, because one we understand it (that's the point of an inside joke) |
| 03:03.31 | [TK]D-Fender | only* |
| 03:03.38 | JT | IOscanner: echo is usually caused by far end analogue lines |
| 03:03.39 | killfill | hm.. and that box.. is actually a PBX?.. i dont really need a separate asterisk box, isnit?.. i can point the sip phones over it?.. |
| 03:03.45 | [TK]D-Fender | blitzrage, And you know what... it just keeps getting better! |
| 03:03.47 | blitzrage | ya, but you'd think people would have seen us saying it a lot over a long period of time :) |
| 03:03.50 | blitzrage | it totally does |
| 03:03.54 | JT | killfill: no, it's a media gateway, not a pb |
| 03:03.55 | JT | pbx |
| 03:03.59 | JT | it doesn't do pbx stuff |
| 03:04.02 | killfill | ah |
| 03:04.11 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.11 | [TK]D-Fender | killfill, No, its not a PBX. It takes the calls in from T1 and spits out SIP. Thats all. |
| 03:04.39 | IOscanner | Well the echo only happens on our side the person on the Cisco phone can hear only their voice echo. the caller can't heer it. It is not present when using VOIP. |
| 03:04.53 | JT | killfill: most ITSPs use boxes like that |
| 03:04.55 | JT | IOscanner: i know |
| 03:04.58 | killfill | hm.. |
| 03:05.02 | JT | IOscanner: that is cimpletely normal |
| 03:05.14 | JT | IOscanner: it is that far end analogue lines doing it |
| 03:05.24 | killfill | so if i want reduncacy.. i would use standart IP methods.. (like the tyipical IP switching like CARP)... |
| 03:05.26 | JT | it becomes aparent when you convert to digital, if there's no EC |
| 03:06.09 | IOscanner | What far end? That can't be the old PBX system didn't have a problem with it. |
| 03:06.09 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-224-149.dhcp.insightbb.com) [NETSPLIT VICTIM] |
| 03:06.09 | *** join/#asterisk SplasPood (n=nnnijwb@schizophrenia.paravolve.net) [NETSPLIT VICTIM] |
| 03:06.09 | JT | IOscanner: was the old pbx analogue? |
| 03:06.11 | IOscanner | yep |
| 03:06.12 | killfill | and coule easily do a round-robin on it too.. like for scalling |
| 03:06.14 | JT | IOscanner: the phone line on the other end of the conversation |
| 03:06.17 | killfill | is this true?.. |
| 03:06.26 | IOscanner | The call on the other end was a cell phone |
| 03:06.36 | JT | IOscanner: echo isn't really apparent if both ends are analogue |
| 03:06.40 | IOscanner | doesn't matter if it is cell phone, VOIP. |
| 03:06.52 | IOscanner | even analog |
| 03:07.19 | IOscanner | If we call inbound on the analog lines echo for the user at the cisco phone |
| 03:07.47 | *** join/#asterisk guille1983 (n=chatzill@190.73.188.118) |
| 03:08.06 | JT | well it's also possible you have bad phone lines to your ast server |
| 03:08.16 | [TK]D-Fender | IOscanner, Tweak your software EC or get a HWEC card instead. |
| 03:08.25 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 03:08.36 | JT | i hate daling with analogue |
| 03:08.40 | JT | dealing |
| 03:08.53 | IOscanner | Nope tested the line. connected it back to the old system and it is fine. I even put octware on it. |
| 03:09.00 | killfill | the cheaer is like 3000.. |
| 03:09.01 | killfill | hm.. |
| 03:09.06 | IOscanner | It helped a little |
| 03:09.33 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 03:09.42 | IOscanner | Any other tests I can do with asterisk to tune EC |
| 03:09.53 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-9215168ddeb72ac8) [NETSPLIT VICTIM] |
| 03:10.02 | [TK]D-Fender | ~echo |
| 03:10.19 | jbot | methinks echo is an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ... |
| 03:10.20 | killfill | and something like this? http://www.telephonyware.com/telephonyware/tw00381.html |
| 03:10.20 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) [NETSPLIT VICTIM] |
| 03:10.20 | killfill | thats not a pri->sip gateways.. isnit.. |
| 03:10.25 | JT | IOscanner: please don't keep saying things like "old system was just fine", your old system operates COMPLETELY DIFFERENTLY to asterisk |
| 03:10.34 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) [NETSPLIT VICTIM] |
| 03:10.40 | [TK]D-Fender | killfill, That might work, but ICK... not standards based so I wouldn't touch it. |
| 03:10.51 | JT | if it was analogue, or digital with hw ec, completely different kettle of fish |
| 03:10.53 | killfill | :S |
| 03:11.30 | JT | killfill: also, redfone units whilst cheap, have no HW EC |
| 03:12.18 | JT | they use TDMoE, not SIP |
| 03:12.29 | JT | that's the non standards based bit |
| 03:12.30 | toerkeium | guys, what's the reason why a single sip call could make the server response time higher from, lets say 300ms to 2000ms? it only happens some times and with only 1 sip call |
| 03:12.31 | IOscanner | Yes, but the lines are clean. So I know it is something with the cards or configuration of the EC |
| 03:12.41 | IOscanner | Thanks I will review the links thanks |
| 03:12.43 | JT | IOscanner: ok, get a card with hardware EC |
| 03:12.56 | JT | IOscanner: how did you test the lines? |
| 03:13.54 | killfill | JT, asterisk supports TDMoE.. what would i a practical disadvantage about using TDMoE? |
| 03:14.21 | *** join/#asterisk mihinomenest (i=M0Ur@cerebus.clandestineresearch.com) |
| 03:14.27 | JT | killfill: you can only use asterisk, and you have to depend on a rarely used aspect of asterisk working correctly and continuing to be supported |
| 03:15.02 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 03:15.05 | *** join/#asterisk Nivex (n=kjotte@user-0ce2kma.cable.mindspring.com) |
| 03:15.16 | IOscanner | Fluke tester |
| 03:15.42 | JT | what sort of test does it perform? |
| 03:16.34 | IOscanner | connects a call over it and tests quality, audio, noise looks for wire issues etc. |
| 03:16.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:16.53 | JT | how long is the pair? |
| 03:16.54 | killfill | hm.. |
| 03:17.04 | IOscanner | 5 feet |
| 03:17.19 | IOscanner | wired direct to the 66 block |
| 03:17.20 | killfill | so TDMoE is cheap but non-standart (and can see plent of ppl having probls with it) |
| 03:17.21 | JT | IOscanner: err, is this the wire to the telco... the analogue FXO lines? |
| 03:17.33 | JT | IOscanner: the 66 block has more wire on the other side |
| 03:17.43 | JT | goes in the street |
| 03:17.44 | killfill | a gateway is too high.. (min is like 3000, its out of my budget).. |
| 03:17.45 | JT | to the CO |
| 03:17.54 | IOscanner | yep |
| 03:17.58 | killfill | JT: what did you mean by " an L1 PRI failover hardware box" ? |
| 03:18.11 | JT | IOscanner: that's what i meant by how long, not just one part of it |
| 03:18.16 | JT | killfill: junghanns.net makes one |
| 03:18.19 | IOscanner | They have DSL in the location and it was under 1,000 |
| 03:18.20 | JT | a few others do to |
| 03:18.41 | JT | "t1 failover" the others may not be designed for asterisk as much as the junghanns one is |
| 03:18.53 | JT | IOscanner: 1000 what? |
| 03:18.54 | IOscanner | They have a 7MB link and the CO is around the corner |
| 03:18.58 | IOscanner | ft |
| 03:18.59 | JT | ok |
| 03:19.16 | JT | IOscanner: is it all calls that sound echoy, or just some? |
| 03:19.43 | IOscanner | some not all some are better then others. |
| 03:19.56 | JT | exactly |
| 03:20.00 | JT | it's the far end lines |
| 03:20.04 | IOscanner | all seem to do it a bit |
| 03:20.11 | JT | anyway, best bet is to buy a card with hardware EC |
| 03:20.16 | IOscanner | How can I resolve it. EC training |
| 03:20.41 | IOscanner | Do they have them for 4port FXO? |
| 03:20.47 | JT | digium's TDM2400P has that option, and sangoma's A200 and A400 have the option i think, and also come with 6 chans of software premium EC now i think |
| 03:21.00 | JT | IOscanner: you can get a TDM2400P with just 4 FXO ports |
| 03:21.17 | IOscanner | What about something like Octware? |
| 03:22.01 | *** join/#asterisk saftsack (n=oliver@p54A7CDAA.dip.t-dialin.net) |
| 03:22.13 | JT | yes |
| 03:22.30 | JT | they do hw and sw solutions |
| 03:22.31 | IOscanner | I have it installed on this box |
| 03:22.34 | JT | hw obviously works better |
| 03:23.20 | blitzrage | anyone verify if I can negate an expression via: !$[...] ? |
| 03:24.53 | [TK]D-Fender | blitzrage, You'd have to do it in an eval |
| 03:25.06 | [TK]D-Fender | blitzrage, Because the negate itself is a expression :) |
| 03:25.19 | IOscanner | Thanks I will go play and see what I can do |
| 03:25.21 | *** join/#asterisk teyus (i=Mateus@unaffiliated/teyus) |
| 03:25.38 | blitzrage | [TK]D-Fender: actually, that's what I was doing: $[!$[...] & $[...]] |
| 03:25.46 | blitzrage | although I realized I don't need to use it :) |
| 03:25.51 | blitzrage | I was making the expression too complex |
| 03:25.56 | blitzrage | but I think that'll work |
| 03:26.01 | [TK]D-Fender | blitzrage, see, you don't even fully understand jsut how right you are! |
| 03:26.16 | [TK]D-Fender | AWESOME |
| 03:26.59 | blitzrage | I'lll show you the whole Exec() when I'm done here :) |
| 03:28.31 | [TK]D-Fender | blitzrage, I love it when you code dirty ;) |
| 03:28.41 | [TK]D-Fender | blitzrage, ! ! ! |
| 03:28.43 | [TK]D-Fender | :O |
| 03:28.48 | blitzrage | lol |
| 03:28.54 | blitzrage | just wait.... |
| 03:33.40 | [TK]D-Fender | "Rob say Code Monkey very diligent, but his output stink. His code not functional or elegant, what do Code Monkey think?" |
| 03:33.51 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:33.52 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:38.26 | blitzrage | [TK]D-Fender: here it is |
| 03:38.29 | blitzrage | exten => h,n,Exec(${IF($[${GROUP_COUNT(${USERNAME}@${PBX})} > 0 | ${GROUP_COUNT(${USERNAME}@${PBX}-internal)} > 0 | ${ISNULL(${USERNAME})} | ${ISNULL(${PBX})}]?NoOp():${IF($[${EXISTS(${FROM_QUEUE})} & "${DIALSTATUS}" != "ANSWER"]?RemoveQueueMember(${HASH(queue|name)}|Local/${MEMBER_ID}-${MEMBER_TECH}-${MEMBER_NAME}@queue_members/n):NoOp())}) |
| 03:39.27 | Qwell | blitzrage: You need to write a new chapter for the book |
| 03:39.38 | Qwell | "How do get a consultancy gig - and keep it." |
| 03:39.58 | Qwell | "Step 1) write dialplan that's nearly impossible for others to read" :P |
| 03:40.33 | mog | lol |
| 03:40.39 | Qwell | or that should be a question on the dcap |
| 03:40.48 | Qwell | "What does this line of dialplan do?" |
| 03:41.07 | blitzrage | This code segment checks to see if a phone is using atleast one channel, if so, then we run NoOp() (we've already logged them out somewhere else -- in the queue GoSub(sub-queues)). Else, then check to see if the Local channel which determines the status of the Agent was attempted to be called, and if they were called, but they did not answer, then they are probably not at their desk, therefore, remove them from the Qu |
| 03:41.07 | blitzrage | eue() |
| 03:41.18 | blitzrage | Qwell: those would be great questions actually |
| 03:41.57 | Qwell | there are probably really not that many people who could explain what that does |
| 03:42.01 | blitzrage | I've been looking and writing embedded applications and dialplans so much in the last 8 months that I typically don't need syntax highlighting and still know how many brackets to put on them all |
| 03:42.10 | blitzrage | Qwell: I would think not |
| 03:42.13 | blitzrage | I have mad dialplan foo |
| 03:42.17 | Qwell | totally |
| 03:42.33 | Qwell | I'm shocked though... no odbc calls there |
| 03:42.34 | *** join/#asterisk saftsack (n=oliver@p54a7c48b.dip.t-dialin.net) |
| 03:42.39 | blitzrage | not a single one there :) |
| 03:42.48 | blitzrage | the line above it does have it though |
| 03:42.52 | Qwell | ahh, heh |
| 03:42.58 | blitzrage | ; Set the peer status in the relational database for next caller into the Queue() (not active to this caller incase we loop) |
| 03:42.58 | blitzrage | exten => h,n,Exec(${IF($[${GROUP_COUNT(${USERNAME}@${PBX})} > 0 | ${GROUP_COUNT(${USERNAME}@${PBX}-internal)} > 0 | ${ISNULL(${USERNAME})} | ${ISNULL(${PBX})}]?NoOp():Set(QUEUE_MEMBER_STATUS(SIP,${USERNAME}#${PBX})=${IF($[${EXISTS(${FROM_QUEUE})} & "${DIALSTATUS}" != "ANSWER"]?0:1)}))}) |
| 03:43.03 | Qwell | That must be where USERNAME and FROM_QUEUEgets set ;) |
| 03:43.20 | blitzrage | nope -- that is set in the sub-queues.inc file |
| 03:43.26 | blitzrage | which is a subprocedure (GoSub()) |
| 03:43.42 | Qwell | oh :p |
| 03:43.43 | blitzrage | all my features and such are in a separate 'features' directory, which contains variable include files |
| 03:43.49 | boch | why not using AEL to avoid that kind of exts? |
| 03:44.01 | blitzrage | AEL would still require something like that |
| 03:44.33 | blitzrage | it would just be slightly different syntax |
| 03:44.53 | blitzrage | but I don't think dialplan functions and applications are handled any differently there |
| 03:44.59 | boch | but easier to understand |
| 03:44.59 | blitzrage | just the exten => h,n, part really |
| 03:45.02 | blitzrage | how? |
| 03:45.14 | [TK]D-Fender | blitzrage, That is one nested psycho pile of code... |
| 03:45.28 | blitzrage | although I admit my lack of AEL foo |
| 03:45.38 | blitzrage | but from my understanding of it, I don't think ti would look much different |
| 03:45.44 | blitzrage | [TK]D-Fender: only if you're not used to looking at something like that |
| 03:46.09 | [TK]D-Fender | blitzrage, .... www.drphil.com ... before its too late... |
| 03:46.24 | blitzrage | I could probably teach a whole week class just on my dialplan code :) |
| 03:46.25 | Qwell | blitzrage: ${USERNAME}#${PBX} < typo? |
| 03:46.44 | blitzrage | Qwell: nope -- we parse on a [username#vpbx] username |
| 03:46.49 | blitzrage | (sip.conf) |
| 03:47.12 | blitzrage | but when I bring it in I split the username#vpbx into two separate variables |
| 03:47.22 | blitzrage | that's just putting them back together for the sake of the database, etc.... |
| 03:48.39 | blitzrage | I love when stuff works first try |
| 03:52.48 | *** join/#asterisk bmg505 (n=leon@196.209.176.213) |
| 03:54.03 | Aces1Up | yeh, i am installing asterisk NOW, so i can be just like you guys! YES! |
| 03:54.15 | JT | :o |
| 03:55.35 | [TK]D-Fender | Aces1Up, Because yes, we clearly all use GUI's to run our show! |
| 03:56.51 | Aces1Up | with asterisk NOW, can i get down to the nitty gritty (nacho libre style) and code where you guys are at? |
| 03:58.01 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 03:58.11 | [TK]D-Fender | Aces1Up, there will always be some freakish it of something that it creates that you won't like how it does it. You'll either suffer (loud or quiet), or move on. |
| 03:58.57 | Aces1Up | tkd, hrmm, ok well should i just install linux then put then install asterisk and configure it all manually? would that be the way to go? |
| 03:59.33 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:59.57 | Aces1Up | i suppose that would be the best way if i want to really learn the system eh? |
| 04:00.10 | JT | ~thebook |
| 04:00.35 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:01.11 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 04:01.33 | *** join/#asterisk saftsack (n=oliver@84.167.195.243) |
| 04:01.58 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) [NETSPLIT VICTIM] |
| 04:07.42 | blitzrage | Aces1Up: depends how much time you need to spend coding in Asterisk |
| 04:07.52 | blitzrage | I've been at it for 5 years to be able to understand the dialplan I wrote up there |
| 04:09.04 | Aces1Up | blitz well.. hrmm. |
| 04:09.45 | Aces1Up | want to be up and running quickly but also be able to learn with no umm i guess cushions that will cause me not to get to the guts of the system. |
| 04:10.07 | Aces1Up | I want to eventually develop these sweet new applications everyone is talking about for the future. |
| 04:13.38 | Aces1Up | just curious what flavor of linux has the most support for asterisk? |
| 04:13.47 | Aces1Up | just trying to figure out what flavor to use. |
| 04:14.28 | mosty | use whichever linux distribution you are the most comfortable with |
| 04:14.46 | mosty | there is no real difference in support between the major distributions |
| 04:15.05 | ManxPower | Aces1Up: the people that know AsteriskNOW are, oddly enough, on the #asterisknow channel. |
| 04:15.42 | ManxPower | Aces1Up: And I want a Unicorn. |
| 04:15.43 | Aces1Up | i've thrown that idea out the window, just going to compile it myself. |
| 04:16.11 | Aces1Up | so just curious what flavor if i run will give me the most support for asterisk. |
| 04:16.31 | JT | no difference |
| 04:16.33 | JT | seriously |
| 04:16.45 | JT | if you're compiling, it's a pointless question really |
| 04:16.53 | JT | whatever you know best... |
| 04:17.51 | ManxPower | Aces1Up: most distros have very few asterisk oddities with Astertisk |
| 04:18.44 | Aces1Up | manxpower ok. sounds good thanks for the help. |
| 04:18.50 | Aces1Up | and no unicorn just yet ok? |
| 04:19.27 | ManxPower | Aces1Up: if you can find a decent GUI for Asterisk, then there is still hope for my search. |
| 04:29.26 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 04:31.07 | mosty | i want to design a PBX that runs on multiple asterisk servers for fault tolerance, with SIP and IAX clients, what would be a good way to do this? dundi for IAX and SRV records and/or openser for SIP? |
| 04:31.31 | JT | it depends what sort of fault you're trying to mitigate against, and what you want impact you want a fault to have |
| 04:32.58 | mosty | i'll start with 2 asterisk servers, if asterisk dies on one, i would like clients to automatically register to the other, and new calls to be made successfully |
| 04:33.49 | JT | what are the inputs and outputs? |
| 04:34.59 | mosty | each asterisk server will have PRI circuits and IAX/SIP accounts for terminating PSTN calls |
| 04:35.44 | mosty | and each server should allow any known client (SIP/IAX) to register to it in and make/receive calls |
| 04:36.08 | mosty | clients should also be able to call each other regardless of which server they're registered to |
| 04:37.36 | JT | different pri circuits at each one, or the same ones? |
| 04:38.35 | mosty | different PRI circuits (i wasn't aware that you could share them) |
| 04:38.51 | JT | you can with a pri L1 failover box |
| 04:40.20 | mosty | ok well i'll start with different PRI circuits and consider that later |
| 04:41.08 | mosty | jt: do you understand what i'm trying to do? |
| 04:41.08 | JT | how many pris do you have? |
| 04:41.08 | JT | yes |
| 04:41.29 | mosty | i'm not sure of the exact number, at least 5 from memory |
| 04:42.09 | JT | do any or them use NFAS? |
| 04:42.25 | mosty | JT, i'm not sure, i would have to contact our telco's tech support |
| 04:42.40 | JT | do they exist yet or not, these pris? |
| 04:42.49 | JT | are they your pris or customer pris |
| 04:42.53 | JT | because if they're yours |
| 04:42.57 | JT | makes sense to use nfas |
| 04:43.15 | mosty | they exist and are in use now, there is one that isn't currently plugged in to anything |
| 04:43.34 | JT | or do they connect to different telcos |
| 04:44.01 | mosty | they all connect to the same telco |
| 04:44.24 | JT | nfas allows grouping a number of PRIs to share D channels |
| 04:44.51 | JT | it gives you more B channels and allows you to easily combine the services basically |
| 04:45.20 | mosty | i'm reading about it now. does nfas allow you to share a single d channel between 2 different machines with a pri card each? |
| 04:45.34 | JT | no, not with asterisk anyway |
| 04:45.44 | JT | you need a minimum of 2 D channels for NFAS |
| 04:45.46 | JT | for reliability |
| 04:46.45 | mosty | well nfas won't really help in this case, we are trying to cover the situation where asterisk crashes (hence wanting to run asterisk on two machines with clients failing over to the other server if the one they're connected to dies) |
| 04:46.55 | *** join/#asterisk saftsack (n=oliver@p54A7EEC5.dip.t-dialin.net) |
| 04:47.23 | JT | mosty: yes but 2 machines with a pri card, nfas, and an L1 failover box will help |
| 04:47.29 | mosty | ahh ok |
| 04:47.32 | *** join/#asterisk dunder (n=grndslm@23-67.69-92-cpe.cableone.net) |
| 04:47.57 | JT | nfas mitigates the problem of one pri dieing causing loss of connectivity to your DIDs |
| 04:48.08 | mosty | but how do i get sip/iax clients to failover from one asterisk server to another? i'm more worried about that right now than the termination side |
| 04:48.44 | JT | ip takeover would probably be the most seamless scheme |
| 04:48.58 | mosty | i would also like to use some sort of load balancing |
| 04:49.01 | JT | otherwise you could try using SRV records |
| 04:49.19 | JT | oh and sip is easier |
| 04:49.29 | JT | as you can stick openser in front of the boxes |
| 04:49.38 | JT | as a load balancer/failover |
| 04:49.47 | mosty | one problem with SRV is that we have lots of asterisk 1.2 clients, and it looks like 1.2 doesn't support SRV for IAX |
| 04:50.17 | JT | do you really have that much iax clients? iax is much dirtier for failover |
| 04:50.34 | mosty | most of our larger clients use iax trunks |
| 04:50.46 | mosty | most of our single extension clients are sip |
| 04:50.52 | JT | damn, that sucks |
| 04:51.10 | JT | imho iax isn't as scalable |
| 04:51.48 | mosty | we're using iax for bandwidth savings and nat survivability |
| 04:52.12 | JT | the tradeoff is practical scalability solutions |
| 04:52.41 | mosty | i could just ask our iax clients to use some dialplan magic to try dialling via each of our gateways randomly, but that wouldn't solve the registration problem |
| 04:53.08 | JT | just do dns or ip takeover |
| 04:53.54 | JT | are you mainly trying to guard against hardware or software failure? |
| 04:54.48 | mosty | JT, software primarily (asterisk crashes), but i would like it to guard against hardware failure also |
| 04:55.04 | JT | hrm |
| 04:55.18 | *** join/#asterisk AdamW (n=AdamW@mandriva/channel-support/AdamW) |
| 04:55.22 | JT | there are solutions on the market that almost mitigate hardware failure |
| 04:55.31 | JT | but yeah, software is still an issue |
| 04:55.45 | *** part/#asterisk AdamW (n=AdamW@mandriva/channel-support/AdamW) |
| 04:56.14 | *** join/#asterisk saftsack (n=oliver@p54A7E6CE.dip.t-dialin.net) |
| 04:57.18 | *** join/#asterisk AdamW (n=AdamW@mandriva/channel-support/AdamW) |
| 04:57.24 | *** part/#asterisk AdamW (n=AdamW@mandriva/channel-support/AdamW) |
| 04:58.41 | JT | mosty: you will likely need some sort of load balancer in any case |
| 04:58.49 | JT | be it software or hardware based |
| 04:59.54 | mosty | jt: i'm trying to figure out what that should be. my current thoughts are SRV and/or openser for SIP, and dundi for IAX |
| 05:00.32 | JT | iax may need physical packet redirection, i dunno |
| 05:02.09 | *** join/#asterisk slingr (n=san@helix.fiberuplink.com) |
| 05:02.14 | slingr | hello all |
| 05:02.54 | *** part/#asterisk slingr (n=san@helix.fiberuplink.com) |
| 05:04.20 | *** join/#asterisk mkl1525 (n=qwertz@pD953057F.dip0.t-ipconnect.de) |
| 05:04.45 | mosty | jt: it looks like dundi might be a good start, but i don't understand it well enough to know |
| 05:05.36 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 05:06.08 | JT | yeah, no idea about if dundi will help |
| 05:08.20 | *** join/#asterisk wotcha (n=jim@cust4716.qld01.aanet.com.au) |
| 05:09.00 | mosty | i wonder if there's a macro-like feature for iax so that asterisk can register to multiple iax servers |
| 05:09.36 | mosty | without having to manually manage N different IAX server account details |
| 05:11.22 | nowork | JT: can you please help me on a SIP issue:: pp_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 05:11.41 | JT | nowork: that's a pretty straightforeward error |
| 05:11.56 | JT | means it can't establish a sip link at the ip level |
| 05:12.36 | nowork | hhmm, when i do sip show peers , i got status UNREACHABLE |
| 05:12.46 | JT | that's a given |
| 05:12.50 | nowork | but, I can ping the remote ip |
| 05:13.13 | JT | that doesn't mean sip is running on it |
| 05:13.53 | nowork | so, I should check with remote end? I hv this same issue on three remote ip , that is why i thought my problem |
| 05:14.13 | nowork | they asked me to treat them as gateway. they use VPS and MERA |
| 05:14.38 | JT | maybe you have a firewall |
| 05:14.40 | JT | or they do |
| 05:16.50 | nowork | ok, I will doublecheck this ..thank you JT.. |
| 05:17.42 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 05:23.25 | *** join/#asterisk boch (n=fran@190.48.214.119) |
| 05:26.08 | mkl1525 | Hi, havin problems with queue commands timeout. the agents have "exten => s,1,Dial(SIP/${ARG1},900)" as call option and the Queue a timeout of 15 seconds. but the queue won't timeout in 15 seconds but after 900 - so did I something wrong? |
| 05:26.28 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 05:29.44 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 05:31.13 | *** join/#asterisk saftsack (n=oliver@p54A7CD18.dip.t-dialin.net) |
| 05:31.59 | mosty | jt: i have an idea for the delivery of calls to clients in the case where the server they're registered to dies- if each server caches a list of registered peers at itself and the other servers, you could use AGI to figure out the location of dynamic hosts (if one is known), the only requirement is that the client accepts calls from unknown servers |
| 05:32.35 | JT | how is the server unknown? |
| 05:33.14 | mosty | jt: i want the asterisk server pool to be able to grow without having to let the clients know |
| 05:33.44 | JT | why would the client not accept the call anyway? |
| 05:33.53 | JT | if it's authed |
| 05:34.16 | mosty | if it's authed it will work |
| 05:34.35 | JT | then it should be fine |
| 05:34.37 | mosty | but does the client know about all N asterisk servers? |
| 05:34.45 | mosty | assuming it's using SRV |
| 05:34.49 | JT | it should auth on call setup |
| 05:36.57 | ManxPower | um, Asterisk will pretty much accept unauthed calls anyway |
| 05:36.57 | mosty | ManxPower, it's configurable in the dialplan obviously, yes |
| 05:38.12 | mosty | this sounds like it would work reasonably well, now i just need to make sure that dundi or something else can't do it for me already |
| 05:40.52 | ManxPower | mosty: Yeah. It annoys me that Asterisk is basically insecure by default. |
| 05:42.26 | mosty | ManxPower, default should be whatever configuration it ships with |
| 05:42.39 | ManxPower | mosty: it does not ship with a working config |
| 05:42.59 | mosty | ManxPower, that's my point. no working config, is perfectly secure |
| 05:46.56 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 05:53.13 | *** join/#asterisk jacq (n=jal@203.187.143.130) |
| 06:03.22 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 06:03.22 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits, #astridevcon -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 06:03.37 | *** join/#asterisk NirS (i=Nir@87.68.2.248.cable.012.net.il) |
| 06:05.34 | mkl1525 | and another Problem: when using voicemail I see the recording in the tmp of the voice box and it grows larger but after caller quits the tmp file is removed but not in INBOX - any hints where the file goes to? debug + verbose 10 weren't helpful |
| 06:09.11 | mosty | mkl1525, is it being emailed and deleted? |
| 06:11.00 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 06:11.16 | *** join/#asterisk gmustafa (n=gmustafa@202.5.145.13) |
| 06:13.03 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:15.57 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
| 06:17.05 | mkl1525 | mosty, thanks you're my hero :) - it was too obvious... |
| 06:19.34 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 06:26.29 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 06:31.58 | *** join/#asterisk zol_ (n=z@AClermont-Ferrand-156-1-79-186.w90-9.abo.wanadoo.fr) |
| 06:37.18 | *** join/#asterisk NirS_ (i=Nir@87.68.5.42.cable.012.net.il) |
| 06:39.31 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:43.56 | Shaun2222 | whats the usb port for on the polycom 650's |
| 06:44.59 | nowork | hi, Any idea how much is a 1port FXO PCI card for Asterisk? |
| 06:45.11 | nowork | or FXO 1port sip device ? |
| 06:45.19 | *** join/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 06:45.22 | n0n4m3 | morning |
| 06:46.22 | n0n4m3 | i've got a little ol' question |
| 06:46.58 | Hmmhesays | heh |
| 06:47.10 | n0n4m3 | i'd like to change the sip header in case a call comes in from one sip account |
| 06:47.12 | drrt | nowork, 1 port fxo card for zap channel is simply analog modem |
| 06:47.46 | Hmmhesays | how so? |
| 06:48.39 | n0n4m3 | in case i call to freewordldialup i have to enter prefix 9... and i'd like to set up dialing plan so in case a call comes from fwd, i'd like to add a new prefix, 9 |
| 06:51.36 | *** join/#asterisk sundarr (n=sundarr@122.167.64.129) |
| 06:52.28 | n0n4m3 | exten => _9.,3,Dial(SIP/${EXTEN:1}@fwd-outgoing) |
| 06:53.14 | n0n4m3 | so here's my outgoing plan |
| 06:53.41 | n0n4m3 | incoming is like this |
| 06:53.46 | n0n4m3 | exten => ${FWDNUMBER},3,Set(CALLERID(number)=9${NUM}) |
| 06:53.53 | n0n4m3 | but unfortunately it doesn't work |
| 06:54.00 | n0n4m3 | i keep getting 9 |
| 06:54.04 | n0n4m3 | only 9 |
| 06:54.44 | flenders | ~pb |
| 06:55.27 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 06:59.39 | *** part/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 07:02.22 | ManxPower | n0n4m3: The value of ${NUM} is empty or was never set |
| 07:04.00 | ManxPower | Perhaps you are looking more for: exten => ${FWDNUMBER},3,Set(CALLERID(number)=9${CALLERID(number)}) |
| 07:09.53 | creativx | damn ip10s' keep falling out of the network |
| 07:09.53 | creativx | wtf |
| 07:13.46 | Aces1Up | is there a flowchart that shows the various config files and their interaction with each other? |
| 07:31.52 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 07:31.58 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 07:35.59 | n0n4m3 | ManxPower prolly... i'll check it out |
| 07:40.11 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 07:40.53 | *** part/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 07:41.24 | Zeeek | looks good |
| 07:41.28 | Avochelm | just set up two clients on our internal pabx to forward to each other... went up to about 38 missed calls in a matter of seconds... |
| 07:41.41 | Avochelm | nice little infinite loop... going to see if we can crash the server :) |
| 07:41.54 | JT | i bet they love it :P |
| 07:42.39 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 07:49.03 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 07:51.20 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 07:55.56 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 07:57.45 | Uatec | lol |
| 07:57.59 | Uatec | i'm going to give that a go |
| 08:00.49 | *** join/#asterisk samlt (n=sam@fla93-1-81-57-168-33.fbx.proxad.net) |
| 08:01.16 | n0n4m3 | bummer |
| 08:01.32 | n0n4m3 | fwd says i've got misconfigured asterisk :( |
| 08:02.02 | Zeeek | sip or iax? |
| 08:05.11 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:08.19 | *** join/#asterisk grEvenX (n=even@ti500720a080-7918.bb.online.no) |
| 08:10.10 | n0n4m3 | sip |
| 08:10.28 | n0n4m3 | the strangest thing is... it seems to work if i call myself |
| 08:10.29 | n0n4m3 | :D |
| 08:10.48 | Zeeek | so all is well? |
| 08:12.49 | n0n4m3 | no... the call test doesn't work :( |
| 08:13.06 | Zeeek | echo test? Time of day? |
| 08:13.08 | n0n4m3 | do you happen to have account at freeworlddial? |
| 08:13.19 | Zeeek | I used to use it |
| 08:13.29 | Zeeek | not registered with the server though |
| 08:13.37 | n0n4m3 | bummer |
| 08:13.42 | Zeeek | pastebin your configs and I'll compare them to mine |
| 08:13.48 | *** part/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 08:14.18 | Zeeek | FWD was down too often to be of use even testing |
| 08:18.03 | n0n4m3 | http://rula.net/12 |
| 08:18.06 | n0n4m3 | here you go |
| 08:18.15 | n0n4m3 | i belive i pasted all the necessary parts |
| 08:18.27 | Zeeek | are your registered with fwd ok? |
| 08:18.39 | n0n4m3 | it seems so |
| 08:19.07 | n0n4m3 | i don't get any errors when i run asterisk |
| 08:21.03 | Zeeek | mine is a peer without fromdomain for outgoing |
| 08:22.59 | Zeeek | which part gives you problems, calling out or incoming? |
| 08:26.52 | *** join/#asterisk cayorde (n=flexable@host164-105-dynamic.16-87-r.retail.telecomitalia.it) |
| 08:30.30 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:30.50 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:31.32 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 08:36.17 | *** join/#asterisk extr3m (i=caligo@c-d87d71d5.011-143-6c756c1.cust.bredbandsbolaget.se) |
| 08:37.38 | *** join/#asterisk matsk (n=mk@194.68.102.172) |
| 08:47.04 | creativx | anyone here has any suggestions to bluetooth headsets/adapters for windows + softphone? |
| 08:48.18 | *** join/#asterisk SoftIce (n=psmith@dsl-242-118-235.telkomadsl.co.za) |
| 08:48.31 | SoftIce | hi anyone ever had this error Invalid card number lenght defined in configuration with a2billing when trying to view the Admin page? |
| 08:49.28 | dejandinic | Hello ppl, is there any chance to send "beep" sound on OTHER conversation made by dial(zap/g1/${EXTENSIONS}) |
| 08:49.57 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:51.08 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 08:56.47 | n0n4m3 | Zeeek the problem is the test call |
| 08:56.51 | n0n4m3 | from the web |
| 08:57.34 | Zeeek | best way to fix it is to ask someone on the fwd forum to call you |
| 08:57.56 | Zeeek | or find someone patient enough to call one of their DID and enter the codes to call you |
| 09:00.53 | Uatec | i've got 10 smart phones to connect to my asterisk system |
| 09:01.05 | Uatec | it helps the stupid wifi stay up long enough to figure out why sound is only going one way |
| 09:05.59 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:09.47 | *** join/#asterisk tenzind (n=tenzind@202.144.144.11) |
| 09:28.10 | *** join/#asterisk friedrich| (n=friedric@e177240072.adsl.alicedsl.de) |
| 09:28.49 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:32.11 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 09:33.28 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 09:38.43 | *** join/#asterisk pestouille (n=adrien@meu77-2-82-244-230-97.fbx.proxad.net) |
| 09:38.52 | pestouille | Hi |
| 09:39.51 | pestouille | I would like to use an ISDN Diva 2.02 PCI card but I can't find any driver for it using mISDN or CAPI just the I4L. So I should use the chan_modem_i4l but it seems deprecated any ideas ? |
| 09:42.48 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 09:42.55 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:46.30 | *** join/#asterisk stony (n=oloch@p5B152131.dip0.t-ipconnect.de) |
| 09:46.31 | stony | hi |
| 09:46.44 | stony | i'm trying to use a linksys spa-901 with asterisk but it isn't working |
| 09:46.56 | stony | are there any howtos available for telephone configuration ? |
| 09:47.03 | stony | i googled around but i didn't find anything |
| 09:52.12 | *** join/#asterisk pejo_ (n=peter@136.240.13.217.in-addr.dgcsystems.net) |
| 09:54.18 | *** join/#asterisk SoftIce (n=bongo@vc-196-207-45-253.3g.vodacom.co.za) |
| 09:54.32 | SoftIce | good day, i'm having an issue with a2billing, when I go to the admin page I get this displayed on my browser. |
| 09:54.40 | SoftIce | Invalid card number lenght defined in configuration |
| 09:54.58 | SoftIce | I find only 1 google post about this issue and half way down the user states he allready solved the issue without stating what the issue was |
| 09:58.27 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 10:00.53 | *** join/#asterisk n0n4m3 (n=NoName@noname.rula.net) |
| 10:04.47 | Uatec | it's helpful when people do that, isn't it... |
| 10:06.57 | creativx | indeed |
| 10:08.02 | *** join/#asterisk porche (n=porche@88.239.82.203) |
| 10:08.18 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 10:08.27 | porche | hi there |
| 10:08.45 | porche | any1 experienced in analog lines + digium cards? |
| 10:09.09 | Strom_C | I am...but only if you promise not to type stupid things like "any1" |
| 10:09.25 | creativx | how about anyein |
| 10:09.30 | creativx | its german |
| 10:09.48 | Strom_C | how about a word that's pronounced "anyone" but is spelled "catsex" |
| 10:10.40 | porche | :) |
| 10:10.51 | creativx | you said sex! |
| 10:11.03 | porche | strom got a hang up problem |
| 10:11.14 | porche | i mean hang up detection problem |
| 10:11.21 | Strom_C | ok...why don't you describe the complete problem, please |
| 10:11.27 | porche | sorry |
| 10:11.32 | porche | it was hard to eat and type |
| 10:11.35 | porche | my installation is |
| 10:11.46 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 10:11.47 | porche | asterisk 1.2.8 (though tried 1.4.4 also) |
| 10:11.55 | porche | got a tdm2400p, with echo cancel |
| 10:12.03 | porche | got ordinary phone lines, analog |
| 10:12.11 | porche | the application is simple one |
| 10:12.14 | porche | users call in |
| 10:12.16 | porche | authenticate |
| 10:12.23 | porche | and get into conference |
| 10:12.27 | porche | the issue is |
| 10:12.32 | Strom_C | it would |
| 10:12.33 | Strom_C | help |
| 10:12.33 | porche | when they hang up the line |
| 10:12.34 | Strom_C | if you |
| 10:12.36 | Strom_C | didnt |
| 10:12.37 | Strom_C | press |
| 10:12.38 | Strom_C | enter |
| 10:12.39 | porche | asterisk cannot detect |
| 10:12.41 | Strom_C | every three |
| 10:12.42 | Strom_C | words |
| 10:12.55 | creativx | o |
| 10:12.55 | creativx | r |
| 10:12.56 | creativx | ly |
| 10:12.56 | porche | ok sure strom, i will do it |
| 10:13.52 | porche | summary, got asterisk 1.2.8 + digium tdm2400 + analog lines, users call in, a simple meetme application, then hang up when they are finished, asterisk cannot detect the hangup |
| 10:14.15 | Strom_C | is the telco providing disconnect supervision? |
| 10:14.19 | porche | i tried hanguponpolaritychange, but no use, since my telco seems that they do not support it |
| 10:14.37 | porche | is disconnect supervision same as hanguponpolarity? |
| 10:14.45 | Strom_C | not necessarily |
| 10:15.06 | Strom_C | usually, the telco will disconnect talk battery for about half a second once the other party hangs up |
| 10:15.11 | porche | i am using fxs_ks, according to the docs, the best one |
| 10:16.02 | porche | well i asked the telco tech guys, they said, actually they have no idea about it, but weakly they said they do not support it |
| 10:16.39 | Strom_C | well, if they can't either provide a polarity reversal or a talk battery drop, then there's simply no way you can detect when the other party hangs up. |
| 10:16.42 | porche | can busydetect or any other tool help to detect? cuz on my telco side, i do have busy signal after disconnect, some busy signal about 10 times |
| 10:17.04 | Strom_C | how many analog lines do you have? |
| 10:17.07 | porche | though, i also tried it, but still cannot detect |
| 10:17.11 | porche | currently 12 |
| 10:17.20 | Strom_C | might be better to get a partial ISDN30e |
| 10:17.37 | Strom_C | or a half dozen ISDN2e :) |
| 10:18.05 | porche | :) i see, then there is no way if they do not change the polarity |
| 10:18.19 | porche | is there a parameter for to detect polarity change |
| 10:18.39 | porche | i only see the zapata.conf hanguponpolarity=yes or no, |
| 10:18.51 | Strom_C | why don't you test the line first and find out what it's doing -- seems more sensible than toggling options all day |
| 10:19.08 | porche | how can I test it btw? |
| 10:19.21 | Strom_C | with a test set? an analog phone? |
| 10:19.40 | porche | ah, lights must go off test you mean |
| 10:19.48 | Strom_C | what? |
| 10:20.05 | porche | i mean, on disconnect, the analog phone's lights must go off? |
| 10:20.33 | Strom_C | ok then |
| 10:20.49 | *** join/#asterisk porche (n=porche@88.239.82.203) |
| 10:21.00 | porche | sorry line is down strom |
| 10:21.23 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 10:21.31 | porche | k let's see |
| 10:21.35 | ghenry | When do you have to type *1 to record a call? |
| 10:21.41 | ghenry | when it's ringing, when it's answered? |
| 10:22.05 | Strom_C | porche: if those lights are powered by the phone line, then that's one way to test it |
| 10:22.23 | Strom_C | but usually you just /listen/ for the battery drop and/or polarity reversal |
| 10:26.47 | porche | i see |
| 10:27.04 | porche | best to find a test tool then |
| 10:27.25 | porche | so you say, busydetect is not an alternative on analog lines |
| 10:28.05 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 10:28.31 | Strom_C | no, you didn't listen to me :) |
| 10:28.35 | Strom_C | you don't need a test tool |
| 10:28.52 | Strom_C | just listen to the line; you'll hear the battery drop or the polarity reversal if there is one |
| 10:29.01 | porche | no nothing |
| 10:29.02 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 10:29.10 | Strom_C | and busydetect is only for call setup, not for call teardown |
| 10:29.14 | jmls | morning all! |
| 10:29.40 | Strom_C | porche: well, then I would suggest you seriously consider ISDN |
| 10:29.50 | jacq | hey wahst a good way to load balance calls to asterisk boxes from SER... ex: dns based, ... |
| 10:32.27 | porche | there is a a small tick |
| 10:32.32 | porche | on close, but i am not sure |
| 10:32.43 | porche | if it's a polarity reversal on close |
| 10:32.55 | porche | yes isdn would be better, but i am stuck with this card |
| 10:33.37 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 10:34.25 | porche | if there is polatiry change |
| 10:34.31 | porche | is the a special config to make it work |
| 10:35.02 | Strom_M | I /believe/ the option is called hanguponpolarityswitch, but don't quote me on that |
| 10:35.23 | porche | ok thanks a lot |
| 10:41.05 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 10:41.49 | MrChimpy | hi guys |
| 10:43.24 | MrChimpy | anyone had problems building zaptel-1.4.2.1? I'm getting a gcc: mxmldoc.o: No such file or directory when it's building the new menuconfig stuff. |
| 10:45.32 | Zeeek | is it possible to have 1.2 and 1.4 on the same machine and be able to start either one anytime without re installing? |
| 10:46.32 | Strom_M | why the hell would you want to do that? |
| 10:46.46 | Zeeek | to test 1.4 over the weekend |
| 10:47.18 | Zeeek | IOW, what is the fastest switchover technique? |
| 10:47.23 | porche | mychimpy |
| 10:47.36 | porche | have you compiled once, and recompiling now? |
| 10:47.53 | MrChimpy | yep |
| 10:48.02 | porche | try with a refresh copy |
| 10:48.05 | porche | it will pass |
| 10:48.08 | MrChimpy | yeah, just about to |
| 10:48.23 | MrChimpy | ok, so it doesn't like being built twice |
| 10:48.43 | porche | yeah, i met too, probably something stuck, |
| 10:49.21 | MrChimpy | thanks porche |
| 10:50.57 | porche | sure, np |
| 10:51.17 | porche | but, i downgraded to 1.2.8 now, seems faster with mysql, just a small note |
| 10:51.23 | n0n4m3 | exten => ${FWDNUMBER},3,Set(CALLERID(number)=9${CALLERID(number)}) |
| 10:51.24 | n0n4m3 | bummer |
| 10:51.29 | n0n4m3 | this doesn't work |
| 10:51.30 | n0n4m3 | :( |
| 10:52.27 | Zeeek | no |
| 10:54.00 | Strom_M | how about (num) instead of (number) |
| 10:54.16 | Strom_M | and also, ${FWDNUMBER} isnt a valid extension name |
| 10:54.22 | Zeeek | unfortunate wording of headline: http://www.wsoctv.com/mlb/13222064/detail.html |
| 10:54.28 | n0n4m3 | why not? |
| 10:54.45 | n0n4m3 | FWDNUMBER=853914 ; your calling number |
| 10:55.11 | *** join/#asterisk saftsack (n=saftsack@pD9E0741B.dip.t-dialin.net) |
| 10:55.22 | Strom_M | the syntax parser doesn't use variables in extension names |
| 10:55.44 | n0n4m3 | you sure about that? |
| 10:55.51 | n0n4m3 | it seems to work nevertheless :D |
| 10:56.18 | Zeeek | yes variable are parsed in extensions |
| 10:56.59 | Zeeek | or at least they have been for the last 3 years or so |
| 10:57.37 | n0n4m3 | what sip client are you guys using? |
| 10:57.45 | n0n4m3 | i have problems with eyebeam on vista |
| 10:57.47 | Zeeek | xlite |
| 10:58.02 | n0n4m3 | that's the same as eyebeam |
| 10:58.16 | Zeeek | yeah, it is. But I don't use vista |
| 10:58.59 | n0n4m3 | mode=quietmp3 |
| 10:59.00 | *** join/#asterisk skyphyr (n=alanj@135.196.58.222) |
| 10:59.00 | n0n4m3 | grr |
| 10:59.07 | n0n4m3 | this doesn't seem 'quiet' at all :S |
| 10:59.12 | n0n4m3 | the on hold music :( |
| 10:59.35 | Zeeek | use sox to make it quiet |
| 11:00.17 | n0n4m3 | umm |
| 11:00.20 | n0n4m3 | on the fly? |
| 11:00.49 | Zeeek | no, once |
| 11:02.15 | cy303 | any of you guys using RAGI or Adhearison? |
| 11:02.20 | n0n4m3 | when i listen it on an mp3 player it seems okay |
| 11:03.22 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 11:05.07 | *** join/#asterisk Foxygnu (n=FoX@2001:41d0:1:44c8:1337:cafe:d00d:6942) |
| 11:06.16 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 11:11.06 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 11:12.14 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:13.37 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:23.51 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 11:25.02 | Polis_ttt | hi, does anyone have a clue how i put my asterisk-server behind nat, and then can connect to it when i'm outside, via wan-port. Does it work if i just forwarding port 5060 and makes some changes in sip.conf? |
| 11:26.40 | jacq | ? nat |
| 11:26.44 | Zeeek | Polis_ttt this is a pretty common question, google asterisk nat and see voip-info.org |
| 11:29.00 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 11:29.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 11:31.22 | mocker | Yay, it's Friday! |
| 11:31.45 | Zeeek | yes, that's the day of the Asterisk Users Conference! |
| 11:31.55 | mocker | Ahh crap. |
| 11:32.42 | Zeeek | no matter, it's recorded |
| 11:33.18 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:33.37 | mocker | There should be a MeetMe bridge. :) |
| 11:33.52 | Zeeek | there is when Russell sets it up |
| 11:35.00 | mocker | OpenSuSE is really annoying. |
| 11:35.07 | Zeeek | why? |
| 11:35.18 | mocker | Zeeek: I hate yast. ;) |
| 11:35.33 | Zeeek | Yet Another Shitty Terminal? |
| 11:35.51 | mocker | Yet Another Setup Tool |
| 11:36.11 | mocker | aka - I take forever to install simple packages |
| 11:37.04 | e-ddie | that's why the distro comes with a windoze size installation by default |
| 11:37.23 | e-ddie | as they hope you wont realize how crappy their package tools are |
| 11:37.36 | Zeeek | here we go :) |
| 11:37.46 | e-ddie | not really |
| 11:37.57 | mocker | Yeah, we're all in agreement that it sucks. |
| 11:37.58 | mocker | :) |
| 11:38.01 | Zeeek | heh |
| 11:38.21 | Zeeek | I kept getting Suze CD in magazines. That's why I never tried it. |
| 11:38.29 | creativx | nothing beats MSI packages anyways. |
| 11:38.32 | Zeeek | Like AOL in mailbox |
| 11:39.58 | *** join/#asterisk NirS (i=Nir@87.68.1.231.cable.012.net.il) |
| 11:40.40 | mocker | NagiosGrapher is awesome. (on a totally different subject) |
| 11:40.55 | Zeeek | by all means, remain positive :) |
| 11:41.17 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 11:41.29 | e-ddie | just like photos |
| 11:41.30 | mocker | heh. |
| 11:41.51 | e-ddie | if we're all positive, it'll be a boring world |
| 11:42.07 | e-ddie | more boring, that is |
| 11:42.14 | mocker | e-ddie: Except nobody would say that. |
| 11:42.15 | Zeeek | entropy would become b0rken |
| 11:42.18 | mocker | Becuase it's negative. |
| 11:42.37 | e-ddie | i would :) |
| 11:44.16 | mocker | Zeeek: Is there a page about the developer conference today? |
| 11:44.22 | e-ddie | as i dont have any need to not say things others wont |
| 11:44.34 | Zeeek | http://x2z.eu |
| 11:48.00 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 11:51.40 | *** join/#asterisk davidcsi (n=davidcsi@82.158.35.53.static.user.ono.com) |
| 11:52.31 | davidcsi | question: When calling SIP->SIP if the caller hangs up BEFORE the called party answers, the called phone keeps ringing... anyony know why this is and how to solve it?? |
| 11:56.17 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-116-242.ph.ph.cox.net) |
| 11:57.43 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 11:59.12 | *** join/#asterisk funnymanva (n=funnyman@12.171.153.130) |
| 11:59.31 | funnymanva | Does anyone know if Ranch Networks is out of business? They're main numbers don't work anymore. |
| 12:00.37 | coppice | they went the way of the dodo |
| 12:00.51 | coppice | maybe ranches just didn't buy enough networks |
| 12:02.35 | funnymanva | Do you know that for sure or when they went out? Does anyone know of any other netsec firewalls? |
| 12:04.08 | coppice | a couple of months ago, I guess. time flies. it might be longer |
| 12:04.29 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:06.53 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 12:09.38 | n0n4m3 | does anyone happen to have a pocketpc with a sip phone? |
| 12:10.00 | porche | yeah tried, eats battery non4m3 |
| 12:10.13 | n0n4m3 | which one do you use? i'm trying to use sjphone on my qtek 9090 but guess it uses too much cpu power :( |
| 12:10.25 | porche | isnt it 400Mhz? |
| 12:10.35 | n0n4m3 | i belive it is... |
| 12:10.44 | porche | yeah must be enough |
| 12:10.55 | porche | one sec, i couldnt remember |
| 12:10.57 | n0n4m3 | Intel PXA263 400 MHz processor |
| 12:11.17 | porche | yeah must be enough, but you are after a portable go with wi-fi ones |
| 12:12.27 | n0n4m3 | i'm after what? |
| 12:12.52 | porche | i mean if you like to have a wireless phone, wi-fi iphones are better |
| 12:16.15 | porche | n0n4m3 stanaphone i believe |
| 12:16.20 | porche | i tried |
| 12:16.22 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 12:21.40 | *** part/#asterisk samlt (n=sam@fla93-1-81-57-168-33.fbx.proxad.net) |
| 12:21.43 | jkiff | G'morning, #asterisk. For some reason, when one of my agents presses *, it disconnects the call and I'm baffled as to why. Disconnect is set to *0 in features.conf. Any ideas? |
| 12:24.05 | n0n4m3 | stanaphone uses sjphone |
| 12:24.06 | n0n4m3 | :S |
| 12:25.50 | porche | n0n, didnt you ask sip? |
| 12:26.02 | *** join/#asterisk ukris (n=ukris@p4004-ipad75marunouchi.tokyo.ocn.ne.jp) |
| 12:27.25 | *** join/#asterisk kkeil (n=kkeil@p54978F9E.dip0.t-ipconnect.de) |
| 12:28.06 | *** join/#asterisk sundarr (n=sundarr@122.167.67.147) |
| 12:32.38 | *** join/#asterisk adr3nalin3 (n=Foobar@65-123-189-218.dia.static.qwest.net) |
| 12:35.54 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 12:44.32 | De_Mon | !paste |
| 12:44.37 | De_Mon | ?paste |
| 12:44.43 | De_Mon | !@#$ |
| 12:44.48 | De_Mon | ??paste |
| 12:46.36 | Corydon76-home | ~pb |
| 12:46.59 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 12:47.00 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:47.00 | creativx | ~ping |
| 12:47.20 | jbot | pong |
| 12:47.22 | De_Mon | 04:51PM <@Corydon76-home> I don't see where you ever got a j option for AddQueueMember anyway. It was never there. |
| 12:47.27 | De_Mon | http://pastebin.ca/527801 |
| 12:47.49 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 12:48.47 | *** join/#asterisk jrenzema (n=josh@h51bafc3a.c46-01-01.dyn.perspektivbredband.net) |
| 12:50.04 | L|NUX | is there any way to get local SIP Call on IAX Client ? |
| 12:50.14 | *** join/#asterisk voipnet-tech (n=voipnet-@rrcs-24-97-250-50.nys.biz.rr.com) |
| 12:50.26 | voipnet-tech | morning |
| 12:50.28 | De_Mon | L|NUX huh? |
| 12:50.39 | De_Mon | L|NUX use a PBX? |
| 12:50.49 | L|NUX | De_Mon : well i am using Asterisk |
| 12:51.00 | jrenzema | Hi. If anyone is interested in a consulting project, I have a project posted at rent-a-coder for an Asterisk Outbound IVR with web interface at http://www.rentacoder.com/RentACoder/misc/BidRequests/ShowBidRequest.asp?lngBidRequestId=696561 |
| 12:51.05 | L|NUX | De_Mon : all i want when call come on SIP it will ring to my IAX Softphone |
| 12:52.23 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 12:54.07 | creativx | re-route it then L|NUX |
| 12:54.10 | [TK]D-Fender | L|NUX: You just don't seem to get it. You set * up to register to all of your providers. It will speak whatever protocol they need. You then set up your soft-phone to use whichever protocol IT uses. * will sit in the MIDDLE of the call and translate each end of the call. |
| 12:54.20 | [TK]D-Fender | L|NUX: * is a B2BUA |
| 12:54.23 | [TK]D-Fender | ~b2bua |
| 12:54.35 | jbot | b2bua is probably a back 2 back user agent |
| 12:55.04 | IPmonger | yes |
| 12:55.09 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 12:55.13 | L|NUX | [TK]D-Fender : well its not working |
| 12:55.19 | L|NUX | [TK]D-Fender : when i try to call a DID |
| 12:55.34 | L|NUX | and i am register on IAX Client which suppose to ring |
| 12:55.36 | L|NUX | i got error |
| 12:56.19 | De_Mon | Corydon76-home line 161 -- http://svn.digium.com/view/asterisk/branches/1.2/apps/app_queue.c?annotate=65389 |
| 12:56.27 | De_Mon | still in 1.2 branch |
| 12:56.37 | L|NUX | <PROTECTED> |
| 12:56.38 | L|NUX | May 31 07:43:48 NOTICE[17252]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 12:56.52 | [TK]D-Fender | L|NUX: Can * answer the call and Playback something to the caller before trying to pass the call to another device? |
| 12:57.20 | [TK]D-Fender | L|NUX: pastebin your sip.confmasking ONLY passwords, and your dialplan (seperately) |
| 12:57.29 | L|NUX | [TK]D-Fender : okay |
| 12:57.37 | *** join/#asterisk iulius_ (n=iulius@mail1.technologieshq.com) |
| 12:57.39 | L|NUX | [TK]D-Fender : i am using realtime |
| 12:57.45 | L|NUX | sip_buddies |
| 12:57.58 | [TK]D-Fender | L|NUX: Pastebin your extensions.conf AND your realtime dumped |
| 12:58.05 | L|NUX | okies |
| 13:00.53 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:00.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:04.07 | L|NUX | [TK]D-Fender: what do you mean by realtime dumped, you need database ? |
| 13:04.22 | [TK]D-Fender | L|NUX: Clearly. |
| 13:04.27 | L|NUX | ok |
| 13:04.50 | [TK]D-Fender | L|NUX: And please make another pastebin of the COMPLETE call from beginning to end |
| 13:05.19 | L|NUX | ok |
| 13:06.24 | [TK]D-Fender | L|NUX: And while you're at it, your IAX.CONF masiking only passwords as well. |
| 13:06.50 | L|NUX | nah |
| 13:07.03 | L|NUX | for IAX i am using same sip_buddies table |
| 13:07.04 | [TK]D-Fender | L|NUX: Keep in mind the BIG PRINT HERE : L|NUX> -- Executing Dial("SIP/XXX.XXX.XXX.XXX-09a23b30", "SIP/14193017227") in new stack |
| 13:07.19 | [TK]D-Fender | L|NUX: Where the hell are you using IAX in this dial line? |
| 13:07.36 | L|NUX | [TK]D-Fender : this is what i wanted to know |
| 13:07.51 | [TK]D-Fender | L|NUX: Don't bothre with the rest then |
| 13:08.09 | [TK]D-Fender | L|NUX: If this is supposed to dial an IAX device, then you sir are on CRACK |
| 13:08.23 | L|NUX | ok |
| 13:08.23 | creativx | bad crack at it |
| 13:08.28 | L|NUX | so its means there is no way right |
| 13:08.49 | [TK]D-Fender | L|NUX: Why the @^#%#@ hell does it say Dial(SIP<----------- |
| 13:09.06 | [TK]D-Fender | L|NUX: You put that there. You are not TELLING it to ring a an IAX device. |
| 13:09.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.24 | L|NUX | [TK]D-Fender : okay |
| 13:09.48 | L|NUX | [TK]D-Fender : but if we are using AGI how can AGI know that user is connected with IAX Softphone or Phone |
| 13:09.48 | n0n4m3 | http://rula.net/13 |
| 13:09.51 | n0n4m3 | any ideas? |
| 13:09.52 | n0n4m3 | o_O |
| 13:10.30 | [TK]D-Fender | L|NUX: try saying that more clearly ... the last part doesn't add up.. |
| 13:10.54 | [TK]D-Fender | n0n4m3: * does not support G.723 natively. |
| 13:11.05 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:11.09 | Katty | weeee |
| 13:11.15 | cpm | waa? |
| 13:11.20 | L|NUX | [TK]D-Fender : the point it |
| 13:11.21 | Katty | woooo |
| 13:11.22 | L|NUX | is |
| 13:11.27 | [TK]D-Fender | Katty: http://www.albinoblacksheep.com/flash/weeee.php |
| 13:11.33 | n0n4m3 | [TK]D-Fender so is there any way to fix this? |
| 13:11.34 | Katty | oh dear. what is this. |
| 13:11.44 | cpm | aaah |
| 13:11.56 | [TK]D-Fender | n0n4m3: Yeah... tell whoever it is sending you the call to use another codec |
| 13:12.07 | [TK]D-Fender | Katty: Gonads! .... and Strife! |
| 13:12.18 | coppice | one more year, and G.723.1 will be FREEEEEEE! |
| 13:12.21 | n0n4m3 | [TK]D-Fender that won't be possible |
| 13:12.28 | [TK]D-Fender | n0n4m3: TFB |
| 13:13.02 | n0n4m3 | so i'm toast :( |
| 13:13.06 | *** join/#asterisk Brandon_W (n=Brandon_@machine76.Level3.com) |
| 13:13.30 | [TK]D-Fender | n0n4m3: Buttered Cinnamon Toast |
| 13:13.35 | Katty | [TK]D-Fender: ^_- |
| 13:13.57 | [TK]D-Fender | Katty: Oh dear... just LOOK at the mess you've made... |
| 13:14.22 | Katty | i'm not even awake yet :< |
| 13:14.24 | Katty | stop picking on me |
| 13:14.54 | drako | configure a Digium B410P seem to be impossssible on Debian |
| 13:15.13 | Katty | nothing's impossible on debian |
| 13:15.40 | [TK]D-Fender | drako: Yeah... Debian has all sorts of text file editors available for it! |
| 13:15.50 | Katty | drako: i prefer emacs. |
| 13:16.10 | drako | [TK]D-Fender, compatibility |
| 13:16.15 | [TK]D-Fender | drako: vi,vim,nano,pico,gedit,kedit,kwrite,mc jsut to name a few! |
| 13:16.40 | [TK]D-Fender | drako: Allso not applicable. Go download the latest tarballs and get to it. |
| 13:16.45 | drako | [TK]D-Fender, digium says that card wont work on kernel higher than 2.6.16 |
| 13:17.00 | drako | Debian etch comes with 2.6.18 and downgrade brake a lot of things |
| 13:17.02 | [TK]D-Fender | drako: And don't let me hear you whine about compiling when your packages are BROKEN. |
| 13:17.04 | drako | including udev |
| 13:20.17 | [TK]D-Fender | L|NUX: .... and your point is .... ? |
| 13:20.41 | De_Mon | since when did downgrading a kernel 'break a lot of things' ? |
| 13:21.03 | Katty | [TK]D-Fender: you /are/ being nice this morning, right? |
| 13:21.06 | Katty | [TK]D-Fender: RIGHT?! |
| 13:21.08 | De_Mon | especially when its TWO MINOR REVISIONS |
| 13:21.14 | drako | De_Mon, since there are many changes like they are from 2.6.17 to 2.6.18 |
| 13:21.22 | L|NUX | [TK]D-Fender : well now we told dev to add option to user if he have choose IAX2/SIP for incomnig then it will ring to IAX2/SIP accordingly :) |
| 13:21.27 | L|NUX | [TK]D-Fender : thanks for your help mate |
| 13:21.29 | [TK]D-Fender | Katty: Not being particularly snarky... just look what I'm responding to:) L|NUX>[TK]D-Fender : the point it L|NUX>is |
| 13:21.41 | Katty | [TK]D-Fender: good. |
| 13:21.46 | De_Mon | drako show me a website that agrees with you |
| 13:21.52 | Katty | [TK]D-Fender: cause i dun serve snarky muffins. |
| 13:21.55 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:21.58 | L|NUX | [TK]D-Fender : nice m00d :) |
| 13:22.07 | drako | De_Mon, read changelog and figures. |
| 13:22.18 | Katty | [TK]D-Fender: oh, i'm moving. |
| 13:22.18 | drako | after 2.6.16 or 2.6.17 one of these |
| 13:22.27 | [TK]D-Fender | L|NUX: Glad to hear you figured out what to do to fix your approach. |
| 13:22.30 | Katty | [TK]D-Fender: you need to help get all my furniture down two flights of stairs. |
| 13:22.43 | creativx | wait, is it friday? |
| 13:22.47 | Katty | yes, yes it is |
| 13:22.54 | creativx | SWEET. |
| 13:22.59 | cy303 | @@@@@@@@ @@@@@@@ @@@ @@@@@@@ @@@@@@ @@@ @@@ |
| 13:22.59 | cy303 | @@@@@@@@ @@@@@@@@ @@@ @@@@@@@@ @@@@@@@@ @@@ @@@ |
| 13:22.59 | cy303 | @@! @@! @@@ @@! @@! @@@ @@! @@@ @@! !@@ |
| 13:23.00 | cy303 | !@! !@! @!@ !@! !@! @!@ !@! @!@ !@! @!! |
| 13:23.00 | cy303 | @!!!:! @!@!!@! !!@ @!@ !@! @!@!@!@! !@!@! |
| 13:23.00 | [TK]D-Fender | Katty: I moved one of my best friends GF 3 times in under a year..... |
| 13:23.00 | L|NUX | [TK]D-Fender : well i have fixed that in this morning but i was instructed to test it again and again on SIP |
| 13:23.02 | cy303 | !!!!!: !!@!@! !!! !@! !!! !!!@!!!! @!!! |
| 13:23.03 | Katty | i also have a funeral today. |
| 13:23.04 | cy303 | !!: !!: :!! !!: !!: !!! !!: !!! !!: |
| 13:23.07 | cy303 | :!: :!: !:! :!: :!: !:! :!: !:! :!: |
| 13:23.09 | cy303 | <PROTECTED> |
| 13:23.12 | cy303 | <PROTECTED> |
| 13:23.14 | [TK]D-Fender | OPS ! |
| 13:23.27 | Katty | how ascii |
| 13:23.51 | [TK]D-Fender | Katty: ASCII stupid question, get a stupid ANSI :) |
| 13:24.07 | Katty | <PROTECTED> |
| 13:24.10 | Katty | err /golfclap |
| 13:24.14 | creativx | /hum |
| 13:24.22 | drako | <PROTECTED> |
| 13:24.23 | Katty | i want my netherdrake |
| 13:24.27 | creativx | /yes |
| 13:24.28 | Katty | stupid lack of 5000g |
| 13:25.03 | Katty | [TK]D-Fender: but yes, i'm moving. |
| 13:25.19 | [TK]D-Fender | Katty: Bigger. Better. MORE! |
| 13:25.20 | Katty | [TK]D-Fender: 3 bookshelves, a computer desk i have to take apart to get outside the door...the kitchen table |
| 13:25.29 | Katty | [TK]D-Fender: not to mention my entire bedroom suite |
| 13:25.39 | Corydon76-home | cy303: don't do that again |
| 13:25.45 | [TK]D-Fender | tzanger: "That wasn't herbal tea, that was HERB!" |
| 13:25.55 | tzanger | [TK]D-Fender: yep |
| 13:25.58 | tzanger | that was a great show |
| 13:26.01 | tzanger | that and cheers |
| 13:26.08 | Katty | what about mash? |
| 13:26.09 | *** join/#asterisk guille1983 (n=chatzill@190.73.188.118) |
| 13:26.12 | [TK]D-Fender | tzanger: "Our Father who is Art in heaven" ;) |
| 13:26.12 | Katty | i liked mash. |
| 13:26.17 | tzanger | I never got in to mash |
| 13:26.21 | Katty | ah. |
| 13:26.22 | tzanger | [TK]D-Fender: heh |
| 13:26.23 | guille1983 | hi, is Message Waiting Indication available with SIP ? |
| 13:26.26 | Katty | and star trek |
| 13:26.31 | [TK]D-Fender | guille1983: Yes |
| 13:26.34 | tzanger | Hi, I'm Harry. But then again, aren't we all? |
| 13:26.34 | Katty | star trek TOS |
| 13:26.41 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:26.41 | Katty | oh, speaking of harry |
| 13:26.43 | guille1983 | [TK]D-Fender: is that new ? |
| 13:26.48 | Katty | there's a harry potter theme park opening in flordia |
| 13:26.52 | Katty | in, uhh, 2009 i think |
| 13:26.52 | tzanger | oh god |
| 13:27.06 | Corydon76-home | tzanger: yes, my son? |
| 13:27.08 | [TK]D-Fender | Katty: I. Am a Graduate. Of. The James.T.Kirk. School of..... OVERACTING! |
| 13:27.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:27.13 | [TK]D-Fender | guille1983: No |
| 13:27.20 | Katty | [TK]D-Fender: teehee. |
| 13:27.21 | tzanger | Corydon76-home: great, now I have the Mclean & Mclean skit in my head |
| 13:27.22 | coppice | Katty: will the locals be holding witch trials? |
| 13:27.24 | Katty | [TK]D-Fender: yes! |
| 13:27.30 | Katty | coppice: probably. my mother is |
| 13:27.34 | Katty | coppice: WITCHCRAFT! |
| 13:27.35 | L|NUX | humm |
| 13:27.42 | guille1983 | [TK]D-Fender: is Name Identification available on SIP protocol as well ? |
| 13:28.15 | [TK]D-Fender | guille1983: Well SIP sends a CID name & number, so I think the answer to that is "yes" |
| 13:28.24 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 13:28.57 | guille1983 | [TK]D-Fender: is there any additional service that h.232 has and SIP doesnt? |
| 13:29.09 | guille1983 | h.323 i meant |
| 13:29.47 | *** part/#asterisk porche (n=porche@88.239.82.203) |
| 13:30.44 | Katty | [TK]D-Fender: i have sip registrations (sip.conf) in two different contexts (upstairs and downstairs). When i go to extensions.conf and setup [upstairs] and [downstairs]... |
| 13:30.53 | [TK]D-Fender | guille1983: Nothing relevent that I'm aware of. |
| 13:31.12 | guille1983 | [TK]D-Fender: thank you very much sir |
| 13:31.15 | Katty | [TK]D-Fender: in the _xxx,1,Dial($EXTEN}) part it works. but I'm guessing the voicemail bit i need to specify which context it's going to. |
| 13:31.16 | [TK]D-Fender | Katty: that does not parse... |
| 13:31.30 | Katty | ksec |
| 13:31.47 | [TK]D-Fender | Katty: Yes, definately, and XXX is not a "nice" way for this. You should hard-code them. |
| 13:32.03 | guille1983 | [TK]D-Fender: any link that I can check to see how to configure additional services on SIP protocol? |
| 13:32.17 | Katty | ok so, under [downstairs] do I need _xxx,1,Dial(${EXTEN}@downstairs,20)? |
| 13:32.23 | Katty | ^- [TK]D-Fender |
| 13:32.23 | *** join/#asterisk jrenzema (n=josh@h51bafc3a.c46-01-01.dyn.perspektivbredband.net) |
| 13:32.24 | [TK]D-Fender | guille1983: Normally there IS nothing to "configure". |
| 13:32.54 | guille1983 | [TK]D-Fender: so they are available once I run a SIP network? :S |
| 13:32.59 | [TK]D-Fender | guille1983: read the sample sip.conf , and read up on "presence" for subscriptions on the WIKI |
| 13:33.01 | [TK]D-Fender | ~wikis |
| 13:33.06 | jbot | it has been said that wikis is http://www.voip-info.org |
| 13:33.33 | Katty | [TK]D-Fender: which i presume would be as simple as [upstairs] 124,1,Dial(124@downstairs,20) |
| 13:33.39 | Katty | with the sip thingy in there |
| 13:33.49 | *** join/#asterisk bintut (n=bintut@cm112.gamma181.maxonline.com.sg) |
| 13:33.53 | [TK]D-Fender | Katty: There is no tech in that line. |
| 13:34.04 | Katty | [TK]D-Fender: tech? |
| 13:34.10 | bintut | anyone here knows a sip client for palm os 5.x? |
| 13:34.14 | jacq | SIP/ |
| 13:34.20 | [TK]D-Fender | Katty: And generally... ICK. that also means that ANYBODY can dial that phone downstairs |
| 13:34.29 | Katty | yes. |
| 13:34.38 | [TK]D-Fender | Katty: tech = SIP, IAX2, H323,ZAP,LOCAL.... |
| 13:34.40 | Katty | she's the receptionist |
| 13:34.47 | Katty | [TK]D-Fender: aye, i typed that right under my line i goofed up |
| 13:34.47 | *** join/#asterisk appletizer (n=erktjgek@62-30-203-36.cable.ubr04.hawk.blueyonder.co.uk) |
| 13:35.00 | appletizer | how easy is it to set up asterisk on a linux based server please? |
| 13:35.01 | Katty | i want everyone to be able to call the receptionist :P |
| 13:35.18 | [TK]D-Fender | Katty: [upstairs] exten => 124,1,Gotol(downstairs,124,1) |
| 13:35.31 | [TK]D-Fender | Katty: [upstairs] exten => 124,1,Goto(downstairs,124,1) |
| 13:35.31 | Katty | [TK]D-Fender: but there is no 124 in the downstairs context. |
| 13:35.35 | Katty | [TK]D-Fender: it's all _xxx |
| 13:35.44 | [TK]D-Fender | Katty: it will pattern-match. |
| 13:35.49 | Katty | [TK]D-Fender: oh. |
| 13:35.55 | Katty | [TK]D-Fender: ooooooooooh |
| 13:36.26 | appletizer | and just wondering, does Asterisk come with a web-based administration panel? |
| 13:36.36 | [TK]D-Fender | appletizer: No. |
| 13:36.42 | appletizer | [TK]D-Fender ah |
| 13:36.44 | Katty | [TK]D-Fender: sweet ^_^ |
| 13:36.45 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:36.45 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:37.08 | anthm | w00t |
| 13:37.14 | [TK]D-Fender | appletizer: There are SEPERATE products out there for that though, but be warned, your soul should be worth more than what they offer... |
| 13:37.21 | Katty | anthm: don't let anyone call me till at least 10 :< |
| 13:37.25 | appletizer | [TK]D-Fender, lol |
| 13:37.31 | appletizer | [TK]D-Fender, any recommendations for that please? |
| 13:37.44 | [TK]D-Fender | appletizer: Was I not clear enough? :) |
| 13:38.08 | Katty | [TK]D-Fender: i'm gonna pastebin what i've got so far. |
| 13:38.08 | appletizer | [TK]D-Fender, lol yeah still i'm curious as to how much they're worth in the region of |
| 13:38.13 | appletizer | :P |
| 13:38.18 | anthm | did someone call you early? |
| 13:38.25 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-387g9ui.cable.mindspring.com) |
| 13:38.28 | *** join/#asterisk Vec (n=Vec@dsl-244-210-176.telkomadsl.co.za) |
| 13:38.39 | [TK]D-Fender | appletizer: What do you expect out of said interface? |
| 13:38.47 | VJFROMGT | someone is asking me if i am in GK or GW mode, ,, what doe that mean? |
| 13:38.48 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:39.02 | De_Mon | drako I read the 2.6.18 changelog and don't see anything that would make me think downgrading to an earlier kernel would beak something |
| 13:39.05 | De_Mon | http://packages.debian.org/changelogs/pool/main/l/linux-2.6/linux-2.6_2.6.18.dfsg.1-12etch2/changelog |
| 13:39.15 | [TK]D-Fender | appletizer: Tell you what.... go download Trixbox. Play around with it. If you like it and need help with it, this isn't the place. If you don't like it head on back. |
| 13:39.26 | appletizer | [TK]D-Fender, just a simple centralised area from which to change configuration scripts, it need not be fanciful... it can even be *gack* a simple text-based editor which access the conf files |
| 13:39.45 | appletizer | [TK]D-Fender, excellent |
| 13:39.47 | appletizer | thanks |
| 13:39.48 | [TK]D-Fender | appletizer: SSH <- |
| 13:39.49 | creativx | appletizer: putty -> vi -> free |
| 13:40.05 | creativx | it even comes in black and white if you wish. |
| 13:40.10 | appletizer | lol |
| 13:40.11 | appletizer | ha ha :) |
| 13:40.22 | appletizer | i'm just wondering if there are ready-made solutions |
| 13:40.28 | appletizer | or whether i have to actually script one myself |
| 13:40.35 | appletizer | but it's good to know, there's something like Trixbox |
| 13:40.59 | creativx | well |
| 13:41.26 | creativx | either the gui is for admin or it is for call mangling |
| 13:41.42 | appletizer | well yeah some of the admins aren't so tech savvy... |
| 13:41.49 | appletizer | hence the requirement for a web-based admin panel :\ |
| 13:41.50 | appletizer | blah |
| 13:42.16 | creativx | but then again asterisk can be tech hell |
| 13:43.07 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:45.47 | [TK]D-Fender | creativx: This isn't "The Simpsons"... you don't put a MORON in charge of a nuclear reactor.... |
| 13:46.29 | creativx | indeed. there's no fancy buttons with colors to push. |
| 13:47.07 | dlynes | VJFROMGT: gatekeeper vs gateway mode, i would imagine |
| 13:47.17 | dlynes | VJFROMGT: You're talking for H.323, right? |
| 13:50.15 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:51.59 | VJFROMGT | SIP |
| 13:52.01 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 13:52.02 | zeeesh | hi |
| 13:52.11 | VJFROMGT | dlynes,, SIP |
| 13:52.23 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:52.45 | VJFROMGT | they are also telling me that my equipment is reporting FAS |
| 13:54.08 | bintut | what are the usual reasons why calls are disconnected with this kind of routes: x-lite <--> fw <--> internet <--> pbx <--> dial 9+pstn_no <--> pstn <--> pbx <--> sip_hardphone |
| 13:58.26 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-99.ph.ph.cox.net) |
| 13:59.45 | NirS | VJ ,the terms GateKeeper, Gateway Mode and FAS are mostly common with H323 |
| 13:59.45 | NirS | are you you sure you are talking about SIP ? |
| 13:59.58 | rue_mohr | bintut, have you done a quality test on the route over the internet? |
| 14:00.05 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 14:00.20 | VJFROMGT | yes |
| 14:02.03 | NirS | weird |
| 14:02.22 | Mercestes | bintut: The call was done so one of the users hung up |
| 14:02.23 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:02.38 | guille1983 | question: Do i need to have a asterisk server in all my company's facilities ? |
| 14:02.55 | Mercestes | guille1983, Yes. Absolutely |
| 14:03.13 | guille1983 | oh ok |
| 14:03.13 | Mercestes | I suggest the business edition with full digium support contracts\ |
| 14:03.36 | [TK]D-Fender | ~mercestes |
| 14:03.45 | jbot | mercestes is definitely a total nub |
| 14:03.58 | Mercestes | ... |
| 14:04.03 | guille1983 | ? |
| 14:04.04 | Mercestes | f.u. fender |
| 14:04.20 | [TK]D-Fender | guille1983: Take everything he says with a grain of salt, a pinch of pepper, and a dash of paprika (for colour) |
| 14:04.20 | Mercestes | guille1983, Maybe you coudl try to tell us what yoru trying to do so we can actually answer your question with a slight degree of accuracy. |
| 14:04.22 | guille1983 | :D what's it all about ? :D |
| 14:04.36 | rue_mohr | bintut, use netperf to do a few tests on the connection over the internet |
| 14:04.37 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-119-99.ph.ph.cox.net) |
| 14:04.39 | Mercestes | It's about D-Fender riding more ass than Mary at a Christmas derby. |
| 14:04.59 | [TK]D-Fender | guille1983: No, you don't necessarily need an * server at each site, but there are bandwidth and independence issue to consider |
| 14:05.11 | [TK]D-Fender | Mercestes: Yee haw! |
| 14:05.44 | guille1983 | ok, I am analyzing a wan network around the country, And I am thinking about the best deployment |
| 14:06.02 | Mercestes | guille1983, How many phones at each site? |
| 14:06.28 | guille1983 | Mercestes: more than 40 less than 300 |
| 14:06.38 | rue_mohr | heh |
| 14:06.45 | Mercestes | guille1983, In that case, my answer was probably correct. |
| 14:07.12 | guille1983 | Mercestes: so that I do a well usage of bandwidth ? |
| 14:07.15 | Mercestes | Otherwise you will have to establish a full network path *twice* for every time someone decides to call someone internally |
| 14:07.21 | Mercestes | Precisely |
| 14:08.04 | guille1983 | Mercestes: so, when a user wants to call to another city its * server will conect him to the another city's * server ? |
| 14:08.20 | Mercestes | I've run about 1000 users all remote from the Asterisk server but.....it's not at all fun and I had some commercial phone switches to help me |
| 14:08.47 | Mercestes | guille1983, If an employee in New York calls an extension in california, then you can use IAX2 to route it internally via the Internet. |
| 14:08.50 | *** join/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 14:08.54 | *** part/#asterisk b1shop (n=b1shop@dsl081-149-253.chi1.dsl.speakeasy.net) |
| 14:09.19 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 14:09.24 | bintut | rue_mohr: i think this is nothing to do with the network.. |
| 14:09.30 | guille1983 | Mercestes: I am studying the case they have a frame relay network |
| 14:09.37 | Mercestes | If an employee calls someone outside of *your* asterisk network, then likely not, unless you set them up a direct sip or iax2 connection with them. |
| 14:10.26 | bintut | Mercestes: actually, i was the callee having the sip hardphone.. |
| 14:11.15 | Mercestes | guille1983, You could do it with fewer servers, but I think your *best* case is a pbx at each office. |
| 14:11.27 | *** part/#asterisk appletizer (n=erktjgek@62-30-203-36.cable.ubr04.hawk.blueyonder.co.uk) |
| 14:11.30 | Mercestes | That way if someone unplugs it you loose one office instead of several. |
| 14:11.35 | guille1983 | Mercestes: they already have a pbx on each city |
| 14:11.44 | Mercestes | asterisk pbx |
| 14:12.01 | guille1983 | Mercestes: so, you first answer was right :D |
| 14:12.04 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 14:12.06 | Mercestes | If it's sevreal offices in a city you can do it that way too. on an internal LAN you should ahve enough bandwidth to support that. |
| 14:12.41 | Mercestes | I wouldn't say "right" it's mostly an opinion on my part and yours based upon several factors and variables. |
| 14:13.12 | guille1983 | yeah right, that was not wrong neither |
| 14:13.28 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 14:17.08 | Mercestes | Would you say most of yoru traffic is internal traffic or external traffic? |
| 14:18.32 | guille1983 | Mercestes: Let me ask the admin network |
| 14:18.40 | rue_mohr | bintut, until you know, test the hell out of it |
| 14:18.51 | ManxPower | Must. Resist. Clicking. SEND. Before. First. Cup. Of. Coffee |
| 14:19.53 | guille1983 | Mercestes: what is the difference between they use more internal traffic or external traffic ? |
| 14:20.34 | skyphyr | hi all - can anyone recommend a UK based provider for DID and PSTN outbound that I can use for business running our own asterisk server? |
| 14:20.50 | skyphyr | well not necessarily UK based, but providing UK (London) numbers |
| 14:20.52 | Mercestes | guille1983, If 80% of your calls are outbound outside of your network, then you wouldn't need a PBX at each location so much because most of yoru calls would be leaving your entire network anyways. |
| 14:21.08 | Mercestes | If 80% of yoru calls are interoffice then you would want a PBX at each location... |
| 14:21.40 | Mercestes | If 80% of yoru calls are interoffice at remote office locations, then you might be able to do regional PBXs instead of a PBX at each site. |
| 14:21.44 | Mercestes | It's a matter of traffic analysis. |
| 14:22.07 | guille1983 | Mercestes: when you say external do you mean calls to the PSTN ? |
| 14:22.44 | ManxPower | I didn't resist, |
| 14:22.50 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 14:23.05 | Mercestes | guille1983, Precisely |
| 14:23.32 | Mercestes | ManxPower, :( Better luck next time, lad. |
| 14:25.24 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 14:26.02 | guille1983 | Mercestes: well, i thought that (maybe i am wrong, remember i am a newbee) they connect phones to the asterisk server and the server was connected to their PBX system so when users want to call to the pstn they dial a central extension and if they want to call to another office (wether or not they are on the same city) they had to dial another extension first and when they got the tone they... |
| 14:26.03 | guille1983 | ...dial the number |
| 14:26.59 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 14:27.02 | Mercestes | Asterisk is a PBX. |
| 14:27.23 | Mercestes | tryign to interface Asterisk with an existing (redundant) PBX is a monster in and of itself that will result in much pain. |
| 14:28.06 | *** part/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 14:28.08 | Mercestes | It's like trying to make your '07 Corvette remote control by hooking a remote control car to it and rigging your tires to the steering wheel and pedals |
| 14:28.11 | guille1983 | so my asterisk server will have a interface for my incomming E1 lines? |
| 14:28.20 | Mercestes | If you buy a card for it, yes. |
| 14:28.28 | [TK]D-Fender | guille1983: You just want a basic VoIP bridge between remote PBX's so they can dial between themselves for free? |
| 14:28.46 | guille1983 | [TK]D-Fender: yes |
| 14:29.02 | [TK]D-Fender | guille1983: How many simultaneous calls per site do you need? |
| 14:29.11 | guille1983 | [TK]D-Fender: making usage of their framerelay network of course |
| 14:29.27 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:29.37 | [TK]D-Fender | guille1983: Wrong answer try again. how many simultaneous calls do you need to be able to support at each site? |
| 14:31.05 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 14:31.11 | eeos | hi everybody |
| 14:31.13 | guille1983 | [TK]D-Fender: ok, number of calls at the same time is not calculated yet, i got the formula but i havent calculated yet, besides, it would be different in each site |
| 14:32.01 | guille1983 | [TK]D-Fender: I am talking about like 30 cities network divided in 3 regions |
| 14:32.53 | [TK]D-Fender | guille1983: harware needed at each site will vary based on their needs and what their existing PBX offers by way of connectivity. |
| 14:33.44 | [TK]D-Fender | guille1983: Yes, * can do the job, and based on bandwidth & independence requirements an * server at each sit may be advisable. |
| 14:34.36 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 14:34.52 | guille1983 | [TK]D-Fender: oh ok, they only have in each city a number of E1 lines connected to their old PBX system and phones are connected to it, thanks for your help guys |
| 14:34.54 | jmls | does anyone know where I can buy an IDSN E1 crossover cable in the UK ? |
| 14:35.50 | Uatec | most of the time Digium support are really fast |
| 14:35.57 | Uatec | but i emailed them yesterday, and NO reponse |
| 14:35.57 | Uatec | :( |
| 14:36.28 | *** join/#asterisk blaylock (n=sfv100@c-24-30-250-200.hsd1.va.comcast.net) |
| 14:36.55 | blaylock | is there an on hold notification (such as a beep or tone) available in asterisk? |
| 14:36.57 | [TK]D-Fender | jmls: just MAKE one. |
| 14:37.03 | jmls | Uatec: maybe your email was never delivered ? |
| 14:37.07 | blaylock | to notify me that someone is still on hold? |
| 14:37.22 | jmls | [TK]D-Fender: I have done a couple of times, but |
| 14:37.35 | jmls | A) the quality is not that good |
| 14:37.48 | jmls | B) it takes too much time (banana fingers) |
| 14:37.49 | *** join/#asterisk toerkeium (i=oo@dcc-hq-host-200-59-45-53.dnsba.com) |
| 14:37.56 | [TK]D-Fender | jmls: SAD |
| 14:37.59 | jmls | very |
| 14:38.16 | jmls | my legs are old and bent |
| 14:38.43 | mog | Uatec, whats the prob bob |
| 14:38.56 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 14:39.45 | ManxPower | guille1983: Frame Relay and VoIP are very, very, very hard to make work. |
| 14:40.04 | guille1983 | ManxPower: yes, i've read that |
| 14:40.24 | ManxPower | If your CIR is not the same as your port speed, basically you can't do QoS for real time apps. |
| 14:41.33 | guille1983 | mmm |
| 14:42.32 | ManxPower | The problem is by the time the frame relay newtwork sends the congestion info to you, it is already too late. |
| 14:42.49 | guille1983 | jesus! |
| 14:42.52 | ManxPower | If your port speed and CIR are the same, you should never get that sort of congestion message back from the frame relay network, |
| 14:43.15 | ManxPower | guille1983: Cisco's web site has some information on QoS on FR |
| 14:43.23 | guille1983 | what port speed are you talking about? i am sorry |
| 14:43.57 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 14:48.48 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 14:48.53 | msetim | Hi guys, |
| 14:49.15 | msetim | what the order that asterisk follow to play moh when the mode is not aleatory |
| 14:49.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 14:51.24 | msetim | somebody know? |
| 14:51.40 | tbic | is there any way to get the IP address of the user that is making the call in AGI |
| 14:53.25 | [TK]D-Fender | tbic: Look at the current channel, then dump it. |
| 14:54.28 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 14:54.31 | tbic | ok, how would I dump the current channel though an AGI command? |
| 14:55.07 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 14:57.16 | blaylock | anyone know of a way to do on hold notification? |
| 14:57.27 | blaylock | like ring the phone every few seconds? |
| 14:57.37 | [TK]D-Fender | tbic: You don't. Connect through AMI for that. |
| 14:57.47 | blaylock | kind of the same as MWI maybe |
| 14:58.11 | [TK]D-Fender | blaylock: You can't. If you want the phone to remind you, you'd better hope your phone offers you that feature |
| 14:58.38 | tbic | [TK]D-Fender: thanks, I'll try that |
| 14:59.04 | *** join/#asterisk PierreY (n=Pierre@125-202.206-83.static-ip.oleane.fr) |
| 14:59.18 | PierreY | hi all |
| 14:59.38 | *** join/#asterisk tdi (n=tdi@gvf90.internetdsl.tpnet.pl) |
| 14:59.42 | tdi | hi all ! |
| 15:00.04 | blaylock | [TK]D-Fender, hmmm, probably have to hack some code to get it to work somewhat like mwi or call waiting |
| 15:00.13 | PierreY | I want to write a driver for Firebird SQL backend. Can somebody tell my where to start ? |
| 15:00.29 | [TK]D-Fender | blaylock: What kind of phones are you using? |
| 15:00.54 | blaylock | well at the moment i dont know, my boss was asking me if its possible |
| 15:01.02 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:01.11 | [TK]D-Fender | blaylock: How the hell do you not know what kind of phones you have? |
| 15:01.36 | blaylock | it for a customer, which he didnt get the type of phones from |
| 15:01.42 | [TK]D-Fender | blaylock: And there is no SIP mechanism for what you want. |
| 15:01.45 | tdi | does anybody knows any usable sip phone dor osx ? this xlite crashes on intel mac |
| 15:02.24 | blaylock | [TK]D-Fender, thats what i was looking for, but there doesnt seem to be any SIP method other than MWI |
| 15:02.28 | [TK]D-Fender | blaylock: At absolute best you'd have to do a massive rewrite of code to KILL the call after a certain period of time and try calling back until connected. A massive task with pathetic ROI |
| 15:03.07 | blaylock | [TK]D-Fender, heh probably so...maybe it will be easier to tell him no :-D |
| 15:03.15 | [TK]D-Fender | blaylock: Definately. |
| 15:03.49 | [TK]D-Fender | tdi: check the WIKI or JFGI |
| 15:03.51 | blaylock | [TK]D-Fender, but the thing is, since there is mwi, why couldnt I apply that function to a call thats on hold? |
| 15:04.01 | blaylock | [TK]D-Fender, is asterisk not aware of a call thats on hold? |
| 15:04.07 | [TK]D-Fender | blaylock: MASSIVE code changes..... |
| 15:04.10 | [TK]D-Fender | ~wglwat |
| 15:04.22 | jbot | methinks wglwat is well, good luck with all that |
| 15:04.24 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 15:04.24 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 15:05.12 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 15:06.24 | blaylock | [TK]D-Fender, cool then, thanks for your help |
| 15:06.35 | [TK]D-Fender | blaylock: No so much |
| 15:06.45 | blaylock | heh |
| 15:06.47 | [TK]D-Fender | "help" as "healthy advise" |
| 15:07.01 | tdi | [TK]D-Fender: i did not find any _good_ softphones there, i would not come here and ask without searching there before.. |
| 15:07.03 | blaylock | [TK]D-Fender, gave me an anser that I "wanted" to hear |
| 15:07.05 | blaylock | hah |
| 15:07.06 | tdi | forget it |
| 15:07.29 | [TK]D-Fender | tdi: Well... any on the WIKI you HAVEN'T tried? |
| 15:07.36 | [TK]D-Fender | tdi: try them next :) |
| 15:07.52 | [TK]D-Fender | tdi: Not so many MacOSX users here at any given time. |
| 15:07.54 | *** join/#asterisk bbryant (i=brett@nat/digium/x-6f21df01e84a7d33) |
| 15:10.13 | guille1983 | what Call Parking/Pickup service consists on ? |
| 15:10.28 | neverblue | morning |
| 15:11.53 | *** join/#asterisk saftsack (n=saftsack@pD9E0741B.dip.t-dialin.net) |
| 15:12.18 | [TK]D-Fender | guille1983: ... |
| 15:12.19 | [TK]D-Fender | ~book |
| 15:12.23 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:12.23 | [TK]D-Fender | ~wikis |
| 15:12.25 | jbot | i guess wikis is http://www.voip-info.org |
| 15:12.42 | guille1983 | [TK]D-Fender: thanks (Y) |
| 15:13.09 | Katty | if i want to dial multiple phones, and give it a context, does it look like exten => s,2,Dial(SIP/100@context&SIP101@context)etc? |
| 15:13.13 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:13.59 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:14.26 | VJFROMGT | anyone know hot o link 2 boxes via sip? |
| 15:16.03 | [TK]D-Fender | VJFROMGT: http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
| 15:16.23 | [TK]D-Fender | Katty: there is not such thing as "give it a context" |
| 15:16.39 | Katty | how does an incoming call ring a blast group then? |
| 15:16.40 | [TK]D-Fender | Katty: A SIP device is a SIP device. |
| 15:17.02 | Katty | if i have 101 in context a and 101 in context b, it can't ring /101/ |
| 15:17.14 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 15:17.29 | [TK]D-Fender | Katty: that is NOT a SIP device then, that is a LOCAL CHANNEL |
| 15:17.41 | blitzrage | Local channels rock my socks |
| 15:19.11 | VJFROMGT | tk,, that show me how to do via iax2 |
| 15:19.16 | VJFROMGT | i wanna do by sip |
| 15:20.10 | [TK]D-Fender | VJFROMGT: it shows BOTH. Wake up and READ |
| 15:20.15 | Katty | [TK]D-Fender: does the zap get given an associated context? |
| 15:20.24 | Katty | [TK]D-Fender: i just don't get how it knows which SIPs to ring. |
| 15:20.38 | [TK]D-Fender | Katty: You are asking about random bits that don't belong in the same sentence. |
| 15:20.39 | Katty | [TK]D-Fender: especially if SIP/100 could be in two different contexts. |
| 15:20.41 | Mercestes | Katty: If you have 101 in context a and 101 in context b you cannot ring both 101's in context a and b at the same time. |
| 15:20.50 | Katty | Mercestes: yes, i know this. |
| 15:20.52 | blitzrage | VJFROMGT: it's a bit of a pain in the ass -- I documented it in TFoT 2nd Edition though (which is unfortunately not out yet) |
| 15:20.53 | Katty | Mercestes: how do i tell it which one? |
| 15:21.03 | Mercestes | you have one zap channel coming in? |
| 15:21.14 | Katty | Mercestes: several, lumped into a group |
| 15:21.17 | Mercestes | s/channel/trunk/ |
| 15:21.18 | [TK]D-Fender | Mercestes: Sure you can. |
| 15:21.27 | Mercestes | [TK]D-Fender, demonstrate |
| 15:21.31 | VJFROMGT | i am using iax at this time but once there is a slight fault, trunk goes down until reboot |
| 15:21.41 | Katty | Mercestes: let's say 5 lines are lumped into my [zap-katty] group |
| 15:21.51 | Katty | Mercestes: which, rings a blast group. |
| 15:21.57 | [TK]D-Fender | Mercestes: Dial(Local/101@contexta&Local/101@contextb) |
| 15:22.07 | [TK]D-Fender | *sigh* |
| 15:22.29 | Katty | Mercestes: the blast group is 100, 101, and 102. |
| 15:22.32 | Mercestes | [TK]D-Fender, Didn't you just say you can't do that? |
| 15:22.40 | Katty | Mercestes: but 100, 101, and 102 are in both context a and b |
| 15:22.47 | [TK]D-Fender | Mercestes: No, I didn't. |
| 15:22.56 | *** part/#asterisk teyus (i=Mateus@unaffiliated/teyus) |
| 15:23.08 | Mercestes | [TK]D-Fender, your english fails you then |
| 15:23.35 | [TK]D-Fender | Mercestes: No, your head is on completely backwards and are unable to keep yourself in context with the conversation. |
| 15:23.39 | Mercestes | and I thought it was Sip/device@domain |
| 15:23.43 | Katty | Mercestes: in my old dialplan i had SIP/100&SIP/101&SIP/102 |
| 15:23.56 | Katty | Mercestes: but that won't work now. |
| 15:23.58 | Mercestes | [TK]D-Fender, And you are too busy nit-picking verbage to provide anything useful. :P |
| 15:24.07 | [TK]D-Fender | Mercestes: Katty is mixing up SIP phones an EXTENSIONS.CONF contexts in her question. |
| 15:24.21 | [TK]D-Fender | Mercestes: Her question was poor and you fell for it. |
| 15:24.28 | Mercestes | [TK]D-Fender, that sentence doesn't even make sense. |
| 15:24.29 | [TK]D-Fender | Mercestes: Nub ;) |
| 15:24.35 | Mercestes | 'tard. |
| 15:24.35 | bintut | gtg now.. thanks all.. :) |
| 15:24.36 | Katty | [TK]D-Fender: her question is POOR because she's asking for help. |
| 15:24.39 | Katty | [TK]D-Fender: you nitwit :P |
| 15:24.58 | Mercestes | I'm looking it up now. |
| 15:25.03 | Katty | i dunno how to explain what i'm wanting in any other way than above. |
| 15:26.56 | [TK]D-Fender | Katty: You need to really buckle down on understanding the SEPERATE bits tha make up *. You have all that is sip.conf and extensions.conf jumbled up in your head :) |
| 15:27.00 | blitzrage | Katty: I missed the question I think.... |
| 15:27.10 | Mercestes | Katty, Ok, do you have say, a 101 in both context a and b? |
| 15:27.26 | blitzrage | or rather, what is attempted to be accomplished |
| 15:28.01 | [TK]D-Fender | Mercestes: Yes, she does, and ringing both simultaneously appears to be what she wishes to do. |
| 15:28.07 | *** join/#asterisk nasls_lsa (n=chatzill@ppp078-212.dsl.hol.gr) |
| 15:28.08 | Katty | [TK]D-Fender: No |
| 15:28.17 | Katty | Mercestes: i'll query you |
| 15:28.20 | [TK]D-Fender | Katty: then you are SUPER jumbled then :) |
| 15:28.24 | Katty | [TK]D-Fender: no i'm not |
| 15:28.34 | Katty | [TK]D-Fender: i think iknow how to do it. i'm just asking for confirmation |
| 15:29.14 | *** join/#asterisk angryuser (n=Miranda@df01t2-213-44-92-3.d4.club-internet.fr) |
| 15:29.38 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 15:33.20 | angryuser | how to configure asterisk to apply different dialplans depending on time? for example on the diner time go to messagebox automaticly |
| 15:33.20 | [TK]D-Fender | angryuser: "show application gotoiftime" |
| 15:33.20 | angryuser | ok |
| 15:33.20 | angryuser | reading |
| 15:33.20 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 15:33.21 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 15:34.30 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 15:34.47 | Zeeek | I'll listen to the bhme business radio show for ideas :) |
| 15:41.26 | *** join/#asterisk ploieel (n=manni@Fb325.f.ppp-pool.de) |
| 15:43.15 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 15:43.16 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:47.05 | *** join/#asterisk [1]climber_ (n=climber_@212.6.249.99) |
| 15:47.16 | [1]climber_ | hi |
| 15:47.54 | [1]climber_ | i just have this problem |
| 15:47.55 | [1]climber_ | s isn't the appropriate place to get help. As the topic says, try #asterisk |
| 15:47.55 | [1]climber_ | [17:33] Disconnected (2007-06-01 17:33:38) |
| 15:48.03 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 15:48.05 | [1]climber_ | shit |
| 15:48.15 | Qwell[] | <climber_> http://rafb.net/p/xPrZe125.html |
| 15:48.45 | [1]climber_ | thx |
| 15:48.58 | [1]climber_ | today it is bad day |
| 15:51.21 | Zeeek | so, another guy on your trail? |
| 15:52.03 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com) |
| 15:52.50 | angryuser | [TK]D-Fender: gotoiftime, if time mtches then it is executed, if not script continue? |
| 15:53.10 | *** join/#asterisk gardo (n=gardo@121.97.197.103) |
| 15:53.57 | angryuser | [TK]D-Fender: it is written, that nothing is done when time does no match, i want to know what "nothing" means, does it jumps to next priority? |
| 15:54.30 | Zeeek | there is no such thing as "nothing is done" |
| 15:54.30 | De_Mon | I'm setting up dynamic queues for a bunch of staff members. When I send someone to voicemail, how could I determine if I should use a busy or unavailable greeting? |
| 15:55.07 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:55.31 | *** join/#asterisk dasuberdavid (i=david@nat/digium/x-d0fc8eaf5a6006c3) |
| 15:55.53 | angryuser | Zeeek: it is written in cli output;) |
| 15:56.22 | angryuser | This application will set the context, extension, and priority in the channel structure |
| 15:56.22 | angryuser | if the current time matches the given time specification. Otherwise, nothing is done. cli output |
| 15:56.51 | *** join/#asterisk tris- (i=tristan@camel.ethereal.net) |
| 15:56.51 | *** join/#asterisk Beirdo_ (n=gjhurlbu@unaffiliated/beirdo) |
| 15:57.04 | *** join/#asterisk znoG_ (n=gs@235-180-235-201.fibertel.com.ar) |
| 15:58.34 | *** join/#asterisk mocker (n=mocker@198.247.173.227) [NETSPLIT VICTIM] |
| 15:58.35 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 16:01.26 | angryuser | i found my answer anyway;) |
| 16:01.27 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 16:01.42 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 16:01.42 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-520ac24412e9d329) [NETSPLIT VICTIM] |
| 16:01.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) [NETSPLIT VICTIM] |
| 16:01.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:01.58 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) [NETSPLIT VICTIM] |
| 16:02.41 | Mercestes | angryuser, What it means by "nothign is done' is that it does not perform the specified "goto" and instead continues on to the next priority, yes. |
| 16:02.42 | *** join/#asterisk funnymanva (n=funnyman@12.171.153.2) |
| 16:04.59 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-224-149.dhcp.insightbb.com) [NETSPLIT VICTIM] |
| 16:04.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 16:04.59 | *** join/#asterisk ltd (n=z@nox.amused.net) [NETSPLIT VICTIM] |
| 16:04.59 | *** join/#asterisk Corydon76-home (i=one@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM] |
| 16:04.59 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) [NETSPLIT VICTIM] |
| 16:04.59 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 16:04.59 | *** join/#asterisk MindTheGap (n=iote@c91197b8.bhz.virtua.com.br) [NETSPLIT VICTIM] |
| 16:04.59 | *** join/#asterisk osiris250 (i=b3ts6o0f@bsd02.evansengineering.net) [NETSPLIT VICTIM] |
| 16:04.59 | *** join/#asterisk antonyo14 (n=oper@206.135.21.162) |
| 16:05.00 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) [NETSPLIT VICTIM] |
| 16:05.00 | antonyo14 | has anyone run across this error |
| 16:05.00 | antonyo14 | Registration from '<sip:25@10.4.0.201>' failed for '10.4.0.149' - Not a local domain |
| 16:05.00 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) [NETSPLIT VICTIM] |
| 16:05.00 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) [NETSPLIT VICTIM] |
| 16:05.00 | De_Mon | ah hah, if nobody is in the queue they are "busy" |
| 16:05.25 | De_Mon | now I just need a way to remove all members from a queue. |
| 16:05.28 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) [NETSPLIT VICTIM] |
| 16:07.26 | *** join/#asterisk johann8384 (n=johann83@gateway.myogre.com) [NETSPLIT VICTIM] |
| 16:07.42 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 16:08.36 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 16:09.02 | Zeeek | russellb are you there? |
| 16:10.09 | *** join/#asterisk bmd (n=bmd@72.54.252.34) [NETSPLIT VICTIM] |
| 16:10.21 | De_Mon | Im not seeing a way to remove all queue members from a queue |
| 16:11.31 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 16:11.52 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 16:12.03 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 16:12.12 | elg | in 1.4.4, I execute ChanSpy and the console reports that it's spying, and volume changes when I press #, and changing which channel I'm spying with *, but I hear absolutely nothing. no announcements, nothing |
| 16:12.20 | elg | am I doing something wrong? |
| 16:12.27 | file | Zeeek: poke |
| 16:12.36 | Zeeek | peek |
| 16:13.24 | file | Zeeek: russellb just called me... he got held up with an appointment at his house so he'll be in the office a little after 11:30... but he'll call in when he gets there |
| 16:13.41 | Zeeek | that shopuld be good |
| 16:13.51 | file | he is in his car now driving |
| 16:13.54 | file | vroom vroom |
| 16:13.55 | Zeeek | the SIP channel wll be up in a few |
| 16:14.07 | Zeeek | what no cell to call in to meetme? :) |
| 16:14.20 | file | we don't want Russell to drive off the road... |
| 16:14.27 | Zeeek | no we don't |
| 16:14.36 | Zeeek | but in Huntsville there's only one road |
| 16:15.08 | file | ...yeah... |
| 16:15.26 | casimir | antonyo14, I've not seen it before, but you might want to check allowexternaldomains in sip.conf |
| 16:15.31 | Zeeek | What is it? Jimmy Walker Drive? |
| 16:15.34 | killfill | hey |
| 16:15.51 | killfill | Channel 0/1, span 1 got hangup request Zap/1-1 is circuit-busy.. why would i get that? |
| 16:16.06 | killfill | i get it sometimes.. and zap channels are obviously not busy... |
| 16:16.21 | *** join/#asterisk ber111 (i=brad@neu.cow.org) [NETSPLIT VICTIM] |
| 16:18.42 | Zeeek | SIP channel is open |
| 16:19.01 | *** part/#asterisk PierreY (n=Pierre@125-202.206-83.static-ip.oleane.fr) |
| 16:19.40 | killfill | Zeeek: what do you mean? |
| 16:19.45 | Zeeek | that was about the asterisk users conf |
| 16:20.00 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 16:20.23 | Zeeek | if Russel doesn't kill himself on the road |
| 16:20.23 | blebleble | where can i set asterisk to only listen on a single ip and not bound to 0.0.0.0? |
| 16:20.37 | casimir | blebleble, sip.conf |
| 16:20.57 | casimir | blebleble, bindaddr |
| 16:20.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:21.35 | blebleble | casimir: outstanding thanks |
| 16:22.25 | *** join/#asterisk ectospasm (n=ectospas@adsl-146-241-62.mob.bellsouth.net) |
| 16:25.37 | Zeeek | ~http://x2z.eu |
| 16:28.34 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 16:29.15 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:29.21 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:29.33 | lirakis | my phones (gxp-2000) do not seem to get sip notify messages to light the MWI. Is there some thing I have to set in asterisk to send these sip notify messages as soon as voicemail is left? |
| 16:30.51 | guille1983 | [TK]D-Fender: sorry i was on the phone |
| 16:30.52 | [TK]D-Fender | lirakis: You need your SIP entry to indicate which mailbox to check, and thats it |
| 16:31.07 | lirakis | [TK]D-Fender: in sip.conf, or on the phone? |
| 16:31.16 | De_Mon | file project meeting? |
| 16:31.17 | lirakis | [TK]D-Fender: i have it set in sip.conf .. |
| 16:31.49 | lirakis | mailbox=5000@device |
| 16:32.26 | [TK]D-Fender | lirakis: then go check if your voicemail.conf is set up to match |
| 16:32.26 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:32.39 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 16:33.09 | lirakis | [TK]D-Fender: ahh.. voicemail.conf shows context [default] .. so probably should change that so they both match up |
| 16:33.10 | lirakis | ;) |
| 16:33.12 | lirakis | duh |
| 16:33.47 | Zeeek | How's russel? |
| 16:35.07 | lirakis | [TK]D-Fender: hmm when i changed [default] to [device] in voicemail.conf .. i no longer had a voicemail box |
| 16:35.11 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 16:35.16 | BSD_Tech | zeep join #ast-conf |
| 16:35.20 | [TK]D-Fender | lirakis: You piocked the SWRONG person to change |
| 16:35.30 | Qwell[] | type much? :p |
| 16:35.32 | BSD_Tech | I just made it |
| 16:35.35 | Zeeek | #ast-conf |
| 16:35.38 | Qwell[] | oh, umm |
| 16:35.45 | Qwell[] | that's today, isn't it? |
| 16:35.49 | lirakis | [TK]D-Fender: .. i didnt change an extension.. i changed the context in voicemail.conf |
| 16:35.55 | Qwell[] | where is russellb... |
| 16:36.11 | *** join/#asterisk taupin974 (n=taupin97@89.237.79.244) |
| 16:36.21 | [TK]D-Fender | lirakis: that LOSES your boxes |
| 16:36.21 | lirakis | .. i will try making my sip.conf entry for extension 5000 to have mailbox=5000@default ... |
| 16:36.24 | blitzrage | russellb <---- Qwell[] right there |
| 16:37.05 | Qwell[] | Zeeek: has he already dialed in? O.o |
| 16:37.21 | Qwell[] | oic |
| 16:37.22 | Qwell[] | nevermind |
| 16:38.21 | lirakis | [TK]D-Fender: yeah .. im confused... because .. my VM was working fine before.. everything except the MWI light on the phone |
| 16:40.05 | russellb | finally here |
| 16:40.11 | russellb | now i have to find that extension i use to call in ... :( |
| 16:41.01 | blitzrage | hrmmmmmm |
| 16:41.26 | [TK]D-Fender | lirakis: Well if the context doesn't match you get no MWI |
| 16:41.42 | lirakis | [TK]D-Fender: .. okay.. |
| 16:42.38 | Zeeek | http://x2z.eu |
| 16:42.46 | lirakis | ah.. son of a .. i forgot to reload after i changed the sip.conf |
| 16:44.10 | lirakis | now.. can i monitor the messages of more than one extension on a single phone? |
| 16:45.05 | Zeeek | http://kfuq.net/asterisk/cfgs/ |
| 16:46.54 | [TK]D-Fender | lirakis: add multiple "mailbox=" statements |
| 16:47.14 | Qwell[] | russellb: let me know when we're linked up :D |
| 16:47.16 | lirakis | [TK]D-Fender: hmm... interesting i didnt know it was that simple |
| 16:47.41 | russellb | Qwell[]: yeah, i will ... i'm just ... having trouble |
| 16:47.41 | Zeeek | hello russel |
| 16:48.18 | blitzrage | ;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes |
| 16:48.24 | lirakis | wow.. that was.. crazy easy |
| 16:48.31 | blitzrage | lirakis: sip.conf.sample is also handy to read :) |
| 16:48.54 | Zeeek | blitzrage come on in |
| 16:49.05 | blitzrage | Zeeek: I'm at my parents house with no VoIP line.... |
| 16:49.24 | Zeeek | call on phone! |
| 16:49.29 | blitzrage | I'm cheap |
| 16:49.43 | blitzrage | and I need to go get foooooooood sooooooooooon |
| 16:49.51 | blitzrage | I have yet to eat breakfast, and its nearly 1pm |
| 16:50.21 | russellb | Qwell[]: done |
| 16:50.25 | Qwell[] | already on :D |
| 16:51.18 | Zeeek | ok |
| 16:51.19 | Qwell[] | Zeeek: what's the caller count? |
| 16:51.29 | Zeeek | dozen or so visible |
| 16:51.58 | elg | for the record, my chanspy problem was solved by adding option b |
| 16:52.02 | elg | though I don't understand why |
| 16:52.37 | Zeeek | Qwell we can't see the number of anon streamers |
| 16:52.41 | kFuQ | http://www.webhostingtalk.com/showthread.php?t=608936 <--- link for infomart fire |
| 16:53.38 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 16:53.44 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:53.46 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 16:56.04 | angryuser | is it possible to send a person ito voicemail box and do not play a it's standart entry message 'talk after bip' ? i just need the bip ;) |
| 16:57.03 | [TK]D-Fender | angryuser: "show application voicemail" |
| 16:57.25 | De_Mon | angryuser (yes) |
| 16:57.30 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-fb242ac1fbb7ecbe) |
| 16:57.30 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:57.36 | De_Mon | now, how about deleting all members from a queue |
| 16:57.44 | De_Mon | err "removing" |
| 16:57.59 | guille1983 | is it that possible to block some services as callwaiting, voice mail, etc. to some users ? |
| 16:58.22 | angryuser | lazy me, reading again |
| 16:58.24 | angryuser | ;) |
| 16:58.34 | *** join/#asterisk bluedemon (n=belamark@merlintechs.kvinet.com) |
| 16:58.34 | Qwell[] | russellb: you should maybe go over a few of the disadvantages too |
| 16:58.50 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:01.25 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@www.airlinksystems.com) |
| 17:01.35 | BSD_Tech | russel join #ast-conf |
| 17:01.53 | angryuser | what has the upper priority, s extension or exact match of 123,1,whatever? |
| 17:02.12 | angryuser | for incoming calls from isdn |
| 17:02.13 | [TK]D-Fender | guille1983: Yes. Go read THE BOOK. You need to learn the basics of how * works. |
| 17:02.14 | [TK]D-Fender | ~book |
| 17:02.27 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:02.27 | [TK]D-Fender | guille1983: And then set up a test server to experiment with |
| 17:03.02 | Zeeek | Listen to the Asterisk Users COnf live here: |
| 17:03.03 | Zeeek | http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 |
| 17:03.10 | guille1983 | [TK]D-Fender: oh ok, based on your experience, is there any reason to block services to users ? |
| 17:03.10 | CoffeeIV_ | are there versions of the app_rxfax.c and app_txfax.c applications that use spandsp to send / received faxes, that worked for asterisk 1.4 ? I just tried to compile the ones that work with asterisk 1.2 and I got an error |
| 17:03.14 | Zeeek | no login required |
| 17:03.22 | kFuQ | russellb: what about SRTP support? |
| 17:03.28 | bluedemon | Is there any handsets out there that would allow you to display queue stats on the display? |
| 17:04.42 | *** join/#asterisk savaticus (n=chatzill@sta-206-168-96-69.rockynet.com) |
| 17:04.45 | angryuser | tommorow i put the asterisk server in production, yahoooo ;) |
| 17:05.25 | BSD_Tech | blue polycom can show queues |
| 17:05.31 | BSD_Tech | with a xml page |
| 17:05.37 | bluedemon | thx, I will check it out |
| 17:05.57 | savaticus | as cann some aastras that have xml app support |
| 17:06.02 | Zeeek | kFuQ ask your question when he stops |
| 17:06.38 | kFuQ | k |
| 17:06.39 | *** join/#asterisk kodorna (n=root@200.180.183.86) |
| 17:07.09 | De_Mon | what offset/len can I use to get 123 from ${1234} (all but last character) |
| 17:07.17 | CoffeeIV_ | Never mind, I found what I needed in hte spandsp download area -- my eyes were just missing it somehow |
| 17:07.34 | kodorna | dudes... i need some help with T1 configuration please |
| 17:07.37 | De_Mon | Im thinking id have to use math to do len-1 |
| 17:08.59 | kodorna | im getting this "Signalling requested on channel 1 is FXO Loopstart but line is in PRI Signalling signallin" message |
| 17:09.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:10.22 | savaticus | what key gneration algorithm will it use? how does it establish trust or is it just open? |
| 17:10.42 | Zeeek | savaticus go ahead when you can with that |
| 17:10.44 | savaticus | FXO is for an analog line |
| 17:10.48 | BSD_Tech | to much cut out |
| 17:11.23 | kodorna | its just open, im using signalling=fxo_ls |
| 17:11.27 | kodorna | and signalling=fxs_ls |
| 17:12.03 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 17:13.37 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 17:14.14 | msetim | Hi... GSM format is more heavy that wav? |
| 17:15.08 | Strom_M | well, technically, neither format weighs anything |
| 17:15.45 | tbic | with AGI GET DATA can you playback more than one file? |
| 17:16.08 | guille1983 | that message waiting service consists on warning the user if he has new voice mail messages ? |
| 17:16.52 | [TK]D-Fender | kodorna: should likely be"signalling=pri_cpe" |
| 17:17.05 | [TK]D-Fender | guille1983: yes |
| 17:17.14 | guille1983 | [TK]D-Fender: thanks again |
| 17:18.13 | kodorna | [TK]D-Fender: i've got to implement like fxs_ls and fxo_ls unfortunately |
| 17:18.15 | guille1983 | [TK]D-Fender: I guess it is available on SIP protocol, right ? |
| 17:18.25 | threat | crap, I have a very bad cracklin on the line, I updated linux, I have a tdm400p card (1 FXO, 1 FXS) |
| 17:18.29 | [TK]D-Fender | guille1983: Every channel type I can think of. |
| 17:18.41 | *** join/#asterisk saftsack (n=saftsack@pD9E0741B.dip.t-dialin.net) |
| 17:18.52 | threat | I loaded wctdm module, is this correct? what other modules may I need to load? |
| 17:19.00 | [TK]D-Fender | threat: Was if bad before, after or on both ends of the upgrade? |
| 17:19.13 | threat | no, before it was perfect |
| 17:19.19 | De_Mon | what would cauase QUEUESTATUS to return "JOIN/LEAVE UNAVAIL" |
| 17:19.22 | threat | now it is crackly |
| 17:19.23 | [TK]D-Fender | threat: Change your kernel? |
| 17:19.26 | threat | yes |
| 17:19.32 | threat | from 2.6.8 to 2.6.18 |
| 17:19.34 | [TK]D-Fender | threat: Recompile Zaptel |
| 17:19.55 | threat | yep, I did, m-a a-i zaptel-souece |
| 17:20.00 | [TK]D-Fender | De_Mon: I would think thats kind of self-explanitory... |
| 17:20.21 | msetim | Strom_M, my doubt is that recording calls using the GSM format be have heavy if a transcoding was necessary.. |
| 17:20.22 | [TK]D-Fender | threat: Not sure what to tell you then... |
| 17:20.30 | threat | hmmmm |
| 17:20.32 | threat | :( |
| 17:20.38 | *** join/#asterisk putnopvut (n=putnopvu@user-24-214-124-177.knology.net) |
| 17:20.38 | threat | any thingies I can test? |
| 17:20.53 | Strom_M | msetim: you're not making any sense |
| 17:20.53 | kFuQ | it would be cool to have a (features.conf) option to change gain on the fly |
| 17:21.18 | kodorna | <PROTECTED> |
| 17:21.18 | kodorna | Jun 1 14:25:18 ERROR[4455]: chan_zap.c:7050 mkintf: Signalling requested on channel 1 is FXO Loopstart but line is in FXS Loopstart signalling |
| 17:21.21 | kodorna | Jun 1 14:25:18 ERROR[4455]: chan_zap.c:10472 setup_zap: Unable to register channel '1-24' |
| 17:21.24 | kodorna | Jun 1 14:25:18 WARNING[4455]: loader.c:415 __load_resource: chan_zap.so: load_module failed, returning -1 |
| 17:21.27 | kodorna | Jun 1 14:25:18 WARNING[4455]: loader.c:500 load_modules: Loading module chan_zap.so failed! |
| 17:21.30 | kodorna | thats the thing |
| 17:21.46 | *** join/#asterisk troy- (n=troy@206-248-177-177.dsl.teksavvy.com) |
| 17:21.46 | De_Mon | [TK]D-Fender i'm not getting it |
| 17:22.24 | [TK]D-Fender | De_Mon: Meas the queue is set up to kick out callers if there are no members avail on join, or if the all leave. |
| 17:22.25 | De_Mon | I could maybe understand JOINUNAVIL being when monitor-join = strict |
| 17:22.50 | De_Mon | ahh, so they are available when the enter the queue and then go unavailable while holding? |
| 17:22.55 | threat | [TK]D-Fender, should I reboot after compiling and installing the module? |
| 17:23.09 | [TK]D-Fender | kodorna: Your zaptel & zapata do not agree. Go clean them both up and when you're done, try again. When it fails, PASTEBIN them both. |
| 17:23.11 | [TK]D-Fender | ~pb |
| 17:23.23 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 17:23.25 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 17:23.44 | *** join/#asterisk sysdebug (n=sysdebug@200.195.161.164) |
| 17:23.45 | [TK]D-Fender | De_Mon: That is one possibility |
| 17:25.33 | threat | [TK]D-Fender, so wctdm is definitly the right module ? |
| 17:25.54 | [TK]D-Fender | threat: for that card, yes |
| 17:26.44 | threat | [TK]D-Fender, hmmm I think I unload and loaded the module too many times, I am not getting a dial tone now |
| 17:26.53 | msetim | Strom_M, sorry :(. I will try explain again... I have many calls that are recording, and I would like to know what is the lowest and compact format. |
| 17:27.00 | threat | I can hear the button when I press the keypad, but it does nothing |
| 17:27.35 | [TK]D-Fender | msetim: GSM is very small in file size and nominal transcoding weight from G.711/ZAP |
| 17:28.08 | *** join/#asterisk Lann (i=Dewayne@adsl-63-200-88-82.dsl.scrm01.pacbell.net) |
| 17:28.28 | Lann | heyas, i cant find a chan for this specific question. Do any of you know if skypecasts work with skype mobile? |
| 17:28.38 | De_Mon | uhhh |
| 17:28.52 | De_Mon | does anyone use skype in here? |
| 17:29.00 | Lann | dunno |
| 17:29.10 | shido6 | is it just me |
| 17:29.14 | shido6 | or is paypal down right now? |
| 17:29.14 | Lann | i want to join group voice chat with my pda |
| 17:29.31 | Lann | my pda that i will buy if it works heh |
| 17:29.33 | Zeeek | shido6 send me your password and I'll check it |
| 17:29.48 | blitzrage | shido6: try shooting me some money and I'll tell you if I get it |
| 17:29.49 | De_Mon | [TK]D-Fender I see a leavewhenempty queue setting, but not a leavewhenunavail |
| 17:29.53 | shido6 | username is noreally@thisisserious.com password is problematicforebay |
| 17:29.54 | blitzrage | ok... LUNCHTIME!!!! |
| 17:30.07 | Zeeek | go blitzrage |
| 17:30.19 | angryuser | bye everybody, and thank you for people who helped me |
| 17:30.30 | De_Mon | ahhh... |
| 17:30.38 | De_Mon | ok I grok now |
| 17:31.46 | De_Mon | i'll submit a patch that fixes the queues.conf template to describe this clearer, some day ;). |
| 17:33.35 | Zeeek | russellb again thanks |
| 17:33.41 | De_Mon | yes - callers donot leave a queue with no members or only unavailable members |
| 17:33.54 | De_Mon | strict - callers do leave a queue with no members or only unavailable members |
| 17:33.55 | threat | need help |
| 17:33.59 | threat | crackling!!! |
| 17:34.18 | threat | any settings I can teak? |
| 17:34.23 | threat | tweak |
| 17:34.24 | russellb | Zeeek: you're welcome! |
| 17:44.08 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 17:44.08 | *** mode/#asterisk [+o Corydon-w] by ChanServ |
| 17:44.08 | *** join/#asterisk keulin (n=cray@nat30-2-88-160-17-233.fbx.proxad.net) |
| 17:45.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:48.10 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 17:48.15 | msetim | [TK]D-Fender: thanks :-D |
| 17:49.15 | *** join/#asterisk matdon (n=matdon@eagle.bsd.st) |
| 17:49.17 | matdon | hi |
| 17:53.22 | matdon | anyone know of a iax client for windows 98? |
| 17:53.45 | msetim | matdon: you can consult the list http://www.voip-info.org/wiki-Asterisk+IAX+clients |
| 17:56.41 | matdon | nice thanks |
| 17:58.35 | De_Mon | voip-info is your friend |
| 17:58.37 | *** join/#asterisk asymptote (n=weldon@phobos.asee.org) |
| 18:01.07 | asymptote | <PROTECTED> |
| 18:03.49 | kodorna | :q |
| 18:08.22 | *** join/#asterisk funnymanva (n=funnyman@12.171.153.133) |
| 18:10.27 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 18:10.50 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:11.01 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:11.53 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 18:16.52 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 18:17.01 | Dr-Linux | what's difference btween user and peer? :S |
| 18:17.22 | coppice | a user goes to the peer to catch the ferry |
| 18:18.33 | [hC] | one uses drugs, and the ohter pushes them onto you? |
| 18:18.51 | msetim | Dr-Linux: is a biggest question :) Try understand what says here: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?rev=58779&view=log |
| 18:19.26 | msetim | Dr-Linux: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?view=markup |
| 18:20.02 | Dr-Linux | :) |
| 18:20.27 | Dr-Linux | actually outcall program support users but not peers that what they say |
| 18:22.53 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-230.static.twtelecom.net) |
| 18:25.28 | *** join/#asterisk mvanbaak (i=michiel@vanbaak.xs4all.nl) |
| 18:30.46 | *** join/#asterisk b00gz (n=b00gz@d233-124-245.col.wideopenwest.com) |
| 18:30.48 | b00gz | I want to make it so extensions 101,102,103 all use a different outbound route order then 201,202,203 ... Can this be done? |
| 18:31.59 | Mercestes | outbound route order on what? |
| 18:33.56 | blitzrage | b00gz: huh? |
| 18:33.56 | Mercestes | If you mean on a PRI, you can use G instead of g to transpose the outbound order, but incoming calls come top down and normally outbound calls go bottom up, so atleast one of your groups could conflict with incoming calls. |
| 18:34.26 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:34.35 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:34.57 | *** join/#asterisk antonyo14 (n=oper@206.135.21.162) |
| 18:35.54 | jkiff | For some reason, when one of my agents presses *, it disconnects the call and I'm baffled as to why. Disconnect is set to *0 in features.conf. Any ideas? |
| 18:36.57 | Mercestes | Is this for one agent or all agents? |
| 18:37.15 | jkiff | All of them. |
| 18:38.06 | jkiff | We noticed when one of them tried to do an attended transfer with *2, and upon further investigation, the hangup occurs immediately after the *. |
| 18:38.32 | threat | hmmm |
| 18:38.47 | antonyo14 | I just re-did my whole dialplan with *gui, sip users register, but when I dial an extension it goes busy tone can anyone help? |
| 18:38.50 | threat | the crackling seems to happen when I have alot of hard disk drive activity |
| 18:39.00 | threat | is this possible? |
| 18:39.17 | threat | would it be a electromagnetic thingy or an interrupt dealy? |
| 18:39.31 | De_Mon | hrmmm reload app_queue.so will reload queues.conf right |
| 18:39.57 | De_Mon | i set timeout=20 but it's not timing out. |
| 18:40.02 | threat | hi |
| 18:40.17 | threat | gay |
| 18:40.25 | De_Mon | well, it says it timed out but yet, the person in queue still rings |
| 18:41.01 | Mercestes | jkiff, what does the CLi say? |
| 18:41.02 | casimir | antonyo14, can you turn on sip debug in console to see what's happening? |
| 18:41.12 | antonyo14 | casimir: sure |
| 18:41.13 | *** join/#asterisk PaYTaTz_PiNgViN (n=77889789@cpe-76-173-56-41.socal.res.rr.com) |
| 18:41.14 | PaYTaTz_PiNgViN | hi |
| 18:41.17 | PaYTaTz_PiNgViN | anyone here? |
| 18:41.21 | Mercestes | no. |
| 18:41.26 | PaYTaTz_PiNgViN | lmao |
| 18:41.30 | PaYTaTz_PiNgViN | hey Mercestes |
| 18:41.33 | PaYTaTz_PiNgViN | how are you? |
| 18:41.39 | Mercestes | meh. yourself? |
| 18:41.46 | PaYTaTz_PiNgViN | im fine thnx |
| 18:41.50 | Mercestes | ASL? |
| 18:41.55 | PaYTaTz_PiNgViN | LOL |
| 18:41.57 | jkiff | Mercestes: Hold on, let me get you a pastebin. |
| 18:42.16 | PaYTaTz_PiNgViN | Question, ok if i want to not user VoIP and want to use regular phone lines what card do i need? i have 8 Phone Lines |
| 18:42.29 | antonyo14 | i'm gettting No such context 'macro-stdexten' for macro 'stdexten' |
| 18:42.35 | Mercestes | Octo-port FXO board |
| 18:42.38 | Mercestes | or a couple quad ports |
| 18:42.52 | Mercestes | antonyo14, guess what that means. |
| 18:42.56 | PaYTaTz_PiNgViN | any ones model you think i should get thats easy to setup? |
| 18:42.58 | PaYTaTz_PiNgViN | RHINO ? |
| 18:43.01 | jkiff | Mercestes: The "<--" is where * was pressed. |
| 18:43.15 | b00gz | Mercestes, I have 8 locations and I want it so when location 1 dials out it they use a different order of trunks then location 2. |
| 18:43.16 | Mercestes | PaYTaTz_PiNgViN, tdm400p is nice. But, whatever yoru comfortable with, really |
| 18:43.22 | antonyo14 | mercestes: i have looked in my confs and cannot find stdexten anywhere |
| 18:43.25 | casimir | antonyo14, that's the default macro for extension in gui |
| 18:43.36 | Mercestes | antonyo14, Then asterisk is correct, you do not have that macro programmed. |
| 18:43.42 | Mercestes | b00gz, Ok, I take it back, you make no sense. |
| 18:43.47 | jkiff | http://pastebin.ca/526052 |
| 18:43.58 | jkiff | Mercestes: I guess pasting the link would help too. :-P |
| 18:44.08 | casimir | antonyo14, I have *gui configured |
| 18:44.11 | antonyo14 | mercestes, didn't know I had to, so I have to write the macro |
| 18:44.12 | Mercestes | chan_telapathy.so |
| 18:44.16 | b00gz | Mercestes, my provider does not allow Caller ID spoofing, I want it so when a office calls out if shows up as there number and not another stores ... |
| 18:44.23 | Mercestes | antonyo14, it comes with make samples. |
| 18:44.29 | PaYTaTz_PiNgViN | Mercestes THank You |
| 18:44.47 | Mercestes | b00gz, what are you dialing out on? |
| 18:45.02 | b00gz | Mercestes, I have 8 locations all connected to 1 asterisk server |
| 18:45.06 | antonyo14 | casimir, what file do I edit to make it work? |
| 18:45.45 | PaYTaTz_PiNgViN | Mercestes, what do you think about digiums TDM840B ? |
| 18:45.58 | casimir | antonyo14, should be extensions.conf |
| 18:46.07 | Mercestes | PaYTaTz_PiNgViN, I don't think I've used one but it' sprobably a good card. |
| 18:46.12 | Mercestes | All the digium stuff I have has been great. |
| 18:46.25 | Strom_M | I like my TDM844B |
| 18:46.39 | antonyo14 | casimir, do you think you could pastebin that macro ? |
| 18:46.47 | dasuberdavid | digium hardware is fantastic |
| 18:46.55 | PaYTaTz_PiNgViN | so i need 2 x tdm400p cards? |
| 18:47.00 | Mercestes | b00gz, Please don't make me talk like [TK]D-Fender. what are you dialing out on? |
| 18:47.02 | casimir | antonyo14, you bet. one sec |
| 18:47.05 | *** join/#asterisk braker (n=email@bas9-ottawa23-1088837129.dsl.bell.ca) |
| 18:47.11 | Mercestes | Yea, I'm starting to prefer it over Sangoma. |
| 18:47.11 | MrWup | how do you define multiple extensions to go into the same priority? |
| 18:47.18 | Mercestes | Digium has never paniced my kernel |
| 18:47.22 | MrWup | say i wanted _5xx and _6xx to both dial the same thing? |
| 18:47.39 | Mercestes | MrWup, _{5,6}XX,1,Do(something) |
| 18:47.47 | MrWup | thanks =] |
| 18:47.50 | Mercestes | np. :) |
| 18:47.56 | [TK]D-Fender | Mercestes: Let the hate flow through you! |
| 18:47.57 | Strom_M | _[56]XX |
| 18:47.59 | [TK]D-Fender | :D |
| 18:48.00 | b00gz | Mercestes, Aastra SIP Phones. |
| 18:48.06 | Mercestes | oh, thanks Strom. :D |
| 18:48.27 | Mercestes | b00gz, what TECHNOLGY are you LEAVING your Asterisk server on??? |
| 18:48.56 | *** join/#asterisk drrt (n=junior@ppp-static2-140.tis-dialog.ru) |
| 18:49.37 | b00gz | Mercestes, VoIP via SIP? |
| 18:49.37 | Nugget | yow! |
| 18:49.53 | Mercestes | b00gz, There are no sip trunks, so therefore ,there is no order. |
| 18:50.19 | Mercestes | So therefore, you cannot change the outbound order of a non-existant sip trunk no more than you can castrate the left nut of a polkadotted unicorn. |
| 18:50.20 | b00gz | Mercestes, I have 8 SIP Trunks? |
| 18:50.51 | Mercestes | You may have 8 individual sip authentications which outbound to another sip gateway. |
| 18:50.57 | Mercestes | but they are not sip trunks. |
| 18:51.24 | Mercestes | So what you mean is, "I dial out over a series of sip connections, how do I rearrange the order of the sip connectiosn I dial based upon where my call is coming from?" |
| 18:51.33 | casimir | antonyo14, http://pastebin.ca/528916 |
| 18:51.44 | b00gz | Mercestes, yes that is correct |
| 18:51.51 | Mercestes | in which case, you would use different contexts for each originating location, and a different OUTBOUND context for each incoming context, which manually dialed the sip connections in different orders. |
| 18:52.39 | MrWup | hmf |
| 18:52.46 | MrWup | im a bit confused about how to manage dialling out in the UK |
| 18:52.56 | MrWup | as most UK and mobile telephone numbers here are 11 digit |
| 18:52.58 | MrWup | which makes things easy |
| 18:53.00 | Mercestes | so yo uwould have [location-1] Dial(Sip/${exten}@provider1) and [location-2] Dial(SIp/${exten}@provider2) |
| 18:53.02 | MrWup | but sometimes they are 10 digits |
| 18:53.03 | antonyo14 | casimir, and i just insert that in extensions.conf |
| 18:53.06 | MrWup | but less commonly |
| 18:53.10 | jkiff | Mercestes: Did you have a look at my paste? |
| 18:53.19 | MrWup | whats the best way of dealing with this conundrum? |
| 18:53.23 | Mercestes | jkiff, Sorry, you'll have to sip debug. |
| 18:53.26 | MrWup | all i can think of is a timeout after the 10th number |
| 18:53.29 | Mercestes | MrWup, Set(Timeout(digit)) |
| 18:53.45 | jkiff | Mercestes: Ah, I'll do that. |
| 18:53.47 | *** join/#asterisk Greek-Boy (n=g@196.45.144.42) |
| 18:53.56 | b00gz | Mercestes, so you are stating make location 1 dial 9, location 2 dial 8, before they dial the number? |
| 18:54.06 | Mercestes | MrWup, You'll have a 1-3 second wait on 10 digit calls and immediate matches on 11 digit calls. |
| 18:54.07 | casimir | antonyo14, think so, sounds like your context is already looking for it, do a extensions reload and you should be good |
| 18:54.21 | Mercestes | b00gz, no. I'm saying use contexts to make it automatic. |
| 18:54.24 | MrWup | Mercestes, where do i set that timeout? |
| 18:54.38 | Mercestes | MrWup, You can do it in the phone in the digitmap |
| 18:55.03 | MrWup | its more of an asterisk thing you see |
| 18:55.09 | Mercestes | MrWup, Set(Timeout(digit)) is for IVRs but it does the same thing, if you were in an IVR atleast. |
| 18:55.23 | Mercestes | MrWup, well, no, you have to control when the phone sends the number..unless your already *in* asterisk |
| 18:55.41 | MrWup | Mercestes, i dial 9 to get into the outside dialling context |
| 18:55.48 | MrWup | so the phone is already connected to * |
| 18:55.50 | Mercestes | Does it use DISA? |
| 18:56.01 | MrWup | no i opted not to use disa because i needed more functionality |
| 18:56.03 | antonyo14 | casimir, thank you so much! it works |
| 18:56.14 | MrWup | like monitoring and logging the outgoing calls, blocked numbers, speed dial etc |
| 18:56.14 | Mercestes | MrWup, Until you hit send your not in asterisk. |
| 18:56.32 | MrWup | no no i am |
| 18:56.40 | MrWup | i use waitexten |
| 18:56.58 | MrWup | u press 9, get taken to an asterisk context which says "outside line" then it waits for you to key in digits |
| 18:57.23 | Mercestes | Oh... |
| 18:57.34 | Mercestes | then you would use Set(Timeout(digit) before your WaitExten |
| 18:57.41 | MrWup | ah |
| 18:57.43 | Mercestes | and then match both 10 and 11 digit numbers. |
| 18:58.06 | MrWup | Set(Timeout(digit) = 3) |
| 18:58.07 | MrWup | ? |
| 18:58.40 | Mercestes | Yea |
| 18:58.50 | MrWup | thanks again! |
| 18:58.54 | Mercestes | np |
| 19:00.48 | casimir | antonyo14, glad to help, the gui is really nice once you get it working |
| 19:01.27 | antonyo14 | casimir, whoever made the gui are saints :) |
| 19:01.43 | Hmmhesays | la dee da dee da |
| 19:01.45 | antonyo14 | at least for newbs |
| 19:01.53 | antonyo14 | haha |
| 19:02.12 | antonyo14 | hmmhesays, i though you were busting a song for us |
| 19:03.02 | casimir | I'm a bit of a newb myself |
| 19:03.35 | antonyo14 | i guess it's all a matter of perspective... |
| 19:03.39 | casimir | had to come here and get berated by [TK]D-Fender in order to get my sip phones to have 2-way audio :) |
| 19:04.07 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:05.01 | Hmmhesays | lol |
| 19:05.43 | antonyo14 | casimir, [TK]D-Fender has put the dunce cap on me several times :) |
| 19:06.45 | [TK]D-Fender | nah... I just have a really big mirror, and am not afraid to use it ;) |
| 19:07.18 | antonyo14 | haha |
| 19:07.55 | *** join/#asterisk saftsack (n=saftsack@pD9E0741B.dip.t-dialin.net) |
| 19:08.03 | *** join/#asterisk Prato (n=Prato@dslb-088-073-105-233.pools.arcor-ip.net) |
| 19:09.03 | Dr-Linux | :S |
| 19:09.07 | MrWup | xten => _{5,6}XX,1,AGI(speeddial.php) |
| 19:09.14 | MrWup | it says invalid extension 502 when i key in 502 |
| 19:09.21 | MrWup | exten => that is |
| 19:09.23 | Mercestes | Sorry. |
| 19:09.31 | Mercestes | use [5,6] instead fo {} |
| 19:09.36 | MrWup | ah |
| 19:09.36 | MrWup | thanks |
| 19:09.39 | Mercestes | np |
| 19:09.44 | Mercestes | Thank Strom-M, he reminded me. |
| 19:10.08 | MrWup | oh |
| 19:10.13 | MrWup | now it says wrong usage of [ ] |
| 19:10.21 | MrWup | exten => _[5,6]XX,1,AGI(speeddial.php) |
| 19:10.29 | [TK]D-Fender | MrWup: no "," in that. exten => _[56]XX,1..... |
| 19:10.30 | Mercestes | =/ |
| 19:10.35 | MrWup | ohh |
| 19:10.37 | MrWup | thanks |
| 19:10.39 | Mercestes | gah |
| 19:10.56 | Mercestes | Why can't it be like an array? {1,3,4-7} that makes so much more sense. |
| 19:11.00 | Prato | hello, i hope someone can help me, i search now for four weeks for an answer how to enable the detction of the hook-flash key of a sip phone, can someone help me? |
| 19:11.23 | Mercestes | Prato, that's up to your phone, and probably does not detect hook flash. |
| 19:11.34 | jkiff | Mercestes: How's this: http://pastebin.ca/528961 That's the whole call from answer to hangup. The * was pressed around line 241. |
| 19:11.37 | Mercestes | That's because hook-flash is largely unnecessary on a sip phone. |
| 19:11.38 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 19:12.05 | Qwell[] | Mercestes: it can be a range |
| 19:12.10 | Qwell[] | [134-7] |
| 19:12.14 | FuriousGeorge | hey all |
| 19:12.18 | Qwell[] | pretty sure |
| 19:12.18 | Prato | I have two different SIP Phones that send a short break (not dtmf) when pressing the flash button, one can hear the break but asterisk does not detect it |
| 19:13.05 | Mercestes | 234: CSeq: 103 BYE |
| 19:13.11 | Prato | so the problem is, that different sip provider with asterisk can detect this feature and make an attended transfer but my asterisk does nothing |
| 19:13.23 | Mercestes | Qwell[], i know..I just expect both {} and 1,2,3 from previous syntaxes. :( |
| 19:13.29 | Mercestes | Qwell[], you are correct tho |
| 19:13.38 | FuriousGeorge | last time i asked about 1.4.3 someone i could trust told me it was not ready for production. I notice 1.4.4 has been out for a while. does anyone i trust think this one *is* ready for production? |
| 19:14.04 | Mercestes | Prato, Are you trying to transfer over a POTS line using a hook-flash? |
| 19:14.04 | *** join/#asterisk saftsack (n=saftsack@pD9E0741B.dip.t-dialin.net) |
| 19:14.18 | [TK]D-Fender | Prato: Hook-flash is not a SIP concept, not the way to do a SIP transfer. Check your phone's manual. |
| 19:14.36 | [TK]D-Fender | Prato: Perhaps you could tell us what you're using... |
| 19:14.42 | Prato | I register the device via sip |
| 19:14.55 | Mercestes | Prato, I wrote code to do that once. |
| 19:15.15 | jkiff | Mercestes: Yeah, I saw that. So is it the phone that's hanging up? It's odd because * doesn't do that when the person isn't an agent. (i.e., They're member => SIP/201 instead of member => Agent/201) |
| 19:15.22 | FuriousGeorge | my issue is that im using this metermaid patch for sip presence with parking, and i think its occasionally deadlocking my server. i say this because i have an identical server elsewhere, if anything that one has inferior memory, and it never ever deadlocks |
| 19:15.30 | Prato | so all devices are registered in sip.conf, the extensions are ready and blindtransfer works via dtmf, but the key for hook/flash is not detected |
| 19:15.47 | Mercestes | Prato, here is an example: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash |
| 19:16.02 | Prato | I can not change it on the phones, it is always a short break that is sent |
| 19:16.15 | [TK]D-Fender | Prato: Again, what kind of phone are you using? |
| 19:16.17 | FuriousGeorge | my understanding is that parking presence comes with 1.4 stock, so maybe it will make my deadlocks stop (mercifully, once and for all) |
| 19:16.50 | Mercestes | jkiff, Hrm. I wonder if * is somehow designated as "terminate call" in app_agents |
| 19:16.53 | [TK]D-Fender | Prato: And AGAIN, hook-flash is not a SIP function. Asterisk used DTMF for transfer for phones that don't natively offer that functionality (very few) |
| 19:17.22 | [TK]D-Fender | Mercestes: hint : it is for app_agentlogin |
| 19:17.57 | Prato | Mercestes: I know this wiki page, but I think its only for sending a flash signal from asterisk, or am i wrong? |
| 19:18.18 | [TK]D-Fender | Prato: that is to send a flash TO an analog line that is in use. |
| 19:18.26 | Mercestes | precisely |
| 19:18.32 | [TK]D-Fender | Prato: That has nothing to do with your SIP phone jst transferring a call. |
| 19:18.51 | Prato | I use a new Aastra and a Siemens C 450 phone with newest firmware |
| 19:18.54 | [TK]D-Fender | Prato: Final request : What make & model of phone are you using? |
| 19:19.18 | [TK]D-Fender | Prato: Go read their manual to see how to transfer a call. |
| 19:19.30 | FuriousGeorge | ive tested the memory and the system stability, changed motherboards and analog hardware, noting helps. i guess nothing left to do but try different software |
| 19:20.08 | jkiff | Mercestes, [TK]D-Fender: Does it not follow features.conf? In there disconnect is set to *0, though I notice the default is *. |
| 19:20.29 | Prato | one moment, i search for the exact info for the aastra |
| 19:20.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:20.50 | antonyo14 | casimir, is your voicemailmain extension working? |
| 19:21.13 | [TK]D-Fender | jkiff: AgentLogin is its own world, and for features.conf to apply I'd have to see proof that you dialed your agent with the options to DO this. last one I recall seeing did not. |
| 19:21.49 | Hmmhesays | ~seen junk-y |
| 19:22.44 | jbot | junk-y <n=junky@modemcable105.205-56-74.mc.videotron.ca> was last seen on IRC in channel #asterisk, 16h 41m 57s ago, saying: 'JT: ive to jet, i will let you help him :)'. |
| 19:22.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:22.47 | casimir | antonyo14, yeah, having problems with it? |
| 19:22.47 | Prato | Aastra 57i and Siemens C450, I changed the mode in features conf and it reacts on a dtmf (for example *2) |
| 19:22.48 | jkiff | [TK]D-Fender: Ah, I see. |
| 19:24.24 | antonyo14 | casimir, well it wont let me save an extension for checking voicemail |
| 19:24.26 | Prato | sorry for my late answer, had a connection error. So transfer is activated and works, only the flash key is not detected. But it should work, because other sip providers can detect it with the same phones |
| 19:24.54 | antonyo14 | i save it and it saves as 'New Entry' but as soon as I click off it it goes away |
| 19:25.04 | antonyo14 | and tells me i have to set an extension |
| 19:25.27 | [TK]D-Fender | Prato: Aastra 5i series has a soft key for transferring calls. Hookflash is NOT the way to go about transferring a call |
| 19:26.00 | Prato | so my idea was to compile ztdummy, because it has a better timer and the hook signal has to be detected,but it had no effect. how can i debug the problem? i would be so happy for an answer |
| 19:26.27 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 19:27.03 | [TK]D-Fender | Prato: last time : There is no such thing as hook transfer for SIP. Zaptel has nothing to do with SIP functionality. |
| 19:27.53 | Prato | the astra uses hook flash per default and it is not changeable in the webadmin |
| 19:28.13 | [TK]D-Fender | Prato: No, it does not, and I have a 57i CT right in front of me. |
| 19:28.47 | Mercestes | lol\ |
| 19:29.36 | jkiff | [TK]D-Fender: Hrm, I don't see an applicable variable or agentlogin option to set. |
| 19:29.39 | Prato | we bought several phones and now i have the big problem that the users does not accept using others keys than the flash button for the option. it would be a dead key if it will not be recognized |
| 19:29.52 | [TK]D-Fender | Prato: XFer is the top-left softkey on the bottom set wihle on a call |
| 19:29.53 | jkiff | Oh, unless DYNAMIC_FEATURES for Dial() is what I'm looking for. |
| 19:30.08 | casimir | antonyo14, sorry to refer you back to the cli, but do you get any meaningful error message there when you try to save? |
| 19:30.08 | [TK]D-Fender | Prato: FORGET FLASH. |
| 19:30.20 | Prato | you are able to use the flash key? |
| 19:30.57 | antonyo14 | casimir, the problem is the gui will not let me set an extension for it |
| 19:31.08 | [TK]D-Fender | Prato: there is no flash in SIP. this is not how you transfer calls on that phone. Period. Get off this train of thought. |
| 19:31.13 | antonyo14 | i made an extension for voicemailmain but it is not working either |
| 19:31.28 | casimir | antonyo14, oh okay |
| 19:31.44 | Prato | but why is there a flash key on the phone, and why does pepphone detect it? |
| 19:32.19 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 19:32.39 | casimir | mine is in the default context |
| 19:32.47 | [TK]D-Fender | Prato: What is "pepphone", and "detects" it how? |
| 19:32.48 | casimir | let me pull it up |
| 19:34.10 | Prato | pepphone is a sip provider, when i register my phone with peppphone (they use asterisk) it is immediately possible to use the flash key, when I use my asterisk on an other line it is not possible |
| 19:34.44 | [TK]D-Fender | Prato: You used your 57i to connect to them directly? |
| 19:35.15 | Prato | Yes, I registered the phone directly to pepphone |
| 19:35.29 | Prato | via sip |
| 19:35.57 | [TK]D-Fender | Prato: And what exactly is this "falsh" key? Aside from the "hook" itself I don't see one. |
| 19:36.21 | jkiff | Grr, that's not it. |
| 19:36.36 | Prato | one moment, i have to call the guy with the phone, do not have it currently on the desk... |
| 19:36.39 | [TK]D-Fender | Prato: Ah, I see some option to program a soft-key for it |
| 19:36.43 | jkiff | How come agent channels are "their own little world"? |
| 19:36.56 | [TK]D-Fender | Prato: But this is definately nothing to do with *. |
| 19:37.01 | casimir | antonyo14, the gui gave me a pretty standard exten => 850,1,VoiceMailMain in the first line of the default context |
| 19:37.04 | [TK]D-Fender | Prato: it will not be able to process this. |
| 19:37.17 | antonyo14 | ok |
| 19:37.26 | lirakis | does anyone use ASTCC ? |
| 19:37.45 | lirakis | im trying to find a good website that talks about it.. digium doesnt seem to have much of anything |
| 19:38.47 | Prato | the display is split into three parts and there are so called softkeys that can be set for the top 6 keys and the bottom 6 keys and we have set the first key on the bottom part of the screen as flash |
| 19:38.54 | *** join/#asterisk dudes (n=nixtux@66-216-227-31.dhcp.stcd.mn.charter.com) |
| 19:44.08 | Prato | when selecting flash in the dropdown window for configuration this works with pepphone but not with my asterisk |
| 19:44.08 | Prato | is it possible to set the value of the flash command in the menu? we did not find it |
| 19:44.08 | antonyo14 | casimir, got it ;) |
| 19:44.09 | [TK]D-Fender | Prato: Ok, well * has no way to deal with that function. |
| 19:44.09 | Prato | the flash key works on pepphone.de, sipgate.de and sipcall.ch and on a voicemart.it pbx but not on my asterisk |
| 19:44.09 | [TK]D-Fender | Prato: So if you're hoping to have * do something of its OWN with it, nope. If your provider needs it passed on for something, you are equally out of luck. Time to shop for another provider. |
| 19:44.09 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 19:44.09 | jkiff | [TK]D-Fender: So is it *possible* to get agent channels to honor features.conf stuff? I haven't found an option or variable to set. |
| 19:44.29 | [TK]D-Fender | jkiff: Typically it calls agents though chan_local. in there you have dialplan control over what they can do. |
| 19:46.00 | Prato | do you have an idea how the threewaycalling is initiated? |
| 19:46.09 | [TK]D-Fender | Prato: "conf" softkey. |
| 19:46.28 | Prato | normally the flash should start threewaycalling |
| 19:46.39 | [TK]D-Fender | Prato: You need to wipe that little word out of your head. Seriously. |
| 19:47.06 | [TK]D-Fender | Prato: fash is a BS analog concept. |
| 19:47.22 | [TK]D-Fender | Prato: And you are dealing with a high-end SIP phone. |
| 19:47.30 | Prato | okay, so i'm possibly on the wrong way |
| 19:47.48 | [TK]D-Fender | Prato: I'm not sure how many more times I can tell you that.... |
| 19:48.14 | *** join/#asterisk yannj_fr (n=yannj@82.227.103.140) |
| 19:48.46 | yannj_fr | hello everybody |
| 19:49.03 | Prato | i believe you. |
| 19:49.49 | yannj_fr | I have a strange problem, no one of call features (attented transfer, call park ..) work |
| 19:50.00 | Prato | okay, i will forget the "flash", but can you give me a hint how threewaycalling is started? |
| 19:50.31 | savaticus | hit the conference button on your sip phone |
| 19:52.17 | Prato | where is this conference button on the 57i? |
| 19:52.17 | [TK]D-Fender | Prato: Be on a call. Press conf key. Next line gets pulled automatically (You can maybe specify which if you have truely seperate identities on the phone). You then dial the 3rd party. they answer. You press conf again. |
| 19:52.17 | De_Mon | yannj_fr how are you trying to use it? |
| 19:52.17 | [TK]D-Fender | Prato: I've done this almost a dozen times today. |
| 19:52.17 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 19:52.17 | savaticus | its a soft key try pressing "more" |
| 19:52.17 | savaticus | or read the 57i users guide |
| 19:52.18 | yannj_fr | De_Mon : during the call is pressed: *2number or #72 for park call (configured it in features.com and feature show give me the right config) |
| 19:52.18 | yannj_fr | but nothing happened |
| 19:52.50 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 19:52.59 | Prato | sorry i do not have it on my desk, i always have to ask the owner that has the problem |
| 19:53.00 | jkiff | [TK]D-Fender: I'm not sure I follow. Since they're members of a queue, I just Queue(blah) and they're called. What am I missing? |
| 19:53.02 | De_Mon | yannj_fr what does your dial command look like that connected to that device? |
| 19:53.20 | De_Mon | yannj_fr also, what phone / protocol are you using? |
| 19:53.29 | [TK]D-Fender | yannj_fr: pastebin the full CLI output of the call that fails to transfer. |
| 19:53.38 | De_Mon | that would work too |
| 19:54.25 | yannj_fr | I tested with Grandstream budgeton 200 / GXP 2000 / Thomson ST2030 / Xlite, I am using SIP with RFC 2233 dtmf signalling |
| 19:54.29 | [TK]D-Fender | jkiff: go look at how the agent is being called |
| 19:55.11 | jkiff | I gotcha. |
| 19:55.16 | yannj_fr | the most strange is that pressing button doesnt show anything on console even with core set verbose 7, sip set debug, core set debug on |
| 19:55.21 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 19:56.02 | De_Mon | yannj_fr pastebin the cli output of your call attempt |
| 19:56.23 | yannj_fr | you mean the call etablishment? |
| 19:56.36 | De_Mon | yannj_fr call start to call end |
| 19:56.41 | yannj_fr | ok |
| 19:57.04 | yannj_fr | I will try to reproduce the problem, it is on my pbx at work |
| 19:57.23 | De_Mon | so? no passwords in the cli logs |
| 19:57.43 | yannj_fr | , just that I am at home, it is 21h57 there |
| 19:57.44 | yannj_fr | ! |
| 19:58.05 | [TK]D-Fender | yannj_fr: how HELPFUL. Perhaps you should ask when you're in a position to DO something about it :) |
| 19:58.32 | yannj_fr | just that IRC is blocked by proxy |
| 19:58.48 | yannj_fr | else I would already have done it |
| 19:58.52 | Mercestes | IRC is blocked by proxy here too |
| 19:58.55 | [TK]D-Fender | yannj_fr: then I guess you're pretty screwed. |
| 19:59.00 | *** join/#asterisk BSD_Tech (n=BSDTech@ppp-71-128-6-42.dsl.irvnca.pacbell.net) |
| 19:59.22 | Mercestes | dyndns and remote desktop > proxy |
| 19:59.35 | antonyo14 | what is the variable that is set from the input of WaitExten? |
| 19:59.59 | Qwell[] | antonyo14: I don't think there is one - it just goes to the exten you dialed |
| 20:00.00 | [TK]D-Fender | antonyo14: ${EXTEN} , and it GOES to that extension. |
| 20:00.11 | antonyo14 | oh yeah |
| 20:01.56 | *** join/#asterisk SeanLostInAsteri (n=SeanLost@p54BE96AE.dip0.t-ipconnect.de) |
| 20:02.52 | dudes | use a shell to a remote linux box using BX or something |
| 20:02.54 | Prato | Sean is the person with the Aastra and can describe the problem in abetter way |
| 20:03.36 | Mercestes | Prato, you brought your coworkers in here to be berated by [TK]D-Fender? wow, bad move man. |
| 20:05.18 | Prato | so the problem was that he never used irc and wanted to help him |
| 20:07.20 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 20:07.38 | nowork | hi, how can I check my OpenH323 , pwlib version?? |
| 20:08.35 | [TK]D-Fender | Mercestes: You're not paranoind... but I'm only out to get YOU ;) |
| 20:08.40 | [TK]D-Fender | Mercestes: (j/k) |
| 20:09.17 | Mercestes | lol |
| 20:09.32 | Mercestes | I know I'm your favorite. :P |
| 20:10.12 | SeanLostInAsteri | hi to every one... i have got a question regarding an aastra 57i and asterisk... how do i initiate a threewaycalling? |
| 20:10.27 | Mercestes | SeanLostInAsteri, the conference button |
| 20:10.37 | jkiff | [TK]D-Fender: Okay, I'm doing a Set(__DYNAMIC_FEATURES=disconnect) just before the Dial(SIP/${EXTEN}) where the agent is called, but it's not working. So I tried to duplicate the disconnect feature in [applicationmap] of features.conf with "disconnect => *0,callee,hangup,", but that's not working either. :( |
| 20:10.53 | [TK]D-Fender | jkiff: PASTEBIN IT. |
| 20:10.58 | [TK]D-Fender | SeanLostInAsteri: [TK]D-Fender>Prato: Be on a call. Press conf key. Next line gets pulled automatically (You can maybe specify which if you have truely seperate identities on the phone). You then dial the 3rd party. they answer. You press conf again. |
| 20:11.24 | SeanLostInAsteri | Mercestes, can i set the value what the conference button does? |
| 20:11.40 | SeanLostInAsteri | my button only send hook flash and this is no good for asterisk |
| 20:11.59 | nowork | hi, it's a linux question, don't kick me... I use ulimit -n 40960 to open more file limitatios, but when I close the shell window, and reconnected , it changed back to 1024, anyway i can do let it changed permenant?? |
| 20:12.03 | Mercestes | SeanLostInAsteri, This should not be the case. |
| 20:12.08 | [TK]D-Fender | SeanLostInAsteri: there is nothing to set. this is a FIXED option on your lower soft-keys |
| 20:12.21 | Mercestes | SeanLostInAsteri, and if you set the valu eof what it does, it won't work as a conference button anymore |
| 20:12.24 | [TK]D-Fender | SeanLostInAsteri: You can't override this. |
| 20:12.58 | SeanLostInAsteri | exactly, it is a fixed key. i can only imagine the current firmware i am using is crap |
| 20:13.49 | SeanLostInAsteri | it should be apparently a new version for europe but even the spelling within the menu is totally wrong |
| 20:13.57 | [TK]D-Fender | SeanLostInAsteri: What version are you on? |
| 20:14.36 | [TK]D-Fender | SeanLostInAsteri: 2.0.1.1076 works fine here |
| 20:14.43 | SeanLostInAsteri | Firmware 2.0.1.1076 |
| 20:14.47 | IOscanner | I have a dual AMD Optron 280 system. Everything works fine, but wav files don't play when using an IVR. GSM, MP3 and other files play fine. |
| 20:14.47 | *** join/#asterisk stevej (n=stevej@mail.joneslinux.com) |
| 20:15.00 | SeanLostInAsteri | Boot Version 1.1.0.1245 |
| 20:15.08 | IOscanner | Where should I look? I have compiled from svn, stable and it doesn't work. |
| 20:15.12 | [TK]D-Fender | SeanLostInAsteri: well try the method I just pasted for you. |
| 20:15.38 | [TK]D-Fender | IOscanner: make sure its 8khz mono. |
| 20:15.51 | [TK]D-Fender | IOscanner: Encoding format matter. Check the WIKI for details |
| 20:15.53 | IOscanner | iIt works on other boxes. |
| 20:16.09 | [TK]D-Fender | IOscanner: other ASTERISK boxes? |
| 20:16.12 | IOscanner | Just started happening on these new AMD64 optrons |
| 20:16.18 | [TK]D-Fender | IOscanner: AH... HRM |
| 20:16.37 | IOscanner | I have other boxes with amd64 +3800 same config same install and they work |
| 20:17.18 | SeanLostInAsteri | [TK]D-Fender did not get anything |
| 20:17.25 | [TK]D-Fender | IOscanner: well if you've double checked the files & everything else sane I'm not sure where to go next... |
| 20:17.26 | IOscanner | I have been tried rebuilding kernel for 32 bit, 64, smp no smp. Then rebuild asterisk and it doesn't change |
| 20:18.02 | [TK]D-Fender | sean place a call, do you have "Conf" bottom key, left side? |
| 20:18.30 | SeanLostInAsteri | yes... bottom display part midle button left side |
| 20:19.03 | [TK]D-Fender | SeanLostInAsteri: Press it. What happens? |
| 20:19.23 | [TK]D-Fender | SeanLostInAsteri: Your 1st line should start flashing and you should get dialtone ont he next line. |
| 20:19.27 | SeanLostInAsteri | when not in a call with some one nothing |
| 20:19.43 | SeanLostInAsteri | while talking to someone on some systems it works and on asterisk with us not |
| 20:19.49 | [TK]D-Fender | SeanLostInAsteri: while ON ca call. |
| 20:19.59 | IOscanner | would it be an asterisk problem or a locak lib problem. |
| 20:20.20 | IOscanner | Where does asterisk play wav files from. Does it have its own libs for that? |
| 20:20.56 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:21.23 | SeanLostInAsteri | i think something is not right with our asterisk... does some one now what the aastra 57i transmits to the asterisk when conf is pressed? |
| 20:22.12 | [TK]D-Fender | SeanLostInAsteri: Its not *. Its your PHONE's config |
| 20:22.30 | [TK]D-Fender | SeanLostInAsteri: I'm betting your provider screwed all sorts of settings on it. |
| 20:23.37 | SeanLostInAsteri | i understand perfectly what you are saying... the threewaycalling is a chip module on the phone. how come does it work with some providers and not with others? |
| 20:23.58 | SeanLostInAsteri | that is the thing i can not explain to my self... |
| 20:24.24 | SeanLostInAsteri | the phone is directly from aastra in switzerland. no provider has ever touched it... |
| 20:24.39 | [TK]D-Fender | SeanLostInAsteri: Phone settings can be messed with. To what degree I don't know what they've done. My 57i CT works just fine. |
| 20:25.16 | [TK]D-Fender | SeanLostInAsteri: Well it didn't configure itself Something is off, but i'm about to leave the office and will not have access to mine for further refernce today. |
| 20:26.09 | SeanLostInAsteri | thank you any way... this is the first time i am here and i am positively impressed... |
| 20:26.45 | Prato | [TK]D-Fender: Thank you for your help |
| 20:27.21 | [TK]D-Fender | well, we tried.... go look on the WIKi to see if they some detailed guides on the 5i series |
| 20:27.30 | [TK]D-Fender | they are rather new... |
| 20:29.00 | *** part/#asterisk BSD_Tech (n=BSDTech@ppp-71-128-6-42.dsl.irvnca.pacbell.net) |
| 20:29.22 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 20:30.25 | jkiff | [TK]D-Fender: http://pastebin.ca/529149 |
| 20:34.51 | yannj_fr | Is there a way to force use of a specific codec for call between certain peers, and other one for calls to other specific peers |
| 20:34.52 | yannj_fr | ? |
| 20:35.05 | *** join/#asterisk mexuar-tim (n=mexuar-t@212.183.134.209) |
| 20:38.15 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:39.02 | jkiff | Grr, this is retarded. Why don't agent channels honor features.conf stuff? |
| 20:41.11 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 20:43.06 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 20:46.32 | IOscanner | Anyone have any ideas how I can at least get an error from asterisk to see why .wav files will not play The .wav files are copied from another asterisk box and confirmed good. |
| 20:46.58 | IOscanner | Only difference in the two boxes one is amd64 +3800 the new one is Dual amd optron 280's |
| 20:47.09 | yannj_fr | did you reload moh? |
| 20:47.12 | kiscokid | are you using high verbosity? |
| 20:47.21 | IOscanner | These are IVR recordings |
| 20:47.37 | IOscanner | we are using mp3 files for moh and they work |
| 20:47.40 | IOscanner | gsm works |
| 20:48.06 | *** join/#asterisk fbffff (n=fbffff@adsl-69-209-215-64.dsl.chcgil.ameritech.net) |
| 20:48.38 | IOscanner | just can't get .wav file to work. I even move the drive from the other machine to test the build that I know works. Everything works but the .wav on the new box. |
| 20:49.13 | yannj_fr | do core set verbose 7 |
| 20:49.14 | IOscanner | Very strange only see it with the optron CPU. Just not sure where or how Asterisk is playing the file. Is it an extenal library? |
| 20:49.20 | yannj_fr | and try to read it |
| 20:49.37 | yannj_fr | and pastebin what happened |
| 20:50.42 | IOscanner | http://paste.uni.cc/15982 |
| 20:50.50 | IOscanner | It sees the file and thinks it is playing |
| 20:51.43 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 20:51.48 | nick125_lappy | Hey |
| 20:51.51 | *** part/#asterisk ctaloi (n=ctaloi@nat-66-218-1-47.usadatanet.com) |
| 20:51.58 | nick125_lappy | I need to mix two legs of a recorded calls, how would I do that? |
| 20:52.06 | *** join/#asterisk [hC] (n=hardcore@69.90.99.197) |
| 20:52.29 | yannj_fr | did you try to convert files in gsm? |
| 20:52.57 | nick125_lappy | I'm not sure what format they are in, let me look, hold on |
| 20:53.19 | nick125_lappy | Nope, they aren't in GSM |
| 20:53.38 | yannj_fr | sorry was for : IOscanner |
| 20:53.42 | nick125_lappy | oh |
| 20:54.17 | IOscanner | no I can |
| 20:54.23 | IOscanner | I know .gsm files work |
| 20:54.31 | yannj_fr | .. |
| 20:54.33 | yannj_fr | ok |
| 20:54.53 | IOscanner | strange that just .wav is not working. |
| 20:55.00 | yannj_fr | what codec are you using on you phone? |
| 20:55.56 | IOscanner | G711u |
| 20:56.35 | yannj_fr | and what is the format of you wav ? (bit, sampling..) |
| 20:56.49 | IOscanner | I am using the same image that is on a normal AMD64 +3800 box and it works everything work. Move the drive to this box and test . Everything works but .wav files |
| 20:57.44 | IOscanner | HelpDeskMain.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 20:58.03 | IOscanner | works on other Asterisk boxes. |
| 20:58.11 | IOscanner | So I don't think that is the problem |
| 20:58.27 | IOscanner | we did a new install and copied as well and pull a drive and try. |
| 20:58.36 | IOscanner | Must be a library issue or something. |
| 20:59.11 | yannj_fr | no wav can be played? |
| 20:59.27 | IOscanner | how can I convert from wav to gsm? Anyone know a command line tool? |
| 20:59.35 | IOscanner | correct no wav can be played |
| 20:59.42 | Greek-Boy | is it possible to signal caller id onto PSTN when calling from a voip sip device? |
| 20:59.47 | ectospasm | IOscanner: I think sox can do it |
| 21:00.02 | IOscanner | Ithought so, but I did remember |
| 21:00.33 | ectospasm | it has been almost two years since I did that conversion, so I can't tell you the exact command line |
| 21:00.35 | yannj_fr | just do file convert file.exorig file.gsm |
| 21:04.51 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:05.58 | IOscanner | cool now I really broke it on the last rebuild. Now no audio plays. |
| 21:06.11 | IOscanner | I tried to rebuild for i686 |
| 21:07.58 | yannj_fr | IOscanner : are you installing asterisk from sources ? |
| 21:08.15 | IOscanner | yes from svn tree |
| 21:08.24 | yannj_fr | svn 1.4? |
| 21:08.35 | IOscanner | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 21:08.38 | yannj_fr | ok |
| 21:08.57 | IOscanner | Not ready for 1.4 I will have to make some dialplan changes and I haven't had time |
| 21:09.16 | yannj_fr | did you enable on menuconfig : format interpreter , format_pcm |
| 21:09.17 | yannj_fr | ? |
| 21:09.41 | IOscanner | menuconfig? where? |
| 21:10.04 | yannj_fr | ./configuree |
| 21:10.06 | yannj_fr | make |
| 21:10.13 | yannj_fr | sorry |
| 21:10.21 | yannj_fr | after ./configure |
| 21:10.26 | yannj_fr | what are you doing? |
| 21:10.58 | IOscanner | there is no ./configure with 1.2 |
| 21:10.58 | yannj_fr | sh*t... |
| 21:10.59 | yannj_fr | sorry I works only the 1.4 |
| 21:11.12 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:11.21 | IOscanner | that is why I have not hear of that. |
| 21:11.35 | yannj_fr | how do you install it then? |
| 21:11.43 | IOscanner | I need to just make the change week. Just going to take a week to change it. |
| 21:12.09 | IOscanner | vi the Makefile then make clean && make && make install |
| 21:12.20 | IOscanner | same as I have for 4 years |
| 21:13.02 | IOscanner | Just trying to build some very large dual and quad boxes for front end systems and we can't asterisk to play audio. Everything else works |
| 21:13.30 | IOscanner | I even tried keeping 32bit instead of 64bit makes no difference |
| 21:14.17 | yannj_fr | no idea, for me |
| 21:14.42 | IOscanner | I don't even get an error |
| 21:14.52 | yannj_fr | sorry, I am not enought good |
| 21:14.53 | IOscanner | nothing to go on |
| 21:14.54 | yannj_fr | ! |
| 21:15.02 | IOscanner | me either |
| 21:15.06 | IOscanner | thanks for trying |
| 21:15.11 | yannj_fr | do you use sip? |
| 21:15.20 | IOscanner | yes and IAX |
| 21:15.42 | yannj_fr | just try something |
| 21:15.48 | yannj_fr | like changing the codec |
| 21:15.53 | yannj_fr | you are using |
| 21:15.55 | yannj_fr | and testing |
| 21:16.01 | yannj_fr | as an example |
| 21:16.09 | yannj_fr | force phone to use gsm |
| 21:17.31 | IOscanner | same thing |
| 21:17.40 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 21:17.42 | IOscanner | let me try with a sip phone too |
| 21:19.45 | pigpen | Hi all, I am getting: http://pastebin.ca/529279 |
| 21:20.01 | IOscanner | same |
| 21:20.28 | pigpen | When I attempt to use call files (*.call) to connect sip extensions, via speaker (polycom) to an extension that does a playback of a gsm file. |
| 21:20.53 | pigpen | I am shoving it out to about 75 phones at a time, with a total of about 157 polycom's. |
| 21:21.03 | pigpen | ideas? |
| 21:21.50 | pigpen | Oh...* server is a Dell PE6850, 8GB ram, 15K SAS Array, etc.... |
| 21:21.51 | [TK]D-Fender | pigpen, Spool them up slower |
| 21:21.55 | pigpen | k. |
| 21:21.58 | pigpen | thank you master. |
| 21:22.01 | pigpen | :) |
| 21:22.34 | De_Mon | grr |
| 21:22.46 | CoffeeIV_ | in the asterisk C code, I see a "dtimeout" field in the pbx structure that sets the timeout between digit presses. It is hard coded to 5 sec. Can this be overriden by something in a .conf file ? I grepped the code but I saw no place where that was overriden from somewhere else. |
| 21:23.34 | [TK]D-Fender | CoffeeIV_, thats not what you're seeing then. |
| 21:23.44 | [TK]D-Fender | CoffeeIV_, "show function TIMEOUT" |
| 21:24.59 | CoffeeIV_ | [TK]D-Fender: thanks, that looks like it might be what I need |
| 21:25.25 | Hmmhesays | finally I got this damn faxing working |
| 21:26.09 | yannj_fr | faxing from analog? |
| 21:27.06 | pigpen | [TK]D-Fender, would it be normal for me to have many of the sip phones to become "unreachable" during this process? |
| 21:27.07 | Hmmhesays | ip to pstn and pstn to ip |
| 21:27.40 | Hmmhesays | now I need someone to send me a fax |
| 21:27.52 | [TK]D-Fender | pigpen, not sure. this is X ULAW calls at a time.... |
| 21:28.08 | [TK]D-Fender | pigpen, sond potentially substantial? |
| 21:28.23 | pigpen | yeah...ulaw. |
| 21:29.19 | pigpen | I think what is happening, is that I have 20-40 going, then when I have several more being placed, it cannot place the call sue to they are unavailable. |
| 21:29.26 | pigpen | I guess I need to space it out more. |
| 21:29.34 | *** part/#asterisk dudes (n=nixtux@66-216-227-31.dhcp.stcd.mn.charter.com) |
| 21:39.58 | *** join/#asterisk saftsack (n=saftsack@pD9E0741B.dip.t-dialin.net) |
| 21:40.26 | pigpen | yeah, 10 at a time is working fine. |
| 21:40.55 | antonyo14 | would anyone be so nice as to pastebin macro-trunkdial for me? |
| 21:41.42 | [TK]D-Fender | antonyo14, This is not a GUI support channel, and I could give you a 1-line macro, but I promise you won't like it :) |
| 21:42.32 | antonyo14 | ok |
| 21:42.47 | antonyo14 | sorry im going home now :( |
| 21:44.08 | *** join/#asterisk runa (i=foobar@201.250.82.66) |
| 21:44.18 | *** join/#asterisk [Airwolf] (n=airwolf@89.205.134.44) |
| 21:45.52 | runa | hey :) I have one of those dumb answering machines provided by my telco (ie, call *123 to pick up your messages). I was wondering if I could write a client to use my grandstream FXO, call the answering machine, check if there are messages, retrieve them and delete. Any hints on where I should start? |
| 21:49.02 | *** join/#asterisk kimosabe (n=tatt@189.175.41.224) |
| 21:49.57 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:50.02 | lisandropm | hello |
| 21:50.14 | lisandropm | has anyone linked asterisk with a hicom 300? |
| 21:50.31 | kimosabe | does anyone know a good voice overip provider with unlimeted usa calling |
| 21:52.10 | [TK]D-Fender | runa, You want * to actualy listen to your machine and do this automated? Your odds are inifiitely approaching zero... |
| 21:53.18 | kimosabe | can some one lead me in the direction of a good voice provider please with great rates |
| 21:57.07 | runa | [TK]D-Fender: damn. I thought so. but It doesn't sounds too difficult.. |
| 21:57.28 | *** join/#asterisk [hC] (n=hardcore@69.90.99.197) |
| 21:58.55 | *** join/#asterisk pruonckk (n=mike@200.212.179.130) |
| 21:59.12 | pruonckk | Hi all, |
| 21:59.17 | [TK]D-Fender | runa, Of course not... thats what dreams are for! |
| 21:59.52 | [TK]D-Fender | runa, * will not know when a message has ended, what to press, and be able to make decisions for you quite like that. |
| 22:00.48 | pruonckk | somebody know a good documentation about how can i implement a account code on asterisk ( im newbie on asterisk ) |
| 22:01.24 | [TK]D-Fender | pruonckk, Please describe exactly what you mean by this as it can be interpreted several ways |
| 22:02.01 | pruonckk | ok, i want that my users to enter with a code (4 numbers of less) after do a dial |
| 22:02.26 | runa | [TK]D-Fender: well, I could script that |
| 22:04.28 | [TK]D-Fender | pruonckk, So like a password to dial out? |
| 22:04.33 | pruonckk | yeah |
| 22:04.35 | pruonckk | like a password |
| 22:04.42 | [TK]D-Fender | pruonckk, "show application read" |
| 22:04.47 | [TK]D-Fender | pruonckk, "show application gotoif" |
| 22:05.09 | Mercestes | runa, If you can script that I definately want to hire you to do something...I dunno what yet but I'll think of something. |
| 22:05.18 | Mercestes | maybe script a front loader to build me an island or something |
| 22:05.49 | [TK]D-Fender | Mercestes, The Tonka master-plan not work out like you planned? :) |
| 22:06.10 | Mercestes | Yea, no, they aren't *really* indestructable. |
| 22:12.39 | festr__ | anyone know, if digium's codec g729 has PLC? |
| 22:16.26 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 22:24.26 | *** join/#asterisk angom_h (n=Angel@189.140.16.141) |
| 22:29.06 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 22:32.27 | De_Mon | hmm.. |
| 22:33.43 | *** join/#asterisk ingenio (n=ingenio@12-216-99-16.client.mchsi.com) |
| 22:34.58 | *** join/#asterisk Kubicek (n=nnnnnnnn@gw.letna.cz) |
| 22:35.11 | ingenio | Just received a tdm02b... are there rj11 to rj45 conversion guides? Basically.. how do I plug my analog phone into the card? :P |
| 22:37.10 | Kubicek | i have a problem with sending faxes from spandsp (txfax) - only part gets transmited and the rest of the pages are lines. any ideas ? |
| 22:38.49 | De_Mon | I'm running asterisk 1.2.14 and am having some trouble with queues |
| 22:38.50 | De_Mon | http://pastebin.ca/529424 |
| 22:39.21 | De_Mon | all members of the queue are busy, yet the exitstatus is timeout |
| 22:39.56 | De_Mon | join and joinempty = strict |
| 22:39.56 | De_Mon | leavewhenempty = strict |
| 22:41.37 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:46.23 | *** join/#asterisk anthm (n=anthm@m010f36d0.tmodns.net) |
| 22:46.23 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:47.51 | [TK]D-Fender | ingenio, the jack may be RJ45, but you can plug your RJ11 right into it. |
| 22:48.05 | [TK]D-Fender | ingenio, Its wired for the center apir... its jsut how they are made... |
| 22:49.16 | poppo | I have a php script that connect using fopen to the asterisk box everything work but when it gets to the exten => 4,2,MYSQL(Query resultid ${connid} UPDATE\ `calls`\ SET\ `verified`=\'Yes\'\ WHERE\ `phone`=\'$[phone]\') |
| 22:49.45 | poppo | the VarSet=$phone from the php script is not passing to the mysql query |
| 22:49.49 | poppo | what is it that i am missing |
| 22:49.59 | [TK]D-Fender | De_Mon, try again, and pastebin this as well : "show queues" before AND afer. please add your queues.conf entry as well. |
| 22:50.38 | Qwell[] | huh, I thought we switched to rj11 |
| 22:50.46 | Qwell[] | must be old stock |
| 22:51.52 | mvanbaak | rj11 ? |
| 22:52.03 | Qwell[] | on the tdm400p |
| 22:52.13 | poppo | I have this in my php scripts fputs($oSocket, "SetVar: phone=$strCallerId\r\n"); |
| 22:52.28 | mvanbaak | I replaced my last rj11 4 years ago |
| 22:52.35 | mvanbaak | it's all rj45 now |
| 22:53.08 | Qwell[] | I still need to wire my house |
| 22:53.29 | Qwell[] | I have *2* (I found a second one in my bedroom, which goes...somewhere...don't ask) phone jacks in my house |
| 22:54.01 | De_Mon | [TK]D-Fender it looks like "SIP/chrisoffice-random is busy" isn't the same as "unvailable" |
| 22:54.08 | mvanbaak | I only have fiber jacks |
| 22:54.22 | [TK]D-Fender | De_Mon, Correct |
| 22:54.49 | De_Mon | damn |
| 22:54.58 | mvanbaak | those connect to netgear gbit switches |
| 22:55.05 | [TK]D-Fender | De_Mon, That response says your agent is not on a call nor paused, and its just the PHONE rejecting the call (looks like DND which is BAD) |
| 22:55.12 | mvanbaak | they are fiber uplink + 24 port gbit rj45 |
| 22:55.19 | mvanbaak | I dont have rj11 anymore |
| 22:55.52 | mvanbaak | and last month I replaced all rj11 jacks in my parents house with rj45 |
| 22:56.11 | mvanbaak | you can run 2 analog lines with one rj45/cat5 cable |
| 22:56.30 | Strom_M | you can run three analog lines on an RJ14 :) |
| 22:56.33 | mvanbaak | or 1 network+1isdn |
| 22:56.54 | De_Mon | dnd is bad? hrm |
| 22:57.02 | mvanbaak | my parents place now has analog phone + network on every rj45 socket |
| 22:57.04 | [TK]D-Fender | mvanbaak, You can do the same on a single RJ11... its a 6 conductor jack you know... |
| 22:57.40 | Strom_M | [TK]D-Fender: if you want to be technical about it, RJ11 is 6P2C, RJ12 is 6P4C, RJ14 is 6P6C |
| 22:58.25 | Qwell[] | what pair does PoE run over? |
| 22:58.45 | ingenio | [TK]D-Fender: nifty. i had no idea. :P |
| 22:58.47 | [TK]D-Fender | Strom_M, Ok, never knew you only renumbered just because of actually USING what you've already got there... |
| 22:58.50 | Strom_M | pair 4, IIRC |
| 22:58.52 | mvanbaak | Qwell[]: no idea |
| 22:58.56 | Qwell[] | 7-8? |
| 22:58.56 | ingenio | [TK]D-Fender: haha, i was about to start splicing. thanks |
| 22:59.24 | Strom_M | [TK]D-Fender: well the RJ number specifies more than just the physical layout of the jack |
| 22:59.26 | Qwell[] | so you could do PoE and still carry phone...hmm |
| 22:59.41 | Strom_M | Qwell[]: yeah, I think so |
| 22:59.50 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:59.56 | Strom_M | but i dont remember definitively |
| 23:00.07 | [TK]D-Fender | ingenio, If you wire for Cat5, you can plug a splitter and use 1 for phone, the other for 10/100. Std Ethernet doesn't use the middle pair so you don't have to do any kind of splicing. |
| 23:00.14 | mvanbaak | I use PoE injectors |
| 23:00.27 | [TK]D-Fender | Qwell[], PoE uses all 8 wires. |
| 23:00.28 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:00.42 | Qwell[] | hmm, suck |
| 23:00.52 | Strom_M | [TK]D-Fender: pair 1 for + and pair 4 for -? |
| 23:00.58 | mvanbaak | but soon I will be using a 3750 for my phones and laptops |
| 23:01.19 | Strom_M | ah, here it is |
| 23:01.20 | Strom_M | http://www.interfacebus.com/Power_Over_Ethernet.html |
| 23:01.21 | [TK]D-Fender | Strom_M, Don't recall the exact pattern.. been a wihle but diagrams I saw say all 8. |
| 23:01.23 | Strom_M | I was wrong :) |
| 23:01.36 | Qwell[] | Wouldn't pair 1 be orange? O.o |
| 23:01.41 | Strom_M | Qwell[]: no |
| 23:01.46 | mvanbaak | no |
| 23:01.52 | Strom_M | blue orange green brown slate |
| 23:01.57 | Strom_M | 1 2 3 4 5 |
| 23:02.02 | Qwell[] | 5? |
| 23:02.05 | [TK]D-Fender | 8 6 7 5 3 0 9 ;) |
| 23:02.09 | mvanbaak | I ordered a 3750 gbit with poe |
| 23:02.23 | Strom_M | Qwell[]: TIA-568B reverses pairs 1 and 3 |
| 23:02.44 | Strom_M | but pair 1 is always blue |
| 23:02.45 | mvanbaak | freaking expensive |
| 23:03.02 | Qwell[] | 12=1, 36=2, 45=3, 78=4? |
| 23:03.07 | Qwell[] | erm |
| 23:03.09 | Strom_M | no |
| 23:03.19 | Strom_M | pair 1 = pins 4/5 |
| 23:03.19 | Qwell[] | I give up |
| 23:03.25 | Strom_M | pair 2 = pins 3/6 |
| 23:03.31 | Strom_M | pair 3 = pins 1/2 |
| 23:03.36 | Strom_M | pair 4 = pins 7/8 |
| 23:03.44 | Qwell[] | wouldn't B reverse pairs 2 and 3 then? |
| 23:03.53 | Strom_M | maybe that's it |
| 23:03.59 | Qwell[] | gotta be :D |
| 23:04.02 | Qwell[] | 1-3, 2-6 |
| 23:04.05 | *** join/#asterisk `Sean (i=Un1x@CPE000c258d147c-CM000a73a94167.cpe.net.cable.rogers.com) |
| 23:04.11 | Strom_M | I know the AT&T 25-pair code better than I remember TIA standards :) |
| 23:04.19 | Qwell[] | nobody uses TIA standards |
| 23:04.41 | Strom_M | uh, cat5 cabling is almost always terminated to TIA-568-A or TIA-568-B |
| 23:04.50 | Strom_M | for ethernet applications, anyway |
| 23:04.56 | Qwell[] | I was trolling :p |
| 23:05.00 | Strom_M | dork |
| 23:05.01 | Strom_M | :) |
| 23:05.27 | mvanbaak | too bad you cant get ibm thinkpads with fiber |
| 23:05.48 | mvanbaak | as soon as that's possible I'll remove all rj45/catX cables |
| 23:06.26 | mvanbaak | I have 12 fiber sockets in my house |
| 23:06.27 | russellb | standards are overrated |
| 23:07.00 | mvanbaak | I use 5port netgear switches just to convert the fiber to rj45 |
| 23:07.45 | mvanbaak | I have 2 cisco phones, so there I'll need those. but all my computers should get fiber cards |
| 23:07.49 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:08.28 | mvanbaak | would be cool to get fiber nicks that support more then 1 color |
| 23:08.46 | ingenio | anyone have any OS suggestions for Asterisk? ie, Fedora vs Ubuntu |
| 23:08.52 | ingenio | s/os/distro |
| 23:09.04 | mvanbaak | use whatever you are used to |
| 23:09.09 | Strom_M | use whatever you're most comfortable with |
| 23:09.33 | mvanbaak | if you're not used to any I suggest debian |
| 23:09.38 | mvanbaak | but that's personal |
| 23:09.41 | ingenio | i grew up with debian using the cli |
| 23:09.49 | Strom_M | I second debian :) |
| 23:09.54 | ingenio | but i really should play with some of the newer guis |
| 23:10.00 | ingenio | hence the ubuntu idea |
| 23:10.08 | Strom_M | you dont want x windows on your asterisk box |
| 23:10.10 | Strom_M | there's no reason |
| 23:10.32 | Strom_M | it'll only cause you headaches :) |
| 23:10.32 | mvanbaak | ubuntu == debian + some patches that break stuff |
| 23:10.36 | ingenio | haha |
| 23:10.55 | mvanbaak | I mean: ubuntu server installs a PREEMPT kernel by default |
| 23:11.00 | Qwell[] | mvanbaak: support more than one color? |
| 23:11.03 | mvanbaak | how fuckedup can that be |
| 23:11.13 | Qwell[] | hey, at least it doesn't install X by default |
| 23:11.14 | mvanbaak | Qwell[]: dark fiber |
| 23:11.23 | Qwell[] | *cough*RHEL5*cough* |
| 23:11.34 | Qwell[] | (and it's non-trivial to remove) |
| 23:11.36 | poppo | I need help [Tk] D-Fender can you help out |
| 23:11.46 | mvanbaak | Qwell[]: 256x15gbit on one fiber |
| 23:11.52 | russellb | are you kidding? Asterisk *requires* X to run |
| 23:12.11 | russellb | we do call control in OpenGL now |
| 23:12.12 | Qwell[] | bah, I just run it with ssh -X |
| 23:12.26 | Qwell[] | I still say we need to get transcoding on video cards |
| 23:12.31 | Qwell[] | That would be *hot* |
| 23:12.36 | mvanbaak | russellb: does it run on XGL ? |
| 23:12.49 | russellb | Qwell[]: even after today's commit, still nobody lauged at X-Disclaimer |
| 23:12.54 | Qwell[] | mvanbaak: yeah, each side of the cube is a context |
| 23:12.59 | russellb | mvanbaak: totally. you should see a call transfer with wobbly windows enabled |
| 23:13.07 | mvanbaak | cool |
| 23:13.19 | Qwell[] | I need to enable XGL on here... |
| 23:13.25 | mvanbaak | I can transfer calls on the edge of the cube now |
| 23:13.26 | russellb | it's hot |
| 23:13.29 | Qwell[] | russellb: You should help me with that on Monday :D |
| 23:13.35 | russellb | heh |
| 23:13.38 | mvanbaak | xgl is hot |
| 23:13.42 | russellb | sudo apt-get install ... |
| 23:13.44 | russellb | :-p |
| 23:13.48 | russellb | it's in the ubuntu repos |
| 23:13.51 | Qwell[] | it doesn't like me+nvidia |
| 23:13.53 | Qwell[] | I use debian now :D |
| 23:14.01 | mvanbaak | debian ftw |
| 23:14.06 | mvanbaak | for linux that is |
| 23:14.06 | russellb | i'm sure it's in there too .. |
| 23:14.16 | russellb | pfft, i use debian windows vista and it pwns |
| 23:14.17 | Qwell[] | gnome+nvidia+debian/ubuntu+me=meh |
| 23:14.29 | mvanbaak | gnome is bad |
| 23:14.34 | mvanbaak | I really dont like it |
| 23:14.59 | mvanbaak | ion3++ |
| 23:15.01 | russellb | well your opinion doesn't matter unless your opinion is the same as mine |
| 23:15.01 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 23:15.19 | [TK]D-Fender | Everyone is entitled to my opinion! ;) |
| 23:15.20 | Qwell[] | darn, I disabled compositing...I need to reenable that |
| 23:15.40 | mvanbaak | Qwell[]: try that with the ati cards ;) |
| 23:15.49 | [TK]D-Fender | Qwell[], plastic & metal don't compost very well ;) |
| 23:16.38 | mvanbaak | xgl+beryl here |
| 23:16.43 | mvanbaak | works great on my ibook |
| 23:17.17 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 23:17.55 | mvanbaak | modprobe zaptel_xgl_ati_pwnd.so |
| 23:18.16 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 23:18.36 | lisandropm | hello. Has any of you had some experience with a hicom 300? |
| 23:18.42 | mvanbaak | use the fglrx fpu to do transcoding |
| 23:18.47 | mvanbaak | that would be great |
| 23:19.43 | poppo | <[TK]D-Fender> : I have a php script that does a fputs($oSocket, "Set(phone=\$strCallerId\)\r\n"); but when asterisk in extension does exten => 4,2,MYSQL(Query resultid ${connid} UPDATE\ `calls`\ SET\ `verified`=\'Yes\'\ WHERE\ `phone`=\'${phone}\') |
| 23:19.43 | poppo | <PROTECTED> |
| 23:21.05 | [TK]D-Fender | poppo NoOp your variable before calling MYSQL |
| 23:27.25 | irule | hi guys, I am faxing with hylafax, iaxmodem, spandsp, and asterisk already, and wonder if anyone here has ever linked CUPS into the equation? I have linux, windows and mac clients :s |
| 23:28.13 | n00dle | russellb: Got the patch and applied it, notes added to bug tracker. :) (Better, but... weird.) |
| 23:29.06 | poppo | [TK]D-Fender: umm i did that in the log it dosent show the variable |
| 23:31.07 | poppo | [TK]D-Fender: my php popen looks like this fputs($oSocket, "SetVar: phone=23\r\n") |
| 23:31.07 | [TK]D-Fender | poppo, well I guess you'd better check how you're setting that variable |
| 23:31.08 | [TK]D-Fender | poppo, And what exactly are you sending that TO? |
| 23:31.26 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:32.10 | poppo | then i am going to use it to update mysql database |
| 23:34.33 | russellb | n00dle: well, i didn't make any functional changes ... just added some debug output |
| 23:34.52 | n00dle | It did change how it's behaving though! |
| 23:34.58 | russellb | dangit! |
| 23:35.10 | russellb | every time i try to debug this problem, adding debug output makes it better :) |
| 23:35.19 | russellb | it annoys the crap out of me |
| 23:35.24 | russellb | so, it's some kind of weird race condition ... |
| 23:35.27 | russellb | anyway, i'll take a look. |
| 23:35.27 | n00dle | Hm... timing changes? |
| 23:35.31 | n00dle | K. :) |
| 23:35.39 | russellb | thanks for getting back to me |
| 23:35.52 | n00dle | No prob... glad to help. |
| 23:36.17 | n00dle | I won't get a chance to work on it over the weekend, though. |
| 23:36.45 | n00dle | ...unless I stop by Sunday afternoon, but this weekend's gonna be a busy one. |
| 23:37.29 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:39.32 | russellb | me neither ... |
| 23:39.33 | russellb | ;) |
| 23:39.37 | russellb | i'm off, have a nice weekend |
| 23:39.47 | n00dle | Ciao Russell! |
| 23:48.41 | De_Mon | [TK]D-Fender so, how is someone supposed to set themselves unavailable? logging out of the queue my only option? |
| 23:50.45 | De_Mon | sweet mythbusters doing the tailgating a semi myth |
| 23:54.12 | poppo | [TK]D-Fender: ok i am having same problem as this guy http://lists.digium.com/pipermail/asterisk-users/2004-October/061952.html |
| 23:54.12 | n0n4m3 | does anyone of you guys have a belco BCIP-300 sip phone? |
| 23:56.02 | poppo | #freepbx |
| 23:57.13 | n0n4m3 | i kinda changed the admin password and i forgot how to get in :( |
| 23:57.28 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 23:57.28 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 23:58.52 | [TK]D-Fender | poppo, Guess i'd have to see your entire script to see where you cound have gone wrong. |
| 23:59.36 | [TK]D-Fender | De_Mon, "show application pausequeuemember |
| 23:59.37 | [TK]D-Fender | " |