00:04.31 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
00:06.20 | *** part/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
00:06.38 | *** part/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
00:08.29 | Hmmhesays | is there a good way to tell by callerid if a number came from north america? |
00:10.00 | *** join/#asterisk coppice (n=chatzill@76.155.17.210.dyn.pacific.net.hk) |
00:13.19 | Hmmhesays | I'm trying to figure out how to set callerid with the country code all the time |
00:16.44 | JT | well it's easy to fake in the US, so i dunno |
00:18.14 | Hmmhesays | well the problem I see is some of the NPA's match country codes |
00:18.57 | *** part/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
00:19.13 | pipwerk | so you can't trust incoming callerid's |
00:19.26 | JT | not in the us, that's for sure |
00:20.12 | pipwerk | in that case it has no use normalizing them either I guess |
00:20.40 | Hmmhesays | what I want to know is how the telco's know when to add a 1 to a 10 digit callerid |
00:21.08 | Hmmhesays | example 972 is an areacode in Texas. and 972 is the country code for Israel |
00:21.11 | coppice | look up NPI and TON |
00:21.45 | Hmmhesays | how is that passed across sip though? |
00:21.45 | *** join/#asterisk javar (n=javar@69.79.134.24) |
00:21.52 | JT | sip is not really telco grade |
00:22.42 | Hmmhesays | it would be cool if they passed along type of number |
00:23.07 | *** join/#asterisk CyBeR_GeeK (n=CyBeR_Ge@201.89.88.30) |
00:23.26 | coppice | SIP is a mickey mouse system. how does a telco know how to interpret dialing to isreal from arizona? a prefix tells them its not for texas. |
00:24.05 | *** part/#asterisk lymeca (n=lymeca@unaffiliated/lymeca) |
00:24.13 | Hmmhesays | yes, but I'm talking about normalizing callerid |
00:24.22 | Hmmhesays | how would you distinguish between israel and texas |
00:24.26 | JT | 1 |
00:24.30 | JT | a 1 in front |
00:24.54 | JT | or an international dialling prefit |
00:24.56 | JT | prefix |
00:25.02 | JT | it's 011 there isn't it? |
00:25.04 | Hmmhesays | many providers don't use a 1 if the call is from a north american phone |
00:25.05 | coppice | when you normalised it you should end up with a number plus and NPI and a TON. A leading + for international may be a substitute |
00:25.20 | *** join/#asterisk JimAustin (n=JimAusti@cpe-24-27-122-207.houston.res.rr.com) |
00:25.42 | Hmmhesays | the problem is I don't have any control over the gateway that the call comes into.. |
00:25.51 | Hmmhesays | it comes to me via SIP |
00:25.55 | Hmmhesays | thats the only place I can normalize it |
00:26.14 | Hmmhesays | So I can't tell if the call came from Texas or Israel |
00:26.34 | Hmmhesays | so I don't know if I should +972 or 1972 |
00:26.54 | Hmmhesays | see what I mean? |
00:26.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
00:27.01 | coppice | wouldn't any international number be longer than any local number? |
00:27.25 | JT | i thought international callerid was flakey at best anyway |
00:27.55 | Hmmhesays | I just got a call from the UK, 44 and 10 digits |
00:27.58 | coppice | yeah, often IDD comes through with a totally bogus caller ID, especially with cell phones |
00:28.25 | Hmmhesays | I'm just looking for a decent way to normalize this, I might just have to put up with the country code to north american area code matches |
00:28.42 | Hmmhesays | does sip even have a mechanism for passing along npi and ton? |
00:28.45 | coppice | the UK is a fairly big country, so the number length is distinctive. I'm not sure if all small places have enough digits to be distinctive |
00:29.23 | Hmmhesays | well the good thing about north american numbers is they are length distinctive, you will always have 10 digits |
00:29.38 | JT | unless it's 911 |
00:29.39 | JT | 411 |
00:29.42 | JT | what else |
00:29.49 | coppice | 112 |
00:29.55 | JT | on gsm |
00:29.57 | Hmmhesays | 911 doesn't call you |
00:30.05 | Hmmhesays | 411 doesn't call you |
00:30.08 | Hmmhesays | etc.. |
00:30.12 | JT | heh |
00:30.20 | Hmmhesays | thats only in russia ;) |
00:30.39 | Hmmhesays | does sip have a mechanism for passing ton and npi? |
00:34.39 | JimAustin | newbie question: is asterisk a good solution for creating a conference bridge type solution for a dozen users? eight dialing in through the PBX to a TDM800 type analog card with FXO ports and four coming in as voip calls. or would you all suggest something else? and is the TDM800 the best way to interface to analog ports on the PBX if a digital connection is unavailable? will i have echo problems? do i need to buy hardware that |
00:35.55 | *** join/#asterisk mtaht4 (n=m@165-115-62-200.enitel.net.ni) |
00:36.06 | JT | <PROTECTED> |
00:36.06 | JT | <PROTECTED> |
00:36.18 | JT | yeah, the ircd doesn't like being flooded ;) |
00:36.34 | JT | conferences should work alright if you have zap hardware |
00:37.11 | JimAustin | is the TDM800 zap hardware? or is this something else? |
00:37.15 | JT | yeah |
00:37.18 | JT | uses zaptel |
00:45.13 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
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00:50.21 | *** part/#asterisk mtaht4 (n=m@165-115-62-200.enitel.net.ni) |
00:55.43 | *** part/#asterisk JimAustin (n=JimAusti@cpe-24-27-122-207.houston.res.rr.com) |
01:01.31 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
01:02.21 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
01:08.24 | *** join/#asterisk wundaboy (n=pat@c-24-21-71-88.hsd1.or.comcast.net) |
01:10.02 | wundaboy | when i call my phone from a landline (pstn -> gateway -> mypbx -> my ipphone), i cant hear anything on my ipphone, but i can on my landline phone |
01:10.09 | wundaboy | what am i doing wrong? |
01:10.56 | Strom_M | is your IP phone running SIP? |
01:10.59 | Strom_M | is it behind a NAT? |
01:11.05 | wundaboy | my pbx is behind nat |
01:11.12 | wundaboy | and the phone is on the same network as the pbx |
01:11.25 | wundaboy | its a polycom ip500 |
01:11.26 | Strom_M | is the pbx talking sip to the gateway? |
01:11.35 | wundaboy | the pbx is talking iax |
01:11.46 | JT | to the gateway? |
01:11.48 | wundaboy | yeah |
01:11.56 | JT | check for firewalling to ip phone |
01:12.14 | JT | although 1-way audio is also possible in iax |
01:13.04 | wundaboy | what do you mean? |
01:13.30 | JT | you are describing a 1 way audio problem, are you not? |
01:14.01 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
01:15.26 | wundaboy | i am |
01:15.42 | wundaboy | what would be causing it? a firewall inbetween the pbx and the gateway? |
01:15.53 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
01:16.37 | JT | possibly |
01:16.52 | JT | so do calls to ip phone that don't go to gateway work fine? |
01:18.11 | wundaboy | umm |
01:18.13 | wundaboy | i dont know |
01:18.17 | wundaboy | i only have one phone |
01:19.43 | JT | make an extension that does an Answer, then Echo |
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01:29.28 | *** join/#asterisk watchy (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com) |
01:29.36 | watchy | i got my iphone and its got sip bitches |
01:29.46 | Strom_M | good for you |
01:29.53 | watchy | i'm lying |
01:29.59 | watchy | i just wanted to be cool |
01:30.38 | watchy | iphones already hitting ebay at $1000 |
01:31.13 | JT | what a rip |
01:31.19 | Qwell | heh, nice |
01:31.28 | Qwell | just goes to show that people would have paid 1k |
01:31.29 | Strom_M | i'm strongly considering going to the local apple store and photographing the schmucks waiting outside |
01:31.37 | Qwell | Strom_M: do it |
01:31.41 | JT | Strom_M: nice |
01:31.41 | Strom_M | alright |
01:31.44 | Qwell | well |
01:31.45 | watchy | hah i'm going to buy 2 iphones tommorow for ebay |
01:31.46 | Qwell | you'd have to be |
01:31.48 | Qwell | yeah |
01:31.48 | Qwell | heh |
01:31.53 | Qwell | leave now :p |
01:31.58 | watchy | luckily i live in redneckville usa |
01:32.15 | JT | did you know if you listen carefully enough to the speaker on an iphone, you can hear Steve Jobs saying "haha sucker"? |
01:32.17 | watchy | so i doubt anyones gonna be at my cingular store tommorow |
01:32.21 | watchy | haha |
01:32.33 | Qwell | watchy: You'd be surprised |
01:32.38 | watchy | i actually think the iphone is going to be cool as shit, i dunno if its worth $600 though |
01:33.03 | watchy | the comercials make it look amazing compared to other phones |
01:33.19 | [TK]D-Fender | watchy, the reviews so far say that for the most part, they are deserved |
01:33.41 | [TK]D-Fender | watchy, though its agreed that its connectivity is SLOW. |
01:34.27 | watchy | you know whats fucking retarded about the iphone? |
01:34.34 | watchy | apparently it has a built in SIM card |
01:34.40 | watchy | you can't just take yours and make it work |
01:34.45 | [TK]D-Fender | watchy, I want one that will have good performance on PDF (maps), normal phone, SIP(or IAX) phone, PDA stuff, music, and a DECENT camera (GOOD = bonus) |
01:35.08 | watchy | i have a cingular 8125 right now with windows |
01:35.11 | watchy | its quite nice |
01:35.37 | wundaboy | whats the difference between type=peer and type=user ? |
01:36.44 | watchy | you guys read how much space the iphone OS takes up? |
01:36.58 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
01:37.32 | [TK]D-Fender | wundaboy, Peer = outgoing, user=incoming |
01:37.49 | wundaboy | and friend is both? |
01:38.20 | *** join/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
01:41.07 | wundaboy | ok i keep getting this why i try and dial out: Jun 28 18:40:25 WARNING[12902]: chan_iax2.c:7140 socket_read: Call rejected by 66.227.100.30: No such context/extension |
01:41.15 | *** part/#asterisk workaphobia (n=workapho@ool-44c30ab1.dyn.optonline.net) |
01:41.21 | wundaboy | i cant figure it out, i have incoming working, but i cant figure it out... |
01:43.43 | Strom_M | wundaboy: there's no matching extension or context for the inbound call in your dialplan |
01:45.01 | wundaboy | Strom_M, what do you mean? this is when im dialing from my polycom to my provider |
01:45.25 | Strom_M | well then you're sending them something other than what they're expecting |
01:45.36 | Strom_M | do they want eleven digits? |
01:45.39 | Strom_M | ten digits? |
01:45.47 | wundaboy | they want 11 |
01:45.50 | wundaboy | and im sending 11 |
01:45.57 | Strom_M | 13115552368 ? |
01:46.03 | wundaboy | exactly |
01:46.14 | Strom_M | show me the dial() line |
01:46.27 | wundaboy | exten => _NXXNXXXXXX,2,Dial(IAX2/jnctn-out,1${EXTEN}) |
01:46.37 | Strom_M | wrong |
01:46.38 | wundaboy | <PROTECTED> |
01:46.55 | Strom_M | IAX2/jnctn-out/NUMBER-GOES-HERE,options) |
01:47.35 | wundaboy | exten => _NXXNXXXXXX,2,Dial(IAX2/jnctn-out/1${EXTEN}) ? |
01:47.39 | Strom_M | yup |
01:48.01 | wundaboy | oh man |
01:48.04 | wundaboy | now i feel like a moron |
01:48.07 | *** join/#asterisk limaunion (n=limaunio@OL160-170.fibertel.com.ar) |
01:48.11 | wundaboy | thanks man |
01:49.57 | Strom_M | welcome |
01:55.20 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
01:57.40 | Strom_M | catsex |
01:58.34 | Hmmhesays | sure |
02:04.50 | watchy | anyone wanna loan me the newest polycom firmware? i will hug you |
02:05.02 | JT | will you give it back? |
02:05.24 | *** join/#asterisk Greenbox (n=Brett@user-24-214-124-177.knology.net) |
02:05.28 | watchy | yes |
02:05.54 | JT | cool |
02:06.14 | watchy | now give it up before i stab you |
02:07.23 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
02:07.50 | watchy | i dont see any firmware but i see you getting stabbed jt |
02:07.50 | watchy | :( |
02:08.10 | JT | i don't have it |
02:08.27 | watchy | the story of my life |
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02:17.28 | flenders | hey, what does the "when convenient" option does when restarting again? |
02:17.45 | Strom_M | when call traffic drops to 0 |
02:17.55 | flenders | thanks |
02:20.04 | flenders | oh shit, when you do a restart when convenient, you can't do anything else on the CLI |
02:20.08 | flenders | is it a bug? |
02:22.28 | *** join/#asterisk riddlebox (n=james@75-132-245-209.dhcp.stls.mo.charter.com) |
02:23.03 | Strom_M | i dont know. i always just wait till call traffic drops to 0 and restart now |
02:23.52 | JT | flenders: haha, that sounds non-optimal |
02:26.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
02:29.06 | flenders | it's terrible |
02:30.24 | flenders | so if you make any changes on zapata.conf, that can wait until there are no calls and you do a 'restart when convenient' but then you realize you need to reload the extensions, youre screwed |
02:30.30 | flenders | you have to wait until it restarts |
02:30.44 | Strom_M | you could also just do a zap restart |
02:33.51 | flenders | but then I would drop all the current calls, souldnt I? |
02:33.55 | flenders | wouldn't I? |
02:34.25 | JT | yeah |
02:34.37 | JT | i think |
02:35.03 | Strom_M | if they're zap calls |
02:36.19 | JT | flenders: so if you attach a console to it, it's non-responsive? |
02:36.33 | *** join/#asterisk Krooks (n=Krooks@202.184.116.210) |
02:36.37 | Krooks | Hello |
02:36.39 | flenders | any commands you type in, you get no response |
02:36.49 | flenders | even a 'core show uptime' |
02:36.49 | Strom_M | hello |
02:37.00 | Krooks | anyone using asterisk on centos 5? |
02:37.19 | Strom_M | no one has ever used asterisk on centos in the entire history of hot dogs |
02:37.33 | Krooks | hehe |
02:38.02 | Krooks | Strom_M: you running on cetnos 5 ? |
02:38.06 | Strom_M | no |
02:38.09 | Strom_M | i use debian |
02:38.13 | Strom_M | but what's your question |
02:38.31 | Strom_M | or did you just want to form a knitting club |
02:38.38 | Krooks | Well I want to do iton centos 5 .I just want to know that its doable before I embark |
02:38.44 | Strom_M | yes it's doable |
02:38.53 | Krooks | ok then |
02:39.32 | Krooks | Its would be nice to get a yes from somebody who _has_ done it on centos 5 though |
02:40.29 | Strom_M | i've done it on centos |
02:40.29 | Strom_M | i just like debian better |
02:40.40 | Krooks | oh ok |
02:40.54 | JT | asterisk runs fine on linux |
02:40.59 | JT | centos is a linux distribution |
02:42.03 | Strom_M | the jizzy mcspurtface operating system |
02:42.11 | Strom_M | free as in boners |
02:42.34 | Krooks | Strom_M : should I svn checkout the trunk or should I just go with the tarred ones |
02:42.52 | Strom_M | don't use trunk in production |
02:42.55 | JT | just use a release |
02:42.59 | Strom_M | if anything, svn checkout the release branch |
02:42.59 | Krooks | ok |
02:43.01 | JT | especially if it's your first time |
02:43.04 | Strom_M | but the tarballs are a good idea |
02:43.27 | Krooks | ok, 1.2.19 or 1.4.5 ? |
02:43.32 | Strom_M | 1.4.5 |
02:43.38 | Krooks | ok I'm on my way |
02:43.41 | JT | 1.2 :P |
02:43.42 | Strom_M | whee |
02:43.50 | Strom_M | JT: fuddy-duddy |
02:44.15 | JT | eh, might consider 1.4 in prodution when it reaches .8 ;) |
02:45.06 | *** join/#asterisk Ergose (n=domainop@cpe-065-190-118-012.triad.res.rr.com) |
02:47.26 | flenders | funny, that I've been running 1.4.x here for 4 months, and it's all good |
02:47.35 | Strom_M | so yay, tomorrow i get to talk to a potential client about ripping out trixbox :) |
02:48.23 | *** join/#asterisk sosoriosv (n=salvador@190.86.82.174) |
02:48.32 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
02:48.43 | Krooks | ok the zaptel and libpri are drivers for the fx cards , right ? |
02:48.59 | JT | libpri is only required for digital cards |
02:49.31 | Krooks | ok |
02:49.36 | Krooks | and zaptel too |
02:50.18 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
02:50.58 | *** part/#asterisk sosoriosv (n=salvador@190.86.82.174) |
02:51.26 | JT | yes, zaptel is also needed by FXO/FXS cards |
02:52.36 | *** join/#asterisk NovceGuru (n=asdf@oh-71-50-248-25.dhcp.embarqhsd.net) |
02:52.54 | [hC] | interesting. I thought i was having network problems but it turns out it was just a bug in OSX's tftp client. |
02:53.05 | [hC] | it cannot download files smaller than 512bytes. |
02:53.21 | JT | sounds like an apple |
02:53.37 | *** join/#asterisk Ergose (n=domainop@cpe-065-190-118-012.triad.res.rr.com) |
02:54.08 | [hC] | on an unrelated side note, any of you guys done a factory reset on a cisco 7970 and have it sit at "Upgrading" --seemingly doing nothing, and not requesting DHCP? |
03:04.49 | *** part/#asterisk Ergose (n=domainop@cpe-065-190-118-012.triad.res.rr.com) |
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03:23.07 | NovceGuru | <PROTECTED> |
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03:23.20 | vn | I hate music on hold :\ |
03:23.27 | vn | especially when played during 30mins |
03:26.43 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:26.58 | Ergose | Has anyone here ever had a problem with IVR not wanting to go more than one level deep before returning back to the home message? |
03:27.22 | Ergose | Just a stab in the dark... |
03:30.19 | Ergose | I believe the bug to be with the asterisk side of things because FreePBX seems to be configuring the extensions_additional.conf correctly, and asterisk does seem to acknowlege them... |
03:31.43 | Ergose | however for whatever reason, I press 1 for first option 1 or 2 for second and instead of going into the 3rd level voicemail menu, I get sent back to 1. |
03:32.10 | *** join/#asterisk wunderkin (n=wunderki@dslstat-ppp-95.fastq.com) |
03:32.19 | Krooks | If I'm not going to use my asterisk over a pstn line, then I don't have to worry about zaptel and libpri, right ? |
03:33.13 | Ergose | don't think so, but I think freepbx was being pissy without the zaptel module |
03:33.14 | JT | Krooks: unless you need zap timing for meetme conferences or iax2 trunking or music on hold, then you need zaptel |
03:33.29 | Ergose | but it was the dummty module |
03:34.01 | JT | yeah but freepbx is evil ;) |
03:34.10 | Ergose | yeah tell me about it |
03:34.21 | Krooks | Is zaptel a hardware thingy, and the zaptel in asterisk is the driver for it ? |
03:34.26 | Ergose | I still can't tell if my prob is with freepbx or asterisk |
03:34.41 | JT | zaptel is for interfacing to hardware normally |
03:34.43 | Krooks | I mean my box is just a standard |
03:34.50 | Krooks | I mean my box is just a standard PC |
03:35.02 | Krooks | I don't have anything called a zaptel in my PC |
03:35.15 | JT | you don't get cards called a zaptel |
03:35.21 | JT | zaptel is the name of the software |
03:35.30 | JT | there's plenty of info online about this |
03:35.36 | Krooks | ok, thats exactly what I want to know |
03:36.01 | Krooks | and libpri is the same thing ? |
03:36.15 | JT | no |
03:36.16 | VJFROMGT | how can i adjust speex codec for 6 kbps? |
03:36.22 | JT | it's a different library |
03:36.43 | JT | for pri circuits, they still need zaptel too |
03:36.44 | Krooks | no I mean its not a driver for a hardware thing |
03:36.56 | JT | what? |
03:37.02 | JT | ask your questions more clearly |
03:37.08 | Krooks | ok nvm |
03:37.12 | Krooks | thanks |
03:37.23 | JT | read the book |
03:37.24 | JT | ~thebook |
03:37.35 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
03:37.36 | Ergose | so JT, any idea as to why I'm getting stuck in a loop on the IVR menus? I'm almost figureing something when wrong on the compile, but everything else works like a charm... |
03:37.42 | Krooks | I should read the manual first before I annoy someone and get kicked out :) |
03:38.20 | Ergose | lol |
03:39.19 | Ergose | people are usually cool on that kindof thing... atleast they used to be.... haven't been on IRC since way back |
03:40.12 | Ergose | brb, going for a smoke before I kill my box... |
03:40.49 | Krooks | One last try, I have a standard PC. you know, sound card , video card etc. nothing extra. I just wanna know if I need to compile zaptel and libpri also. Thats all. I just wanna play with asterisk. Not for any production. Just playing. |
03:42.09 | Greenbox | Krooks, the only reason you need zaptel is if you have one of the hardware cards |
03:42.17 | Greenbox | from digium |
03:43.48 | Krooks | Greenbox: ok perfectly understand now. |
03:44.10 | Corydon76-home | Or if you need a timing source for one of the modules which requires it |
03:44.19 | Corydon76-home | such as app_meetme |
03:44.47 | Ergose | yeah, all I was saying was that if using freepbx I had to load the dummy module, but that's more a freepbx quirk... |
03:45.22 | Krooks | Corydon76-home: you going to confuse me. If I don't have one of the zaptel hardware cards, then why would all that matter to me. |
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03:45.37 | JT | i explained when you need zaptel already, i guess it just needs to be repeated sometimes |
03:45.40 | JT | Krooks: for timing |
03:45.49 | JT | Krooks: please read the book |
03:46.06 | Corydon76-home | Hardware timing provides a more reliable source of evenly spaced interrupts |
03:46.15 | Corydon76-home | Software interrupts just can't stack up |
03:46.36 | JT | some of it is historical anyway |
03:47.02 | Krooks | So even if I don';t have a zaptel hardware car, I still can make use of the zaptel module. Is that what you are saying ? |
03:47.10 | Corydon76-home | Correct |
03:47.18 | JT | yes, through ztdummy |
03:47.33 | Krooks | which is in the zaptel module. |
03:47.44 | JT | no, zaptel is a package |
03:47.47 | Ergose | yeah, I haven't had the time to go through the manual, I'm just trying to help the other admin couse I'm the "Linux guy..." :) |
03:47.51 | Krooks | oko zaptel package |
03:47.51 | Corydon76-home | It's a separate module, but it interfaces with the zaptel module |
03:47.54 | JT | the zaptel kernel module is a part of that package |
03:48.20 | Corydon76-home | zaptel provides a consistent interface across multiple card types |
03:48.35 | Krooks | ok I i'll jsut install zaptel and libpri. It won't hurt , right. |
03:48.43 | Corydon76-home | No, it won't hurt |
03:48.51 | JT | very little chance you'll need libpri though |
03:48.52 | Krooks | ok thats all I want to know |
03:49.13 | Krooks | ok so forget libpri and install zaptel. Great. |
03:49.22 | Ergose | sounds good |
03:50.53 | friedrich| | good night. |
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03:51.09 | Ergose | goodnight |
03:53.03 | Krooks | so I need the kernel source before I install zaptel , right ? |
03:53.03 | Ergose | I just don't get it... the config is linking to the correct IVN sections, so I know the flow is right and it still doesn't work past the second menu... sends me back to the first instead of the third even though should goto 3rd menu... |
03:53.24 | Krooks | nvm |
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03:55.22 | ManxPower | IVR (Interactive Voice Response) |
03:56.44 | wundaboy | how do i setup a menu system? |
03:57.11 | Ergose | lol I'm having prob with our ivr... |
03:57.48 | wundaboy | that sucks |
03:57.50 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
03:58.16 | Ergose | are you using freepbx or just asterisk? |
03:59.13 | Corydon76-home | Most IVRs are coded with Read. For a simple menu, though, WaitExten is fine |
03:59.27 | wundaboy | im using asterisk |
03:59.38 | wundaboy | i just want a simple menu |
04:00.44 | Ergose | ooh, haven't heard about Read... maybe that'll help... I could compare the ivr sections generated and see a comparison |
04:00.57 | Ergose | you got the web address on Read? |
04:02.37 | Ergose | heh... nvrmind stupid question... |
04:02.44 | Ergose | lol |
04:05.36 | wundaboy | ok so i got a question |
04:05.50 | wundaboy | in extensions.conf is there any significance to the [default] context? |
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04:06.49 | rob0 | IIUC not necessarily, but it's the default target for places like sip.conf, iax.conf and others. |
04:07.10 | wundaboy | ok thanks |
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04:08.42 | troy- | when a call comes in on to a Cisco 7941 from Asterisk, for the first 2 seconds there is silence before you can talk, why is that? |
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04:29.52 | wundaboy | if i have someone in an IVR is it possible to then ring a phone on the pstn and transfer the call to that ringing? |
04:29.58 | wundaboy | what command would i use with that? |
04:30.34 | Strom_C | Dial() |
04:30.37 | wundaboy | ok |
04:32.34 | wundaboy | how do i specify how long it will wait for a dtmf response? |
04:32.59 | wundaboy | waitexten? |
04:36.00 | wundaboy | anyone? |
04:36.54 | Strom_C | <PROTECTED> |
04:41.39 | MooingLemur | troy-: maybe it's round-trip lag between the phone and the pbx or other end of the call? |
04:42.31 | troy- | MooingLemur, hmm they are both on the same switch |
04:43.01 | Aces1Up | what are some good suggestions for cards, if i need 6 co lines as well as 6 station lines non IP.? |
04:43.10 | JT | troy-: probably a sip/rtp setup problem |
04:43.16 | Strom_C | Aces1Up: digium tdm2400 |
04:43.28 | troy- | JT, how do i go about verifying that? |
04:43.37 | JT | troy-: packet sniffing perhaps |
04:45.39 | Aces1Up | strom do i need echo cancellation? |
04:46.03 | Strom_C | Aces1Up: how long are your circuits? |
04:46.32 | Aces1Up | length? less than 200ft. |
04:46.40 | Qwell | Strom_M: I got a video call going through the appliance :D |
04:46.49 | Strom_C | your telephone company's CO is less than 200 feet away? |
04:46.55 | Strom_C | qwell: sweet |
04:47.02 | Aces1Up | strom ohh i don't know. |
04:47.12 | Strom_C | i'm still trying to figure out how to get the craft port to respond to my input :/ |
04:48.59 | Aces1Up | strom looking for solution for 6fxo/6fxs with 6-analog handsets for less than 2300.00 can it be done? |
04:49.21 | Aces1Up | aces including asterisk box. |
04:49.23 | Strom_C | Aces1Up: how much is the TDM2422B? |
04:49.26 | JT | haha |
04:49.30 | JT | including asterisk box |
04:49.31 | Aces1Up | 1515.00 |
04:49.33 | JT | if it's free |
04:49.35 | JT | maybe |
04:49.41 | sweeper | uh |
04:49.42 | sweeper | OR |
04:49.47 | Strom_C | Aces1Up: let me offer you one piece of advice |
04:49.49 | Strom_C | don't be a cheapass |
04:49.51 | sweeper | you could buy two 8-port gateways |
04:49.56 | sweeper | $300 each |
04:50.01 | sweeper | one fxo, other fxs |
04:50.08 | JT | $300, from where? |
04:50.09 | sweeper | handsets are what, $20? |
04:50.19 | Strom_C | $20 for the crappy desk sets |
04:50.20 | sweeper | JT: the internets! |
04:50.29 | Aces1Up | ahh well. |
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04:50.35 | JT | better not be much more for handsets or you mayaswell go sip |
04:50.35 | sweeper | I think the grandstreams go for less than 300 |
04:50.40 | Aces1Up | my client wants to go with a toshiba system. |
04:50.41 | JT | yeah |
04:50.48 | sweeper | 300 is for the audiocodes |
04:50.50 | JT | but using a grandstream for a business is braindead |
04:50.53 | sweeper | or the linkys |
04:50.59 | sweeper | so anyways |
04:51.10 | sweeper | that leaves $1500 for the asterisk box |
04:51.10 | JT | can you get 8 port linksys ATAs? |
04:51.16 | sweeper | oh yea |
04:51.42 | sweeper | they require some tweaking |
04:51.46 | sweeper | but they work |
04:52.00 | JT | what model |
04:52.06 | Aces1Up | whats more reliable the gateways? |
04:52.11 | sweeper | SP4000? something like that |
04:52.26 | sweeper | Aces1Up: more reliable would be a channel bank + t1 card |
04:52.46 | sweeper | that would run you $1500 for both, tho |
04:52.51 | sweeper | eh |
04:52.51 | Aces1Up | lol |
04:52.59 | sweeper | you can build a decent system for $800 |
04:53.05 | sweeper | I mean, for six phone lines |
04:53.27 | Aces1Up | yeh, thas all they need. |
04:53.31 | Aces1Up | won't be growing at all. |
04:53.37 | sweeper | you could do that |
04:53.53 | Aces1Up | have a recommendation for the gateways? |
04:54.05 | sweeper | not grandstream |
04:54.16 | sweeper | people say they like audiocodes, but I've not had any luck with them |
04:54.36 | troy- | JT, any idea what this error means: chan_iax2.c:3792 iax2_send: No private structure for packet? |
04:58.21 | wundaboy | so, when i have someone make a choice (press 1 for this, press 2 for this, etc), do i just put as one of the extensions, a Dail command? |
04:58.39 | wundaboy | for instance: exten => 2,1,Dail(IAX2/jnctn-out/15415555596) |
04:59.08 | JT | if you want to dial someone, yes |
04:59.21 | troy- | JT, any idea on that error? |
05:00.09 | JT | no |
05:00.19 | troy- | seems really strange, its continuous |
05:00.20 | JT | except you some sort of packet error |
05:00.32 | troy- | how would i figure out what the problem is? |
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05:01.36 | watchy | anyone wanna loan me the newest polycom firmware? |
05:01.57 | Strom_C | watchy: your reseller should be able to provide that to you |
05:02.21 | watchy | voipsupply is my reseller they are a bitch to get that shit from |
05:02.42 | Strom_C | a good reason not to buy from them again, i suppose |
05:03.02 | watchy | well they are cheap and get me equipment fast |
05:03.14 | wundaboy | watchy, loan? |
05:03.22 | wundaboy | last time i checked you could just copy it... |
05:03.31 | watchy | wunda: well copy it then |
05:03.31 | wundaboy | if i had it i would give it to you |
05:03.37 | watchy | cp firmware firmware2 |
05:03.41 | wundaboy | i have it on my phone |
05:04.01 | wundaboy | got it from my friend, just connected to his ftp server with the phone |
05:04.09 | watchy | ah |
05:04.19 | watchy | well grab it off his ftp :/ |
05:05.39 | wundaboy | i dont have that info anymore, it was a like a month ago... |
05:06.10 | watchy | i have indigestion and its making me sad |
05:07.24 | cybergypsy | i have just upgraded from asterisk 1.2 to 1.4 and the ilbc codec sounds much worse when calling through gizmo service , anyone else getting that problem ? |
05:13.08 | [TK]D-Fender | watchy, No, they are EXPENSIVE, and give shit service apparently |
05:13.31 | Aces1Up | what are the better brands for fxo gateways? |
05:14.07 | [TK]D-Fender | Aces1Up, AudioCodes, MediaTrix, MultiTech (meh), Patton |
05:14.15 | Aces1Up | thanks tkd. |
05:14.43 | watchy | gimme some service tk |
05:14.59 | JT | ? |
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05:16.06 | [TK]D-Fender | watchy, I don't workt he red-light district anymore. |
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05:21.37 | Aces1Up | tkd grandstream aren't good for gateways? |
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05:21.53 | JT | they aren't good for anything |
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05:22.03 | JT | how hard is it to spell his nick properly, btw :P |
05:22.23 | Strom_C | type [t and press TAB |
05:22.37 | troy- | [TK]D-Fender |
05:22.38 | troy- | there :P |
05:23.14 | Strom_C | it amazes me how few people are aware of tab complete |
05:23.27 | JT | yes, not enough cli use |
05:23.31 | JT | more cli osmosis required |
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05:23.47 | [TK]D-Fender | Aces1Up, I don't know how many times its going to take to get this through your head : |
05:23.48 | [TK]D-Fender | ~gs |
05:24.02 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
05:24.04 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
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05:24.35 | Strom_C | Aces1Up: let me offer you one piece of advice again: DON'T BE A CHEAPSKATE |
05:26.16 | Aces1Up | daaaaaaaaaaaaaang |
05:26.32 | Aces1Up | check this out though yehaaaa http://www.asteriskguru.com/review_hardware_7.php |
05:27.15 | Strom_C | yeah? and see the bug in the middle of the review where it doesn't fucking work? |
05:27.24 | Strom_C | that's a pretty major bug right there :) |
05:27.38 | Aces1Up | heh, jeez. you guys are so mean. |
05:27.44 | Aces1Up | i know just kiddin. |
05:27.54 | Aces1Up | go with the audiocodes. |
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05:29.25 | Strom_C | cocks, etc. |
05:30.03 | JT | i want to know what knok wrote that asteriskguru review |
05:30.05 | JT | e low price (approximately 450$) of the device makes it even more intriguing. The medium price of an AT is around 100$ |
05:30.17 | JT | dollars symbols after the number is a major sign of retardation |
05:30.21 | Aces1Up | jt heh |
05:30.32 | Strom_C | not to mention mixing up medium and median |
05:30.39 | Strom_C | they're not even homophones |
05:30.44 | Strom_C | they're just similarphones |
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05:30.51 | Qwell | hell, they aren't even phones |
05:30.57 | Qwell | they're like...similargrandstreams |
05:31.01 | Strom_C | hahahahaahahahahaha |
05:31.04 | Strom_C | qwel++ |
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05:31.08 | Qwell | I *ALMOST* went for the other one |
05:31.12 | Strom_C | s/l/ll/ |
05:31.13 | Qwell | but, decided not to |
05:31.19 | Aces1Up | hey, you guys are so funny haha |
05:31.26 | Qwell | it would've been in bad taste |
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05:31.50 | Strom_C | qwell: I think after "cocks, etc." there's little you can do in the way of bad taste |
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05:32.20 | Qwell | well, the alternative, would've been homograndstreams |
05:32.20 | Strom_C | heheh |
05:32.23 | Qwell | and see, look, now the folks who just came in on a netsplit are like "wtf?" |
05:32.35 | Aces1Up | hey good one qwell hahaha yeh hahaha |
05:33.04 | russellb | netsplits are such teh silly |
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05:33.10 | Aces1Up | :ponders sarcasm in irc. |
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05:33.49 | [TK]D-Fender | Aces1Up, So why are you even considering a SIP FXO gateway? |
05:34.05 | Strom_C | because someone told him it was a better idea than a pci card |
05:34.32 | Ergose | yeah, I haven't been on irc a while, but that was a massive dump of people.... |
05:34.55 | [TK]D-Fender | Ergose, welcome to FreeNode..... where Nodes..... run FREE. |
05:35.06 | Strom_C | FREEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE |
05:35.11 | Ergose | rofl |
05:35.51 | Strom_C | tonight's episode of #asterisk is presented in technicolor |
05:36.09 | Aces1Up | http://voipspeak.net/index.php?option=com_content&task=view&id=113&Itemid=27 |
05:36.13 | wundaboy | how do i get voicemail registration to work with my polycom ip500? |
05:36.17 | wundaboy | like |
05:36.22 | wundaboy | make the light lightup when i have a voicemail |
05:36.43 | Ergose | lol |
05:36.45 | Strom_C | "The GXW-4104 is a slick looking box that is certainly one of the best looking products to ever have come out of Grandstream's engineering department. We wanted to put the GXW-4104 to the test so we decided to put it to use in a production environment in a small business using a trixbox server." |
05:36.47 | wundaboy | and makethe messages button connect to the voicemail system (rather than just having it dial an extension) |
05:36.57 | Strom_C | they care about looks and they're using trixbox |
05:37.00 | Strom_C | i'd run from this review |
05:37.05 | Aces1Up | lol |
05:37.24 | Aces1Up | where are the good reviews at? |
05:37.37 | Strom_C | Aces1Up: listen: forget Grandstream |
05:37.39 | Strom_C | ok? |
05:37.52 | Aces1Up | ok. |
05:37.56 | Strom_C | we've been working with this stuff for a long time and we tend to know what we're talking about |
05:37.56 | Aces1Up | yes masta |
05:38.22 | Aces1Up | ok doke. |
05:38.25 | Aces1Up | no problem. |
05:38.32 | Aces1Up | audiocodes it is . |
05:39.16 | Strom_C | i still say you should go for a TDM2422B, or a TDM808B and some IP phones |
05:39.38 | Aces1Up | well kids, it has been fun, thanks for putting up with my questions this evening, bid you all a farewell and goodnight. |
05:39.46 | Aces1Up | strom it would break the budget. |
05:39.58 | Strom_C | re-evaluate your budget |
05:40.06 | Strom_C | being a cheapskate == headache |
05:40.08 | Ergose | night ace |
05:40.08 | Aces1Up | lol wish i could.. |
05:40.14 | Aces1Up | not my choice. |
05:40.26 | [TK]D-Fender | Aces1Up, Yes it is. |
05:40.34 | Strom_C | well then you come back and say "we can't do it with this budget" |
05:40.51 | [TK]D-Fender | exactly |
05:41.00 | Aces1Up | yep, thas whats i'm a gonna say. |
05:41.36 | JT | "we can do it, but it will be halfarsed and of poor quality and reliability, it will cost us in the long run" |
05:41.58 | Aces1Up | jt, sweet... |
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05:42.39 | Aces1Up | then i can say hey, why, don't you give me 3 more thousand dollars, and then we'll go get some cheesburgers. |
05:42.59 | flenders | Aces1Up: what's the budget? |
05:43.28 | [TK]D-Fender | flenders, "as cheap as possible" <- DUH |
05:43.32 | Aces1Up | well, they are getting the toshiba cix40 system for 3300.. just looking to quote 1000.00 less than that :) |
05:43.49 | flenders | how many handsets? how many lines? |
05:43.52 | [TK]D-Fender | Aces1Up, And what eother equipement is on this quote of yours? |
05:43.57 | flenders | sorry I missed it if you've already said |
05:44.19 | Aces1Up | 6fxo, 6 handsets, voicemail, sla needed. |
05:44.31 | [TK]D-Fender | Aces1Up, SLA? |
05:44.39 | Aces1Up | yes. |
05:44.45 | [TK]D-Fender | Aces1Up, meaning? |
05:44.54 | Aces1Up | Shared Line Appearances |
05:45.03 | [TK]D-Fender | Aces1Up, LOL. |
05:45.11 | [TK]D-Fender | Aces1Up, SO not happening! |
05:45.16 | JT | err |
05:45.16 | JT | sorry |
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05:45.25 | JT | HOW will SLA work on analogue phones? |
05:45.33 | [TK]D-Fender | Aces1Up, "Put down the crack pipe!" (c) JerJre |
05:45.43 | Aces1Up | jt well i found that out, soo will run ethernet... so back to ip phones. |
05:45.53 | JT | ah |
05:45.56 | Aces1Up | tkd its not cracked. |
05:45.58 | JT | you only need a TDM800 then |
05:46.01 | JT | not TDM2400 |
05:46.02 | Aces1Up | lol |
05:46.13 | [TK]D-Fender | Aces1Up, Your entire plan (yes, ALL OF IT) is on CRACK. |
05:46.15 | JT | and 6 Polycom IP320 or IP330 |
05:46.26 | Aces1Up | my pipe is not cracked :) |
05:46.37 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net) |
05:46.39 | Aces1Up | jt.. sweet.. lemme check da numbaz. |
05:46.43 | JT | i think it can be done, if the price doesn't involve your labour costs and a brand new server |
05:46.48 | JT | OK MAN |
05:47.05 | Aces1Up | jt yep, it does involve my labor, is only way i get paid, and the asterisk box. |
05:47.13 | [TK]D-Fender | Aces1Up, You have set yourself up for end-to-end FAILURE. Yuo have no clue about the equipment, nor even *'s capabilities. |
05:47.28 | Aces1Up | tkd I CAN DO IT! |
05:47.29 | Aces1Up | lol |
05:47.47 | [TK]D-Fender | Aces1Up, "Denial.... it's not just a river in Egypt..." |
05:47.50 | Aces1Up | give me a chance, I won't make a fool out of you.. yeh tom cruise, DAYS OF THUNDER baby :) |
05:48.29 | [TK]D-Fender | Aces1Up, Thunder is actually a short way of saying "hot air that likes to make a big noise" |
05:49.12 | Aces1Up | tkd, i don't think i'm full of hot air... i know i don't know that much, and am well aware of my limitations... |
05:49.30 | flenders | Aces1Up: seriously, don't do it too cheap... you'll regret it. |
05:49.36 | flenders | personal experience |
05:49.36 | [TK]D-Fender | Aces1Up, You just don't know that of they gear & solutions you are basing your plans on. |
05:49.57 | Aces1Up | tkd, yeh, your right. i haven taken your advice and will not go with the cheapo solution. |
05:50.12 | Strom_C | http://www.jerkcity.com/jerkcity300.html |
05:50.15 | [TK]D-Fender | Aces1Up, More than that.... SLA <- Not happening. |
05:50.16 | JT | Aces1Up: [tk <tab> :D |
05:52.19 | Aces1Up | well alrighty, you kids don't stay up too late.. |
05:52.25 | Aces1Up | talking bout phones and what not.. |
05:52.46 | Ergose | lol yeah, no sleep for the admins... |
05:53.02 | *** part/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
05:53.20 | Strom_C | what a nub |
05:54.06 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
05:55.24 | flenders | wow, I'll never leave the channel |
05:55.25 | flenders | :) |
05:55.25 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
05:55.25 | *** mode/#asterisk [+o denon] by ChanServ |
05:59.30 | troy- | Strom_C, what an atrocity |
05:59.49 | [TK]D-Fender | $850 Sangoma 86 FXO + HWEC. Phones IP 430 : 6 * $150 = $900, + Server ($1000). can work. |
05:59.58 | Strom_C | [TK]D-Fender: he's already gone |
06:00.07 | [TK]D-Fender | Strom_C, I know. |
06:00.28 | troy- | i cant wait to pay bell $44 for a POTS line |
06:00.46 | [TK]D-Fender | Strom_C, Just proving a point. But his approach sucks, and SLA is NOT happening, and he is all but certainly incompetant to set it all up. |
06:00.52 | JT | at least i'd be happy to be smashed in the face with a mediatrix or adtran, i'd have something usefuul to keep at the end ;) |
06:01.00 | JT | [TK]D-Fender: it had to include his labour costs too |
06:01.04 | troy- | haha yeah |
06:01.11 | troy- | unless he works for free |
06:01.17 | [TK]D-Fender | JT : You could cheap out the server..... |
06:01.29 | JT | yeah, second hand deskto p:P |
06:01.34 | [TK]D-Fender | JT : I could get a passable box for $100 techinally. SFF to boot. |
06:01.34 | troy- | dell has servers starting at $400 right now |
06:01.59 | JT | heh, i think for really small setups without transcoding loads, embedded is the way to go |
06:02.03 | troy- | if configured reasonable you could make it for $700 |
06:02.07 | JT | brand new gear without 32838734 moving parts |
06:02.20 | JT | also takes up less space |
06:02.28 | [TK]D-Fender | JT : Embedded actually COSTS more. |
06:02.41 | JT | [TK]D-Fender: more than what? |
06:02.42 | [TK]D-Fender | JT : And doesn't scale. Never waste money on a DEAD END |
06:02.45 | troy- | JT, usually i buy rackmount boxes with rails and just stick it in a rack with the rest of the company's crap |
06:02.49 | [TK]D-Fender | JT : that a REAL PC |
06:03.05 | JT | pbxes are appliances,a ll some business need are an appliance |
06:03.09 | troy- | usually i spend roughly $1,200 |
06:03.21 | JT | [TK]D-Fender: really depends on the specs of the PCs |
06:03.29 | JT | cheap ones will be much less reliable |
06:03.32 | [TK]D-Fender | JT : These days? SERIOUSLY. |
06:03.40 | JT | [TK]D-Fender: not performance wise |
06:03.43 | JT | reliability |
06:03.50 | [TK]D-Fender | JT : uhhhh SURE... |
06:03.52 | JT | also heat and power consumption and footprint are factors |
06:03.53 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
06:04.05 | troy- | JT, usually i will buy a P4 3.0 1G ram 2*80GB Raid 1 Dual GigE 1U Rails |
06:04.08 | [TK]D-Fender | JT : My server runs rock-sold for the past efw years 24x7 |
06:04.12 | JT | get a shitty desktop, expect shitting reliability |
06:04.19 | JT | [TK]D-Fender: mine too, but they're server grade |
06:04.25 | troy- | exactly :) |
06:04.26 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
06:04.29 | phix | hey |
06:04.33 | phix | Jun 29 16:03:47 WARNING[8482]: codec_g729.c:249 lintog729_framein: Out of G.729 Encoder Licenses! |
06:04.45 | JT | troy-: if you only need 6 extensions, getting a redundant everything server may seem like overkill |
06:04.47 | [TK]D-Fender | JT : its not a server unless its IBM's "iron" :) Everything else is a glorified PC. |
06:04.52 | troy- | time to hand over everything you own to digium phix |
06:04.57 | JT | embedded is a niche |
06:05.02 | JT | many won't fall into it |
06:05.05 | phix | How expensive are ^.729 encoder licences? |
06:05.06 | JT | but it's there |
06:05.16 | troy- | JT, disagree. a redundant everything box will fail a lot less |
06:05.24 | Strom_C | phix: $10 per channel |
06:05.27 | [TK]D-Fender | JT : And there are 2 headed dogs.... Do you want one PERSONALLY? |
06:05.37 | troy- | i know plenty of companies with maybe 4 desks, that push serious sales / revenue and require high availability |
06:05.48 | JT | troy-: you could buy a whole spare embedded box for the price of some servers |
06:05.52 | [TK]D-Fender | ok, bed time for sure..... |
06:05.56 | JT | troy-: others don't have those requirements |
06:05.57 | [TK]D-Fender | later |
06:06.02 | troy- | night! |
06:06.10 | troy- | JT, guess it depends on the target market |
06:06.37 | troy- | i'm just laughing that i pay $10 for a 4 channel DID, and bell is charging $44 for single analog POTS |
06:07.02 | Strom_C | well that DID doesn't involve the leasing of the provider's outside plant |
06:07.12 | troy- | bell owns the plant |
06:07.26 | Strom_C | missing the point |
06:07.45 | phix | Strom_C: really? where from? |
06:07.51 | Strom_C | phix: digium |
06:07.56 | troy- | yes i understand that a copper pair running between the CO and an office building costs money to install and maintain |
06:08.08 | phix | Strom_C: ok thank you |
06:08.27 | phix | Strom_C: $10 USD right? |
06:08.30 | troy- | ya |
06:09.01 | troy- | Strom_C, but that copper is a few decades old and was buried with thousands of other pairs |
06:10.41 | troy- | once that copper terminates at the CO its handed off on to Bell's SONET / ATM backbone and is basically the same thing as voip |
06:11.01 | Strom_C | lol |
06:11.18 | troy- | your argument is simply that the last mine costs money |
06:11.22 | troy- | *mile |
06:12.06 | Strom_C | as does the SONET/ATM network, the class 4/5 switching infrastructure, the personnel to support it, etc etc etc etc |
06:12.09 | *** join/#asterisk harlequin516 (n=sham@styk.net) |
06:12.21 | Strom_C | whereas your DID provider is probably three guys in a shack in calgary |
06:12.47 | troy- | you're telling me that carrier grade voip gear and personnel costs less? |
06:13.06 | troy- | bell charges CLECs a few cents for every DID, plus a few dollars per channel |
06:13.36 | troy- | bye bye :) |
06:13.43 | Strom_C | cocks |
06:13.48 | Strom_C | etc. |
06:13.58 | troy- | yes i have one |
06:14.35 | harlequin516 | I have this problem that I have now finally traced. I can only get a voice throughput from my SIP ATA to my destination if I play some sound at the Asterisk PBX Dialplan before attempting to Dial to connect to destination. Is this a bug? Or does anyone have a clue as to what may be wrong with my setup configuration? |
06:14.52 | Ergose | time to sleep. later. been cool. |
06:14.57 | *** part/#asterisk Ergose (n=Ergose@cpe-065-190-118-012.triad.res.rr.com) |
06:15.10 | phix | corn dogs? |
06:15.14 | harlequin516 | If I don't play a sound, then it keeps attempting to native bridge and never connects the voice. |
06:15.43 | phix | excuse my ignorances but WTF is a corn dog? :) |
06:16.28 | harlequin516 | Is it possible that Asterisk does not setup the voice path unless it needs to (in order to play that sound in the dialplan)? |
06:17.20 | harlequin516 | Could this have something to do with NAT? |
06:21.59 | brettnem | what? bell doesn't charge CLECs for DIDs! |
06:23.44 | Strom_C | http://plif.andkon.com/archive/wc034.gif |
06:36.37 | harlequin516 | brettnem : What does that mean? They don't charge? |
06:37.19 | brettnem | bell doesn't charge for DIDs. they don't even assign DIDs. they have absolutely nothing to do with DIDs. In fact, as a CLEC you tell bell what your DIDs are and you request that they route them |
06:37.38 | brettnem | now, I'm talking about a facilities based CLEC. not a reseller |
06:37.51 | Strom_C | assuming of course that you've received the DID block from CNAC |
06:38.13 | harlequin516 | Hmm.. So does that mean that DID numbers are aquirable from some entity? |
06:38.14 | troy- | if you are facilities-based thats different :) |
06:38.16 | Strom_C | or whatever the canadian numbering plan administrator's abbreciation is this week |
06:38.18 | brettnem | oh is this a canadian discussion? |
06:38.28 | Strom_C | s/cia/via/ |
06:38.29 | troy- | brettnem, yeah |
06:38.45 | brettnem | ah... it's similar up there I believe.. |
06:38.51 | Strom_C | brettnem: ok, so united states, s/CNAC/NANPA/ |
06:38.54 | Strom_C | and same deal |
06:38.55 | troy- | yeah pretty much |
06:38.57 | brettnem | actually |
06:38.57 | brettnem | I think you get your own DIDs for non-fac also |
06:39.12 | troy- | maybe? |
06:39.32 | brettnem | I'm pretty sure.. it's a real similar process.. I've done a lot of fac-based CLEC interconnection in the US |
06:40.36 | brettnem | I've decomissioned a non-fac based CLEC before.. so I've been through a lot of that paperwork. and i know for a fact that the CLEC owns the code and blocks. Bell doesn't have an ordering process for blocks and codes, just TQs and ASRs |
06:41.23 | troy- | could be, i'm busy trying to figure out why asterisk is producing this error |
06:41.51 | *** join/#asterisk samarora (i=minesh@203.88.149.165) |
06:42.08 | samarora | hi there |
06:42.35 | samarora | i need some help regarding connecting two asterisk servers.. |
06:42.38 | troy- | brettnem, bed time for me 2:45 AM :P |
06:42.46 | brettnem | yeah.. ugh.. time |
06:43.02 | troy- | must be up for work in 5 hours :( |
06:43.14 | brettnem | my kids are going to get me up then |
06:43.15 | brettnem | :) |
06:43.49 | troy- | kids are the last thing i want, my dog is lazy and barely likes walking |
06:44.33 | troy- | he's usually ready to turn around before we get out the door |
06:45.40 | brettnem | hah |
06:45.42 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
06:47.01 | troy- | brettnem, i finally understand why they call it man's best friend, if only he drank beer |
06:47.13 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
06:47.18 | brettnem | I hear they drink just about anything |
06:47.24 | *** join/#asterisk AllanLima (n=mediain@unaffiliated/allanlima) |
06:47.35 | troy- | maybe i'll launch a pilot project tomorrow |
06:51.17 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
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06:54.16 | *** part/#asterisk samarora (i=minesh@203.88.149.165) |
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07:16.09 | *** join/#asterisk oej (n=olle@62.97.243.70) |
07:34.06 | *** join/#asterisk basty (n=basty@212.218.65.247) |
07:34.07 | basty | Hi |
07:34.36 | basty | We have several Snom Hardphones on Asterisk 1.2.19, sometimes during a call (external/internal) it breaks up with following Warning in the Asterisk CLI: WARNING[18252] chan_sip.c: Maximum retries exceeded on transmission 3d2cca1cf1b3-c6cbjoxi8rvb@snom320-00041324A7FD for seqno 2 (Critical Response) - anyone knows why ? |
07:35.00 | oej | We're not getting expected response on a SIP message - like it says |
07:35.40 | oej | In an odd way though, but that is what it means. |
07:35.43 | creativx | packetloss? |
07:36.11 | JT | usually it's packet loss |
07:36.17 | basty | yeah - but why ? I mean I used to upgrade Asterisk 1.2.13 to 1.2.19 - allready googled. It seems that there is a change in chan_sip.c from res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2); to initreq 1 |
07:36.21 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
07:36.32 | basty | on Asterisk 1.2.13 it worked just fine... |
07:36.36 | JT | err |
07:36.36 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
07:36.47 | JT | usually it means your network is stuffed, not asterisk :) |
07:37.27 | *** join/#asterisk af_ (n=getsmart@81-174-8-1.dynamic.ngi.it) |
07:37.42 | *** join/#asterisk yassaccan (n=yassacca@admin239.hgo.se) |
07:37.55 | basty | thats weirdo ;) Because it worked before for like 20 weeks without any problems...just after the update to 1.2.19 the "problems" happend |
07:38.02 | basty | happens indeed :) |
07:38.25 | *** join/#asterisk matsk (n=mk@194.68.102.173) |
07:39.26 | jeremy_g | I am so pissed off |
07:39.40 | jeremy_g | why cant i perform make on one distro |
07:39.55 | jeremy_g | and make install on another to just put the binaries in there respective places |
07:39.59 | *** join/#asterisk dharrigan (n=dharriga@dsl-217-155-228-129.zen.co.uk) |
07:40.00 | basty | so anyway..in your eyes, the error might be a packetloss from the snom hardphone to the server ? |
07:40.06 | jeremy_g | why does make install require libs n crap |
07:42.00 | JT | jeremy_g: what are you talking about? |
07:42.11 | oej | basty: Don't look in the source, look in the SIP dialogs. Some device is not responding properly. You have to check what type of message that isn't handled by turning on SIP debug and seeing what's going on. |
07:42.57 | creativx | basty: thats why you dont upgrade if shit works :) |
07:43.36 | jeremy_g | JT:i want to compile asterisk on one machine loaded with compiler and all the libs and copy the binaries to another linux system that doesnt have a compiler and that many lib files. I ldd the binary and install the libraries it requires. |
07:43.38 | basty | creativx: Well I had several cores in 1.2.13 ;-) |
07:43.45 | jeremy_g | i am making a live CD for asterisk |
07:44.02 | *** join/#asterisk kova (n=kova@tech.quentris.com) |
07:44.12 | JT | jeremy_g: do you have a cross compiler? |
07:44.21 | basty | oej: Some devices? It seems that ALL my Snom Phones (about 115) have the Problems :-( |
07:44.29 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
07:44.35 | JT | basty: are all the settings the same? |
07:44.51 | basty | JT: Settings are the same...firmware could be different |
07:44.57 | jeremy_g | JT:both are linux, though different distros |
07:45.06 | jeremy_g | JT:same architectures |
07:45.47 | oej | What's the sip message you're sending? A 200 OK on an invite requires an ACK |
07:45.49 | JT | can't you just copy the binares? |
07:45.54 | oej | so the 2 is correct |
07:46.15 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
07:46.34 | basty | oej: I dont know what the phone is sending..sounds like I have to turn on the SIP debug on Asterisk? |
07:46.54 | oej | basty: That's the only way to figure out what's going on |
07:47.18 | jeremy_g | JT: i copied the asterisk directory in which i successfully executed make, to the other linux system and there did make install which gave this error: http://www.pastebin.ca/595003 |
07:47.54 | jeremy_g | why does make install check for the libraries required |
07:48.05 | jeremy_g | i thought make install was suppose to compile everything |
07:48.31 | basty | oej: just a general question....after "sip debug" how to turn the sip debug off ? ;-) |
07:48.32 | creativx | damn swissvoice crap |
07:48.39 | creativx | note to self: never buy swissvoice again |
07:49.03 | basty | oej: no debug channel sip ? |
07:49.18 | creativx | sip no debug |
07:49.20 | creativx | :) |
07:49.21 | oej | basty: Everything starts with "sip" |
07:49.25 | basty | i am sorry ;) |
07:49.27 | creativx | not very intiutive |
07:49.32 | basty | thanks creativx |
07:49.36 | creativx | should be sip debug [on/off] :) |
07:49.37 | creativx | hehe |
07:49.42 | oej | in 1.6 it's going to be "sip set debug off" |
07:49.45 | basty | yeah ;) |
07:49.48 | jeremy_g | :D |
07:49.55 | creativx | hehe |
07:50.00 | creativx | looking forward to 1.6 |
07:50.06 | creativx | or perhaps not.. 1.2 is working well |
07:50.06 | creativx | :) |
07:50.11 | jeremy_g | in 1.8 its gonna be sip configure debug set off |
07:50.12 | jeremy_g | :D |
07:50.33 | JT | sip no debug.. that style is pretty much cisco style |
07:51.25 | creativx | who said cisco was intiutive :) hehe |
07:51.26 | basty | is there any way to log the sip-debug into a file instead of the CLI ? |
07:51.38 | creativx | /var/log/asterisk/messages basty |
07:51.47 | basty | yeah..but it is still in my cli ;) |
07:51.56 | jeremy_g | JT: any idea which library do i need? http://www.pastebin.ca/595003 |
07:52.00 | creativx | /var/log/asterisk/messages is a file |
07:52.23 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net) |
07:52.29 | basty | creativx: lol - I know that - but I dont want to see the odd sip-debug in the asterisk-cli... |
07:52.34 | oej | basty: the SIP messages will always be to CLI, but you can copy directly to a file by doing asterisk -rvvvddddn | tee /tmp/mysipdebug.txt |
07:52.34 | jeremy_g | basty:i always do it like that, you can forward all verbose output to a file |
07:52.45 | JT | jeremy_g: yes, kernel headers. |
07:52.45 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
07:52.53 | basty | ah cool - thanks oej :) |
07:53.10 | creativx | ahh basty :P |
07:53.36 | jeremy_g | basty:or vi /etc/asterisk/logger.conf and add this line log-sip => verbose |
07:53.47 | basty | thanks! |
07:54.37 | jeremy_g | basty:i then usually tail -f /var/log/asterisk/log-sip |grep 'anything-special-um-looking-for' |
07:55.32 | creativx | then look forward to delving into a buttload o sip messages |
07:55.32 | creativx | =) |
07:55.37 | basty | hehe |
07:57.13 | JT | tzafrir: about? |
07:59.31 | basty | okay |
07:59.55 | basty | I dont think logging all sip messages on this machine is a good idea |
08:00.15 | basty | -rw-r--r-- 1 root root 102M 2007-06-29 09:59 log-sip |
08:00.32 | basty | in..hrm..10mins ? ;) |
08:00.39 | JT | wtf |
08:00.45 | JT | you know why that is |
08:00.49 | JT | because it's failing |
08:00.53 | JT | and retransmitting |
08:01.23 | basty | i have a lot of "Destroying call '763652823562007144222@10.46.0.5'" stuff in there |
08:01.50 | creativx | how is your network? |
08:02.25 | JT | basty: time to start packet sniffing |
08:02.31 | JT | something strange is going on |
08:02.41 | basty | well...I have several Cisco Catalyst 3548XL Switches |
08:02.54 | basty | creativx: or what do you mean by "how" ? ;-) |
08:03.17 | creativx | "how" as in any nasty equipment, low quality? |
08:03.29 | JT | cisco can be nasty |
08:03.40 | JT | especially if you ever use a piece of rubbish known as "PIX" |
08:03.43 | basty | creativx: nah...no low quality..only the best ;-) |
08:03.58 | creativx | hehe |
08:04.05 | creativx | glad i didnt buy a PIX.. netscreen <3 |
08:05.33 | basty | hrm |
08:06.00 | basty | Jun 29 09:58:29 VERBOSE[28550] logger.c: Sending to 10.46.3.100 : 2051 (NAT) |
08:06.06 | basty | why does it use NAT ? |
08:06.12 | basty | I dont want to use NAT ;) |
08:06.32 | creativx | asterisk on a different subnet than 10.46.3 ? |
08:07.01 | basty | well...the network is 10.46.0.0 / 255.255.248.0 |
08:07.06 | basty | so...the same network :) |
08:07.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
08:07.43 | JT | of course localnet= is setup correctly in sip.conf |
08:07.47 | basty | yep |
08:07.56 | JT | maybe you should set nat=no for the phones |
08:08.17 | JT | i thought this was the problem |
08:08.30 | JT | just didn't think it was that important since you didn't pastebin anything |
08:09.00 | basty | hrm |
08:09.30 | basty | so I should insert "nat = no" in the general settings ? |
08:10.02 | JT | you can, but if you have clients/servers over nat, you'll want =yes for them |
08:10.12 | basty | nah..its all in the same network |
08:10.30 | JT | so no voip on the Internet |
08:11.08 | basty | nope |
08:11.19 | basty | just 3 E1 Interfaces |
08:11.35 | JT | ok |
08:12.05 | basty | so the "nat = no" should fix my problem with the disconnecting calls during the conversation ? |
08:12.37 | JT | hopefully, just give it a go already |
08:12.42 | JT | instead of wasting time :P |
08:12.52 | berktr | basty, do you have canreinvite=yes ? |
08:13.01 | JT | and no spaces on parameters in asterisk configuration files, least not in that section |
08:13.25 | basty | berktr: canreinvite=no |
08:13.39 | berktr | ok, and you still have disconnecting calls? |
08:13.42 | basty | yep |
08:13.59 | berktr | like after how many seconds? |
08:14.08 | basty | ...well I dunno what the "nat=yes" did |
08:14.19 | basty | berktr: differnt...lets say..every hour ? :) |
08:14.20 | berktr | nat yes wouldn't cause such a problem |
08:14.25 | JT | did you set it to no? |
08:14.31 | JT | it can |
08:14.37 | berktr | no, i mean after you initiate your call, the other party answers, and the conversation beging |
08:14.41 | berktr | begins* |
08:14.50 | berktr | can you talk for like 15minutes? |
08:15.21 | basty | berktr: oh okay..lets say...2-3 seconds ? |
08:15.31 | berktr | i see |
08:15.33 | basty | JT: nat=no - yep |
08:15.44 | berktr | basty, is this a fresh asterisk installation? |
08:16.12 | basty | not really...i used 1.2.13 for quiet a while....after some core dumps I had to upgrade to 1.2.19 :-\ |
08:16.42 | basty | i didnt change anything..just upgraded...and since that time..i have these weirdo problems. |
08:16.44 | berktr | this most probably won't make sense but i had exactly the same problem with 1.2 after a fresh installation |
08:16.58 | berktr | and i was about to go crazy and upgraded to 1.4 |
08:17.06 | berktr | now i'm having no problems at all |
08:17.23 | berktr | too bad i couldn't figure out what caused my problem but |
08:17.39 | berktr | switching to 1.4 solved it, and i had no time to research more |
08:17.45 | basty | :-( |
08:17.57 | berktr | why do u still insist on 1.2? |
08:18.06 | berktr | it can't even change rtp packet sizes? |
08:18.08 | basty | because 1.2 is actually stable. |
08:18.26 | basty | i dont trust 1.4 for such a big installation |
08:19.17 | *** join/#asterisk berktr (n=cn@teknopet.com) |
08:19.24 | berktr | sorry somehow freenode dropped me |
08:19.54 | basty | np :) well..like I said..i dont really trust 1.4... |
08:20.03 | basty | 1.2 is actually working fine...as in stable... |
08:20.15 | berktr | i used to work with 1.2, but 1.4 is really much better |
08:20.24 | berktr | sip jitter buffer works perfectly |
08:20.37 | berktr | do you use h323 there? |
08:20.45 | basty | nah...just SIP and IAX |
08:20.55 | berktr | okay then, you can switch to 1.4 |
08:21.01 | berktr | it is really easy and takes less time |
08:21.17 | basty | berktr: how many phones are connected on your installation ? |
08:21.24 | basty | and how many E1/T1 Lines? |
08:21.24 | berktr | 130 |
08:21.32 | berktr | here, it's different |
08:21.36 | berktr | no e1/t1 |
08:21.39 | basty | oh |
08:21.41 | berktr | 130 internal phones |
08:21.52 | berktr | and we are directly connected to local telecom operator |
08:21.55 | berktr | over internet |
08:22.04 | berktr | we have did numbers |
08:22.04 | basty | oh okay |
08:22.40 | berktr | asterisk is handling conversations real good |
08:22.48 | berktr | internal phones use ulaw in the network |
08:22.55 | berktr | pstn connections are g729 |
08:23.06 | basty | do you use FAX with Asterisk ? |
08:23.13 | berktr | don't even ask.... |
08:23.23 | berktr | i couldn't get it to work :( |
08:23.34 | berktr | 1.4 supports t38 passthrough |
08:23.46 | berktr | but our telecom provider doesn't support it |
08:23.50 | basty | oh okay |
08:24.53 | basty | well..actually I work with mISDN or Zaptel...for E1 Interfaces...but as a matter of fact I had to realize the fax with Audiocodes Gateways |
08:24.53 | berktr | we have some welltech gateways here |
08:24.53 | berktr | they all support faxing |
08:25.01 | basty | in T.38 ? |
08:25.01 | *** join/#asterisk ramindia (n=ramindia@202.63.96.9) |
08:25.01 | berktr | yes |
08:25.01 | berktr | with ulaw |
08:25.10 | *** part/#asterisk ramindia (n=ramindia@202.63.96.9) |
08:25.56 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
08:25.57 | basty | berktr: where are you from ? |
08:26.04 | berktr | turkey |
08:26.06 | basty | ah okay |
08:26.14 | berktr | why did u ask |
08:26.50 | berktr | http://www.welltech.com/product_e_01.htm |
08:26.50 | basty | I thought we could try to talk in German ;) Because my english isnt that good |
08:27.06 | berktr | ahh i see, only english - turkish - french here |
08:27.15 | basty | what a shame :) |
08:27.33 | berktr | lol :) |
08:27.45 | Zeeek | ich habbe kein camera |
08:27.53 | Zeeek | noch en, bitte |
08:28.26 | phez72 | I'm not english, turkish or french |
08:28.38 | phez72 | :) |
08:28.42 | Zeeek | ein Traum ist unser Leben auf Erden hier |
08:28.54 | berktr | basty, if you want to go to a vacation, with no problems behind you, switch to freebsd, use ports, install 1.4, configure it, and forget about it |
08:29.08 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
08:29.20 | *** join/#asterisk tsurko (n=tsurko@150-190.go.evo.bg) |
08:29.29 | creativx | heute is mein tag |
08:29.41 | basty | zeek: hehe |
08:29.59 | basty | berktr: heheh..I prefer FreeBSD also |
08:30.04 | Zeeek | nadle tsipful des Kafer bumser |
08:30.07 | basty | creativx: "ist" :) |
08:30.14 | berktr | yeah, it's cool, installs in 2 minutes |
08:30.37 | basty | but still..there is no misdn/zaptel support for freebsd |
08:30.40 | berktr | give me a brand new computer with no os, it will take only 1 hour for me to set asterisk up on it |
08:30.56 | creativx | verdammnt basty, i forgot :) |
08:30.56 | Zeeek | berktr that's about right |
08:31.08 | basty | creativx: laugh |
08:31.15 | berktr | i don't know about zaptel support though |
08:31.20 | berktr | i thought it was solved in 1.4 |
08:31.43 | basty | brb....cigarette |
08:31.47 | berktr | k |
08:32.16 | Zeeek | smoking is only permitted during the compilation phase |
08:32.32 | berktr | :) |
08:32.53 | Zeeek | and then only when compiling assembly language... from a cassette tape |
08:33.35 | creativx | ehh.. what the hell am I thinking wrong here.. how do I invoke musiconhold on an extension? exten => s,n,musiconhold(class) right? |
08:33.56 | berktr | here goes a moh question |
08:34.01 | Zeeek | you don't invoke musicon hold |
08:34.03 | berktr | let's say i changed the music on hold music |
08:34.12 | berktr | how can i reload the moh module? |
08:34.15 | berktr | to switch to new music |
08:34.25 | Zeeek | restart |
08:34.29 | berktr | grr |
08:34.32 | creativx | but I want to Zeeek :) |
08:34.32 | creativx | hehe |
08:34.41 | Zeeek | than look up playback |
08:34.47 | creativx | yeah thats what im doing now |
08:34.50 | berktr | without restarting, is it possible |
08:35.04 | *** join/#asterisk l2trace9999 (n=l2trace@fl-67-76-209-28.sta.embarqhsd.net) |
08:35.17 | Zeeek | berktr depends. If you have a bunch of mp3players already running as processes, no |
08:35.31 | basty | back |
08:35.56 | Zeeek | you're right not to inhale |
08:36.20 | creativx | how old is this wiki page about cmd musiconhold() i wonder |
08:36.27 | basty | okay...for right now..i dont have any calls broke up..after setting "nat =yes" so it seems to work...for right now.. ;) |
08:36.35 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
08:36.42 | l2trace9999 | any one know of an option to Dial like g but for the caller |
08:37.30 | Strom_M | ? |
08:37.36 | creativx | Zeeek: reason i wanted to invoke it was im using a ghetto parking extension.. and instead of playback(same file) each time.. i thought moh could handle it via a moh class |
08:38.35 | Strom_M | l2trace9999: i don't understand your question |
08:39.17 | l2trace9999 | the asteisk cmd Dial has an option to it |
08:39.18 | l2trace9999 | g |
08:39.33 | basty | okay..next question..anyone familar with call-queues ? I have setup a hotline queue...with 10 agents...during a 24/7 support only 2 agents are logged in...actually the queue works perfect. But if the 2 Agents are talking and the queue is actually busy...it rings for like 2 seconds...and then it is busy. In the Queue i had to setup "r". How can I fix that ? I dont want that the caller is getting a "ring" before the "busy". |
08:39.46 | l2trace9999 | it allows the dial plan continue when the callee exits |
08:39.57 | l2trace9999 | hangup or disconnected |
08:40.10 | l2trace9999 | but it doesn't continue when the caller exits |
08:40.19 | l2trace9999 | hangup or disconnected |
08:40.50 | Strom_M | l2trace9999: thats because when the calling party hangs up, the call is torn down |
08:41.02 | basty | oh and if I use "exten 1,1,Queue(technik) without the "r" - the caller hears nothing..not even the ringing.. :-( |
08:41.10 | Strom_M | if you want to do post-teardown processing, use the h extension |
08:41.38 | l2trace9999 | for outbound dialling ? |
08:41.47 | Strom_M | for any call |
08:41.55 | creativx | basty: without r they should get moh if configured |
08:41.55 | l2trace9999 | ok |
08:42.21 | basty | creativx: exacly...but I want to have a "Ringing" instead of "moh" |
08:42.48 | basty | so actually i HAVE to use the "r" - but with using the "r" it rings..even when the queue is busy... |
08:43.09 | creativx | yep i know |
08:43.14 | kiscokid | I installed Asterisk Gui on 1.4 but the login web page keeps reloading over and over even after I login |
08:43.15 | l2trace9999 | thanks |
08:43.21 | creativx | i had some run ins with the queues myself |
08:43.23 | Strom_M | l2trace9999: also, some terminology for you |
08:43.29 | Strom_M | "calling party" |
08:43.33 | Strom_M | "called party" |
08:43.37 | Strom_M | not "caller" and "callee" |
08:43.49 | creativx | i ended up with no r, moh silence and really short intervals for queue positioning messages etc |
08:44.02 | dharrigan | Sorry folks for asking here, I'm a right newbie. I'm hoping to setup asterisk shortly but what I don't understand about the last bit - if I have, say 20 internal analogue phones, how do I connect these to a fxs/fxo interface? These interfaces come with like 4 or 8 ports, do I need multiple cards? |
08:44.16 | l2trace9999 | i know but i was using terminology from the voip-info documentation |
08:44.20 | basty | creativx: So there is actually no fix for that? Because I disabled the queue position messages... |
08:44.20 | dharrigan | and will asterisk communicate with all the cards? |
08:44.33 | creativx | basty: i didnt have time to investigate |
08:44.36 | Strom_M | dharrigan: you can get the tdm2400, which is a 24 port card |
08:44.39 | Zeeek | dharrigan yes you need multiple cards |
08:44.44 | basty | ah okay...dang :-( |
08:44.48 | Zeeek | or the 240 |
08:44.50 | Zeeek | or the 2400 |
08:44.51 | basty | or like you said..."verdammt" ;-) |
08:44.53 | Strom_M | or you can do a T1/E1 span and a channel bank |
08:45.07 | Strom_M | l2trace9999: voip-info is not a reliable source for accurate terminology :) |
08:45.26 | Zeeek | dharrigan I think at the level of tens of phones it becomes cheaper to switch to SIP phones |
08:45.36 | Strom_M | it depends |
08:45.50 | basty | creativx: familar with sipsak ? :) |
08:46.00 | Strom_M | if there are existing phones, the channel bank / tdm card option may be cheaper per-port |
08:46.02 | dharrigan | SIP phones - these connect via the internal network yes, then that gets routed to the asterisk server? |
08:46.12 | Zeeek | yes |
08:46.51 | dharrigan | Cool, that's all for answer and clearing up some confusion |
08:46.59 | dharrigan | s/that's/thanks/ |
08:47.21 | Zeeek | do you need to use existing analog phones? |
08:47.27 | Strom_M | dharrigan: of course, your data network has to be up to snuff for SIP phones to work properly also |
08:47.37 | Zeeek | if so, it's more complicated to determine which is better |
08:47.59 | creativx | Zeeek: i got it working now.. there is a way of invoking moh |
08:47.59 | creativx | ;) |
08:48.15 | Zeeek | tell us so we won't die stupid |
08:48.35 | Zeeek | s/stupid/ignorant/ |
08:49.04 | dharrigan | Zeeek: Well, I think the office has about 5 phones, so perhaps easier just to buy sip phones? |
08:49.19 | Zeeek | In my opinion it would be |
08:49.25 | creativx | first its a good idea to load res_musiconhold.so |
08:49.31 | l2trace9999 | thanks now i feel like an idiot because I spent the last 2 days on that |
08:49.33 | Zeeek | but it depends on a lot of things, like the users habits etc |
08:49.38 | Strom_M | dharrigan: yeah, but whatever you do, don't buy grandstream |
08:50.01 | Zeeek | the linksys phones are decent and the polycoms are good |
08:50.11 | creativx | so with res_musiconhold.so, you get application setmusiconhold(class) and musiconhold(class) |
08:50.14 | dharrigan | However, they may not wish to expend money on replacing phones they already have, so I may have to go with analogue |
08:50.17 | Zeeek | linksys used to be called Sipura |
08:50.33 | Zeeek | you can mix and match. Keep 4 analogue |
08:50.36 | Strom_M | Zeeek: no, linksys /purchased/ sipura |
08:50.46 | Zeeek | the result is the same |
08:51.01 | Zeeek | what was SIpura is now called linksys (or cisco even) |
08:51.16 | Zeeek | they're decent phones (941-2) |
08:51.22 | Strom_M | yes, but you made it sound like the one company simply became the other :) |
08:51.33 | Zeeek | what is your native language? |
08:51.45 | Strom_M | mine? |
08:51.48 | Zeeek | ya |
08:51.53 | Strom_M | English |
08:51.56 | creativx | Zeeek: are you prepared to die now then? |
08:52.09 | Zeeek | no because I'm still ignorant! :) |
08:52.32 | creativx | but you could atleast die happy |
08:52.41 | creativx | knowing you can torture callers with moh at your discretion |
08:52.41 | Zeeek | but I'm not happy at the moment |
08:53.00 | Zeeek | creativx I have a much better torture for them: voice recognition! |
08:53.09 | creativx | ouch |
08:53.18 | Zeeek | that shit is REALLY irritating |
08:53.24 | creativx | yes |
08:53.27 | creativx | i was about to say |
08:53.30 | creativx | has that tech matured enough |
08:53.33 | Zeeek | as anyone who's ever flown and reserved on the phone knows |
08:53.34 | creativx | but you already answered :) |
08:53.45 | creativx | what? i order my flights online ;) hehe |
08:53.46 | Zeeek | no it works well |
08:56.35 | kiscokid | anyone using Asterisk Gui with 1.4? |
08:56.53 | vlt | Hello. Is there a dialplan pattern for the following regex? /^555[0-9][0-9]*$/ |
08:57.22 | Strom_M | _555XX. |
08:57.28 | vlt | _555X. doesn't work because it doens'n match 5550 |
08:57.40 | Strom_M | _555X! |
08:57.45 | Strom_M | that'll match 5550 |
08:58.15 | vlt | So I need two different extensions, right? |
08:58.28 | Strom_M | ....? |
08:58.36 | Strom_M | ! maches 0 or more digits |
08:58.49 | vlt | Aaah ... nice. |
08:59.01 | creativx | very nice |
08:59.03 | creativx | make sexy time |
08:59.25 | vlt | So _555X! matches 5550 and 55599? |
08:59.25 | Strom_M | was that a suggestion or a pickup line? |
08:59.28 | Strom_M | yes |
08:59.44 | Strom_M | and 555999999929342394293492593452353623456 |
08:59.50 | vlt | ;-) |
09:02.03 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
09:02.36 | tzafrir_laptop | kiscokid, ask something more specific |
09:02.53 | tzafrir_laptop | or ask on #asterisk-gui |
09:03.38 | creativx | Strom_M: hehe.. indecent proposals.. its friday innit |
09:03.54 | kiscokid | I tried installing Asterisk Gui on 1.4 and it seemed to install correctly but the login page keeps reloading even after logging in |
09:04.45 | creativx | cookie? |
09:05.02 | Zeeek | no thanks, I'm dieting |
09:05.08 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
09:05.09 | *** join/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net) |
09:06.38 | kiscokid | I'll try clearing cookies |
09:07.16 | creativx | hehe |
09:07.22 | creativx | interesting.. 99,9% cpu usage |
09:07.44 | Zeeek | stop surfing pr0n sites |
09:08.21 | Strom_M | http://www.jerkcity.com/jerkcity1578.html |
09:10.33 | kiscokid | creative: that fixed it ! |
09:12.07 | creativx | hehe kiscokid |
09:12.30 | creativx | glad to har |
09:12.31 | creativx | hear |
09:13.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
09:13.59 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
09:16.37 | creativx | exiting: 32253 root 25 0 21180 9916 4864 R 99.9 1.0 2757:06 asterisk |
09:16.50 | creativx | what have I done to my poor asterisk since its chewing 99% cpu |
09:26.18 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
09:27.42 | corruptor | My asterisk 1.4.5 after about 2 days of uptime works very slowly and consumes more and more memory. I relate this to Mixmonitor usage. Which info I have to post mantis? or how can i debug thisterisk after about 2 days of uptime works very slowly and consumes more and more memory. I relate this to Mixmonitor usage. how can i debug this? I need some info to post mantis but don't what can I post here. It's already compiled with dont_optimize and debug_th |
09:27.42 | corruptor | reads |
09:29.25 | Strom_M | paste mishap |
09:29.26 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
09:30.16 | corruptor | oops sorry for double post |
09:39.30 | Zeeek | and now... silence... |
09:39.44 | Zeeek | slowly I turend, step by step... inch by inch... |
09:44.28 | *** part/#asterisk SwK (n=SwK@dhcp64-134-34-241.bwic.chi.wayport.net) |
09:47.40 | *** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) |
09:47.59 | creativx | .mute=true |
09:48.59 | Zeeek | Toggle(${mute}) |
09:56.47 | creativx | set(mute=yes) would be proper cli syntax |
09:56.47 | creativx | ;) |
09:57.02 | creativx | queue pause sip/587 tnx.. lunchtime |
09:57.03 | Zeeek | not with my toggle function |
09:57.07 | Zeeek | http://www.lulu.tv/?p=11368 |
09:57.49 | creativx | haha |
09:57.50 | creativx | ghetto |
09:58.09 | creativx | glad I have to care about is the cat6 cable |
10:00.08 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
10:01.23 | creativx | damn this chicken burrito was disappointing |
10:01.26 | *** join/#asterisk Andri[DK] (n=andri@hydrogen.calidris.com) |
10:02.52 | Andri[DK] | I'm currently in the process of upgrading my old Asterisk installation (1.0 branch) to a more modern installation. Are there any distribution recommendations. I'd prefer to keep everything within the packaging systems but I'd also like to have some sort of GUI for extensions and such. |
10:03.18 | creativx | only option for gui is 1.4 (natively) |
10:03.21 | creativx | afaik |
10:03.33 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
10:03.55 | creativx | and going from 1.0 to 1.4 could be interesting |
10:04.09 | creativx | i would start reading about all deprecated functions and apps, and grepping like a madman |
10:04.56 | Andri[DK] | It's going to be a slow migration. I'll have a second server with a new SIP uplink and when that is working properly, I can start moving the phones |
10:05.10 | Andri[DK] | Can you recommend a Linux distro with 1.4 ? |
10:05.55 | *** join/#asterisk vn (n=nostalge@bas5-quebec14-1177599731.dsl.bell.ca) |
10:06.00 | Zeeek | centos 5 |
10:06.44 | *** join/#asterisk Ebola (n=Ebola@host86-136-134-202.range86-136.btcentralplus.com) |
10:06.57 | Andri[DK] | is that packaged with Asterisk 1.4? |
10:07.06 | Zeeek | no |
10:07.28 | Andri[DK] | I left my company for school and I'm just here for the summer. So I'd like my inexperienced replacement to be able to update it :) |
10:07.51 | creativx | hehe poor sould |
10:07.52 | creativx | -d |
10:08.06 | Andri[DK] | so, there is no distro packaged with 1.4 yet ? |
10:08.20 | Andri[DK] | not even 3rd party repositories for debian or ubuntu ? |
10:08.33 | creativx | i doubt you are looking for trixbox |
10:09.05 | Zeeek | why package? Just get an os and do it |
10:09.19 | Zeeek | someone just said this morning it only takes about an hour |
10:09.29 | mvanbaak | Andri[DK]: try http://www.asterisknow.org |
10:10.03 | creativx | make |
10:11.25 | mvanbaak | anyone from digium here? |
10:11.30 | mvanbaak | I cannot download asterisknow ;) |
10:11.37 | mvanbaak | Iceweasel can't find the server at asterisknow.staging.digium.com. |
10:13.29 | creativx | asteriskLATER |
10:13.31 | creativx | plz try again |
10:13.34 | creativx | :) |
10:13.54 | mvanbaak | nope |
10:14.07 | Strom_M | yeah....like when it's not 5 AM |
10:14.16 | creativx | its not 5 am everywhere |
10:14.21 | Strom_M | it's 5 AM in alabama |
10:14.24 | mvanbaak | Fri Jun 29 12:14:23 CEST 2007 |
10:14.24 | Strom_M | and digium is in alabama |
10:14.25 | Strom_M | therefore |
10:14.27 | Strom_M | BONERS |
10:14.32 | creativx | its noon in norway |
10:14.35 | creativx | n for norwegium |
10:14.37 | Strom_M | good for norway |
10:14.48 | Strom_M | alabama is not in norway |
10:14.51 | mvanbaak | hhmm, the vmware images can be downloaded |
10:15.06 | Strom_M | (a-ha) |
10:15.25 | mvanbaak | wrong link there I guess |
10:16.03 | creativx | Strom_M: but norway is in usa |
10:16.10 | creativx | i think |
10:16.28 | Zeeek | Norwalk is in California |
10:16.43 | mvanbaak | yup |
10:16.45 | mvanbaak | wrong link |
10:16.56 | Strom_M | specifically, southern california |
10:17.04 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
10:17.13 | Strom_M | los angeles county |
10:17.13 | Strom_M | area code 562 |
10:17.16 | mvanbaak | if I copy-paste and replace asterisknow.staging.digium.com with www.asterisknow.org it's working again |
10:17.18 | creativx | Walk the walk |
10:17.27 | Strom_M | cock the cock |
10:17.33 | creativx | shoot the gun |
10:17.37 | Strom_M | bonerpills |
10:17.59 | cpm | sure as a ferret is a ferret |
10:18.30 | Andri[DK] | Zeeek: I'd like the pacacking for my replacement. I can easily install it from source but then it's going to be worse when it comes to patching security problems and such for the guys here that aren't that familiar with Linux or Asterisk |
10:18.59 | creativx | how can they manage asterisk if they cant even compile from source :) |
10:19.09 | creativx | i didnt know the GUI was that far along.. hehe |
10:19.09 | Strom_M | those are some sysadmins |
10:19.09 | Andri[DK] | btw, if anyone is looking for a Linux related job, Iceland is the place to be... Too many jobs, not enough people |
10:19.25 | Strom_M | sunlight 24 hours a day during the summer |
10:19.31 | Andri[DK] | yeah |
10:19.32 | Strom_M | scenic reykjavik |
10:19.39 | creativx | glitnir |
10:19.43 | Strom_M | blblblbblblbbl |
10:19.59 | Andri[DK] | you guys know Iceland? :D |
10:20.05 | Strom_M | skak mat 1972 |
10:20.13 | creativx | i know its expencive as f there |
10:20.19 | creativx | and i know of arctic trucks |
10:20.43 | Andri[DK] | cool, I'm here now for the summer, but it's just because the salary is pretty good, even though it's expensive to live here |
10:20.55 | creativx | do they understand danish? |
10:20.59 | Strom_M | I can spell ? |
10:21.04 | creativx | i can has cheezburger? |
10:21.10 | Andri[DK] | some do, but I'm Icelandic... I'm just studying in Denmark |
10:21.35 | Zeeek | securityR US |
10:21.35 | Strom_M | and also ? |
10:21.36 | creativx | aha |
10:21.37 | Strom_M | no, a-ha is norway |
10:21.46 | creativx | misleading suffix there andri |
10:21.48 | Andri[DK] | take on me *sings* |
10:21.52 | creativx | a-ha sure is |
10:21.56 | Strom_M | the sun always shines on tv |
10:22.03 | creativx | the sun shines in bergen too |
10:22.11 | Strom_M | but especially los angeles |
10:22.19 | Strom_M | and what about cheesecake |
10:22.19 | Andri[DK] | creativx: yeah, sorry... I live in DK, didn't wanna register another nickname :P |
10:24.33 | creativx | so how do you like denmark |
10:24.41 | creativx | are they treating an icelandic well |
10:24.44 | Andri[IS] | very well, cheap beer, pretty ladies ;) |
10:25.06 | Andri[IS] | or was it pretty beer and cheap ladies... it's good anyway |
10:25.08 | creativx | yeah they got the ladies we didnt have room for |
10:25.21 | creativx | here the beer is expencive and the ladies pretty cheap |
10:25.22 | creativx | ;) |
10:25.46 | Andri[IS] | Iceland is like the run-away-child of Denmark. We can still come home to mom and dad, even if we wanna be independent |
10:26.18 | Andri[IS] | it's f.e. cheaper for me to study there, than home in Iceland |
10:27.27 | creativx | hehe |
10:27.29 | creativx | interesting |
10:27.51 | creativx | i know a lot of the fish-business like to have meetings and trips to iceland |
10:27.59 | creativx | they burn money like its 1995 |
10:28.38 | Andri[IS] | yeah, those assholes are like the icelandic royalty, inherit daddy's fishing quotas and spend money like there is no tomorrow |
10:29.10 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
10:29.26 | creativx | money exists for one purpose |
10:29.29 | creativx | to be kept in circulation |
10:29.33 | creativx | e.g. use em while you got em! |
10:30.13 | Andri[IS] | yeah, somebody should tell those Norwegians... All that oil and still just keep the money in the bank |
10:31.01 | creativx | its the norwegian pension fund :) |
10:31.07 | creativx | not even called oil money any more |
10:31.30 | Strom_M | IKEA |
10:32.19 | creativx | jarlsberg |
10:33.16 | Strom_M | johannesburg? |
10:33.25 | Andri[IS] | tuborg? |
10:33.42 | Strom_M | katzegeschlecht? |
10:34.45 | Strom_M | killing the conversation since 2004 |
10:34.46 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
10:34.49 | creativx | hehe |
10:34.52 | creativx | still going strong |
10:35.00 | creativx | bumsenkönig? |
10:37.01 | Strom_M | hollywood, california (not a real city) |
10:38.43 | creativx | mm |
10:38.47 | creativx | i can feel the friday vibe already |
10:41.13 | creativx | perhaps i should go get an icecream in ze fridge |
10:42.40 | *** join/#asterisk javar (n=javar@69.79.134.24) |
10:42.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
10:43.57 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
10:44.09 | linagee | yikes. better have your caller ID set right. LOL. $10,000 fine for doing it wrong. |
10:44.25 | creativx | in what messed up country? |
10:44.30 | linagee | i wonder what happens if you have one of those silly phone cards that let you change it? |
10:44.30 | creativx | one that deals with dollars |
10:45.10 | linagee | creativx: what if it was like, "don't you DARE try to spoof your vendor assigned MAC address" (this is essentially what it is.) |
10:45.31 | linagee | (yes, i am saying a phone number is just as arbitrarily assigned as a MAC address) |
10:48.17 | creativx | yup |
10:48.26 | creativx | i know a guy who benefited from spoofing mac's |
10:48.34 | creativx | he managed to bridge some 4-5 cable modems |
10:48.41 | creativx | and get sweet bandwidth for free |
10:49.19 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
10:49.46 | linagee | creativx: what is "spoofing an outgoing caller id" if for example, you have a call center that all send out your 800 number on the caller ID? |
10:50.49 | linagee | maybe they are somehow magically exempt because they are a business. heh |
10:50.53 | creativx | this is what we would call a legal hell |
10:55.57 | linagee | creativx: hrm. someone asks a legit question on slashdot. :) does the victim of the spoofer get part of the $10,000? (heh. that would be nice.) |
10:57.07 | dharrigan | Hi peeps again! :-) I think I've figured out now about asterisk + analogue telephony. Now, my question is if I want to do voip from my place to another place, I need to go over the net right? I can't do that via an analogue line, so the asterisk server needs some type of network connection - it can route sip/voip calls across the net? |
10:57.42 | dharrigan | so my asterisk installation can act as a gateway to traditional analogue and also do across the net voip calls as well? |
10:58.58 | creativx | dharrigan: correct |
10:59.05 | creativx | you decide in your extensions where to route outgoing calls |
10:59.19 | creativx | be it a sip/iax2 trunk over the interweb, or into a local analog line via zap |
11:00.24 | linagee | wtf? |
11:00.35 | linagee | they intend on branding us all with "National ID cards"? |
11:00.37 | linagee | :( |
11:00.52 | dharrigan | beautiful. so, we have asterisk now routing traditional calls via "zap" and sip calls via the interweb (always liked that term :) What would be the |
11:00.54 | linagee | (USA) |
11:00.57 | Krooks | If I install libpri and I don't need it. does it hurt ? |
11:01.01 | *** join/#asterisk alomelo (n=wyginwys@85.105.204.40) |
11:01.21 | dharrigan | recommended upstream bandwidth requirments for say, 8 phones (and say worse case, 8 all answered at once) whilst doing SIP calls? |
11:01.25 | linagee | maybe they will associate your national ID number with your phone number. LOL |
11:01.53 | linagee | dharrigan: 640 kilobits (and then some) |
11:02.10 | key2 | ? |
11:02.13 | creativx | dharrigan: depends on codec |
11:02.14 | dharrigan | linagee: thanks, is there some type of formula you used? |
11:02.21 | linagee | oh yes. and depends on codec. :) |
11:02.35 | linagee | dharrigan: about 80 kilobits/sec for PCM/ulaw |
11:02.41 | dharrigan | creativx: can you help explain about which codec is best (I know what a codec is) |
11:02.44 | creativx | it also depends if you are trunking or not |
11:02.45 | linagee | afaik. |
11:02.50 | linagee | it might jump to 90 kilobits |
11:02.53 | creativx | dharrigan: theres some good articles on bandwidth calculations |
11:02.58 | creativx | i just dont remember them now hehe |
11:03.01 | dharrigan | creativx: great, can you link me? |
11:03.04 | linagee | ymmv |
11:03.15 | creativx | asteriskguru had some i think, voip-info also |
11:03.45 | linagee | dharrigan: whatever your ISP rates your upstream as, make sure you multiply by 80% or so for a real world number. ;) |
11:03.45 | *** join/#asterisk skyphyr (n=alanj@135.196.58.222) |
11:04.32 | dharrigan | would 512kbs be enough for about 8 simultaneous calls? |
11:04.40 | creativx | doubtfully |
11:04.47 | linagee | kilobitseconds? whoa |
11:04.54 | linagee | time and bandwidth in the same dimension |
11:04.54 | skyphyr | hi all - sorry for the dumb question - I'm just setting up my second asterisk box and going with 1.4.5 for it - wasn't sure how much of my knowledge from 1.2 would be applicable (and how much of the O'Reilly book) - any particular gotchas I need to be aware of? |
11:05.00 | creativx | 512 kb/s sure |
11:05.00 | creativx | hehe |
11:05.01 | creativx | Kb |
11:05.02 | creativx | KB |
11:05.04 | creativx | kBit |
11:05.05 | creativx | etc |
11:05.12 | linagee | oh. kilobits PER second. :P |
11:05.28 | linagee | (yes it does matter) |
11:05.44 | dharrigan | kilobits per second :-) (just getting the exact upstream figure now) |
11:05.53 | linagee | creativx: KB = kilobyte |
11:07.01 | creativx | I only deal in megabits |
11:07.18 | dharrigan | at the moment it's 448 Kbps (this is what my router is telling me) |
11:07.35 | linagee | creativx: hehehe |
11:07.39 | dharrigan | upstream and 4352 Kbps downstream |
11:07.43 | linagee | creativx: what about upstream? |
11:08.01 | creativx | linagee: 10/10 fibre |
11:08.06 | linagee | nice |
11:08.07 | dharrigan | 448 Kbps upstream, 4352 Kbps downstream |
11:08.26 | linagee | creativx: route some of that fiber my way. :) |
11:08.32 | creativx | wouldnt wanna try 8 simultaneous lines on that dharrigan |
11:08.39 | creativx | your users might be angry/disappointed |
11:08.40 | dharrigan | creativx: :-) |
11:08.47 | creativx | linagee: np let me just reroute the linknet |
11:08.52 | dharrigan | How many lines, say about 3? |
11:08.58 | linagee | dharrigan: like i said. no LESS than 640 kilobits/sec upstream for 8 lines |
11:09.08 | linagee | ideally 720 kilobits/sec up |
11:09.52 | linagee | dharrigan: i wouldn't run more than 4 lines off of your stated upstream. |
11:10.04 | dharrigan | Right, I'm beginning to understand. It's a small office, so really learning lots here about the best way to take. Thanks guys. |
11:10.06 | creativx | i would take 448 Kbps with a grain of salt |
11:10.19 | creativx | your ruter sure has a 448 Kbps link to the dslam |
11:10.29 | linagee | creativx: i would use traffic shaping to make sure that the 80 kilobits/sec was actually reserved for each phone IP. ;) |
11:10.29 | creativx | but from there, good luck with maxing that =) |
11:10.34 | dharrigan | linagee: I think I need to find some formulas for working these things out |
11:10.38 | creativx | i would buy more bandwidth |
11:10.39 | creativx | hehe |
11:10.44 | linagee | creativx: indeed |
11:10.48 | linagee | if that's an option |
11:10.51 | linagee | get at least 1mbit |
11:10.55 | linagee | if not more. hehe |
11:10.56 | dharrigan | yes, bandwidth is the key here I think. |
11:11.04 | dharrigan | big giganormous pipes |
11:11.06 | dharrigan | fat ones |
11:11.28 | Krooks | when I make linux26 on zaptel I get this -> grep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file or directory |
11:11.39 | linagee | creativx: in the USA here, they (cox cable modem) would charge me about $100 for 1mbit up, and about $400 for 2mbit up. |
11:11.40 | Krooks | make: *** No rule to make target `linux26'. Stop. |
11:11.53 | linagee | creativx: waaaay higher incrementally more for a slightly more upstream. lol |
11:12.05 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-49-41.inter.net.il) |
11:12.11 | linagee | creativx: it's sick that upstream and downstream don't cost the same. (downstream is typically less than $20 per megabit) |
11:12.53 | creativx | interesting |
11:12.58 | Krooks | oh its ok. I upgraded the kernel I forgot to reboot. |
11:13.02 | l2trace9999 | about 4-6 concurrent providing that no one is doing anything on the internet |
11:13.11 | Krooks | that sucks |
11:13.24 | creativx | linagee: i think our fiber was some 2k usd pr month |
11:13.28 | linagee | l2trace9999: 4-6 apples? |
11:13.33 | creativx | but they had to dig it first =p |
11:13.34 | l2trace9999 | yup |
11:13.40 | l2trace9999 | about 2 smurfs |
11:13.42 | linagee | creativx: how much per megabit? |
11:14.00 | linagee | creativx: did you say 10 megabit up, 10 megabit down? |
11:14.03 | l2trace9999 | or 1.66 smurfs |
11:14.03 | creativx | yes |
11:14.08 | creativx | 10 full duplex |
11:14.28 | creativx | cant remember what we are paying today |
11:14.30 | linagee | creativx: i get like 30mbit up / 30mbit down from my dedicated colo server. ;) |
11:14.47 | creativx | we are hosting in-house :) |
11:14.54 | linagee | (with only 500 gigs of total transfer though) |
11:15.24 | linagee | creativx: i think it's probably almost always more expensive to host in house |
11:15.31 | creativx | it sure is |
11:15.42 | linagee | so why do you do it? lol |
11:15.44 | creativx | i would prefer having our rack in a mountain hall |
11:15.54 | creativx | and remote access cards on all the servers |
11:16.06 | creativx | we are slowly migrating |
11:16.20 | creativx | just bought a new replication/live backup server that we are gonna host offsite |
11:16.26 | creativx | so i've started ;) |
11:16.29 | linagee | creativx: indeed. remote access cards on servers, service contracts for people to go onsite and repair stuff for you, server monkeys onsite to reboot machines that aren't rebooting through usual means |
11:16.55 | creativx | yep |
11:17.06 | *** part/#asterisk yonahw (n=yonahw@IGLD-83-130-49-41.inter.net.il) |
11:17.21 | creativx | but we have some internal servers as well as production servers |
11:17.29 | creativx | that wouldnt be that pain-free to move to a different location |
11:17.32 | linagee | creativx: then you don't have to pay for CRACs, internet access, floor space, data center staff, etc, etc. (usually all included in one bill. sometimes they make bandwidth a seperate cost) |
11:18.31 | creativx | theres always pros and cons |
11:18.44 | creativx | with the new 2u im already running out of rack space within a year |
11:18.55 | dharrigan | If I was to use the 729 codec instead of ulaw, that would lower my bandwidth requirements |
11:19.07 | dharrigan | 729 seems more efficient in encoding |
11:19.30 | linagee | dharrigan: test it and see how it sounds |
11:19.42 | dharrigan | I take it that it sounds, well, not that good ;) |
11:19.54 | linagee | ymmv |
11:20.34 | linagee | creativx: can you believe the interweb (cable modem internet) went down today at 10pm? wtf were they thinking |
11:20.53 | linagee | right in the middle of a demo at this guy's house and everything stops working. fuzz on the TV, no cablemodem connection, LOL |
11:21.00 | creativx | linagee: as long as nobody chops our fiber or any of the other backbone fibres im happy |
11:21.16 | linagee | creativx: is it dual ring? heh |
11:21.18 | creativx | then our customers tend to get whiney |
11:21.36 | creativx | its sad to say it but we have a single point of failure fiber to our serverroom |
11:21.45 | creativx | high availability my arse |
11:22.12 | linagee | creativx: it's highly available as long as you don't cut the fiber and blind all the admins with the laser shooting out. lol |
11:22.40 | linagee | imagine the latency on all the admins losing their eye sight and having no other admins within a 500 mile radius. lol |
11:22.48 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
11:23.01 | creativx | i can live with blind admins |
11:23.06 | linagee | hehehe |
11:23.13 | creativx | i would not have some 800 real estate agents calling us |
11:23.18 | creativx | asking why we are offline |
11:23.20 | dharrigan | btw all, here's the link for some bandwidth calculations : http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
11:24.47 | linagee | creativx: do people still think that when you have web panels, there's someone behind the scenes regenerating the way the website looks for you, then sends it back? LOL. millions of monkeys on typewriters. |
11:25.14 | creativx | i dont know what a web panel is :> |
11:25.23 | linagee | creativx: say like iGoogle |
11:25.44 | *** join/#asterisk torch (n=logan@200.99.7.98) |
11:26.28 | linagee | creativx: hundreds of monkeys |
11:26.40 | creativx | ah |
11:26.58 | creativx | i should get into web panels |
11:27.29 | linagee | creativx: i could say "DIV layer with some controls on it", but that's being a bit technical. i was trying to use a generic term. ;) |
11:27.39 | creativx | yes yes |
11:27.50 | creativx | i have a great need of simplifying our gui |
11:27.51 | torch | hi all ... just got my asterisk+digium TE110P working .... and now I'd like to have a integrated phonebook ...how can I do that? anyone? |
11:28.36 | torch | basically ...I want all my pstn users and voip users ...with the same phonebook .... |
11:29.31 | linagee | creativx: igoogle panel for asterisk control! lol |
11:31.02 | *** join/#asterisk tzanger (n=tzanger@gromit.mixdown.ca) |
11:31.12 | tzanger | morning |
11:31.17 | tzanger | citats: ping |
11:31.25 | torch | morning ... :-) |
11:31.36 | creativx | linagee: i was thinking more of the lines of our other app |
11:31.54 | Krooks | make: *** No rule to make target `linux26'. Stop. <-- I get this when i make linux26 in zaptel's dir . |
11:32.10 | Krooks | whats the usuall reason for this ? |
11:32.25 | tzanger | Krooks: just use make |
11:33.44 | Krooks | no need linux26. The manual says do make linux26. |
11:34.04 | Krooks | maybe thats old |
11:34.11 | creativx | yeah they fixed it in make |
11:34.14 | creativx | it autodetects |
11:34.21 | Krooks | ah ok. thanks |
11:34.46 | Krooks | somone better update all the manuals out there. they all says linux26.:) |
11:34.58 | tzanger | Krooks: yeah but nobody is supposed to read the documentation |
11:35.05 | tzanger | they're supposed to come in here and ask obvious questions |
11:35.47 | Krooks | :) |
11:35.54 | tzanger | you're breaking ranks, private |
11:36.21 | torch | so guys ... no idea about how to implement a integreted phonebook ? (pstn<->voip) |
11:36.54 | tzanger | what do you mean, integrated |
11:37.29 | Krooks | wow, make install zaptel downloading stuffs from digium. |
11:37.40 | Krooks | never seen make install does that kind of stuffs |
11:37.41 | tzanger | ? |
11:37.46 | tzanger | zaptel shouldn't do that |
11:37.48 | tzanger | asteirsk will |
11:38.08 | tzanger | the ec binaries are in the tarball |
11:38.23 | Krooks | maybe you should start reading the manual |
11:38.50 | tzanger | which manual are you talking about |
11:39.08 | torch | well ... someone from a asterisk extension ..will have a phonebook (got from somewhere) ..will all pstn extensions .. |
11:39.11 | torch | and vice-versa |
11:39.21 | Krooks | I don't know. you're the general. |
11:39.25 | tzanger | torch: I'm still not understanding |
11:39.29 | tzanger | asterisk has a directory app |
11:39.35 | tzanger | and it's trivial to have speed dials |
11:41.57 | torch | tzanger. Well .. let's say I have my softphone ... and now I want to open my phonebook right?! ..I want to see all my pstn extensions without having to add them by hand .. |
11:42.14 | tzanger | use a web based click-to-dial then |
11:42.30 | tzanger | or use a softphone which has remote directory capabilities (LDAP or provisioned like polycom's) |
11:42.43 | Krooks | is it easier to set up asterisk on Centos or on Suse ? |
11:42.53 | tzanger | Krooks: I find it trivial to install asterisk on slackware |
11:42.54 | torch | hmmm ..that's sounds much better ... |
11:42.56 | tzanger | everything else gets in the way |
11:43.05 | tzanger | but I'm a crotchety old man |
11:43.05 | torch | and suggestions about which softphone ? |
11:43.17 | tzanger | torch: don't know. don't use softphones for any serious work :-) |
11:43.17 | torch | I mean ..that has the remote directory capability |
11:43.25 | tzanger | I've used twinkle on linux and... well that's about it |
11:43.33 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
11:43.36 | torch | right ..right ... |
11:43.43 | Krooks | but between Centos and Suse, which one would be more trivial ? |
11:43.48 | tzanger | no idea |
11:43.58 | tzanger | if you're installing from source, I imagine they'd be hte same |
11:44.13 | tzanger | centos seems to be what most of the asterisk know-it-alls here use |
11:44.17 | Krooks | You know yesteday, I mentioned twinkle and nobody has ever heard of it. Some laughed at me. |
11:44.23 | tzanger | Krooks: I like it |
11:44.27 | tzanger | it's not a fucking gnome application |
11:44.34 | tzanger | and it's not half bad |
11:44.38 | Krooks | yeah fucking gnome |
11:44.45 | tzanger | (not half good either, but then again every softphone falls into that category IMO) |
11:44.48 | Krooks | hate the interface. |
11:44.57 | tzanger | if the softphones skin themselves to look like a phone... instant trashbin |
11:45.04 | tzanger | if the softphones take over my desktop... instant trashbin |
11:45.07 | Krooks | hate the gloomy color |
11:45.18 | torch | ok tzanger ...one more question .. .would it be possible to have asterisk to use a ldap directory ? |
11:45.19 | Krooks | idedisk is good |
11:45.25 | tzanger | yes there is res_ldap |
11:45.29 | tzanger | I have yet to play with it |
11:45.32 | tzanger | but it looks promising |
11:45.39 | tzanger | idefisk? yeah but it's iax2 :-) |
11:45.45 | tzanger | I gave up on that protocol a long time ago |
11:45.50 | tzanger | I used to be a HUGE proponent of it |
11:46.07 | tzanger | then Olle went and gave Asterisk symmetric RTP and almost every excuse I had to hate SIP went away |
11:46.28 | Krooks | I tried xlite with sip yesterday it wasn't as good as idedisk with iax |
11:46.33 | tzanger | around that same time file introduced IAX2 multithreading and since I was using trunk, it caused me severe problems, so the jump to SIP was pretty much a no-brainer |
11:46.52 | tzanger | xlite falls into the "skins itself into a phone on my desktop" category |
11:46.55 | tzanger | fucking idiotic |
11:47.01 | tzanger | I wish I could bitchslap people over the internet |
11:47.22 | tzanger | what <slap> the <slap> fuck <slap> where <slap> you <slap> thinking <slap> |
11:47.35 | Krooks | hehe |
11:47.49 | tzanger | oh yeah, if I can't control the softphone from dcop or dbus, instant trashbin |
11:48.11 | creativx | what is dcop |
11:48.11 | Krooks | kde has no softphone app ? |
11:48.13 | creativx | or dbus |
11:48.17 | tzanger | bonus points if I can run the softphone entirely from CLI and use aforementioned dcop/dbus to make my own screen |
11:48.20 | tzanger | Krooks: twinkle |
11:48.31 | tzanger | creativx: message busses |
11:48.34 | creativx | ah i see |
11:48.36 | Krooks | twinkle use kdelibs ? |
11:48.43 | creativx | yeah that would be the entire point of a softphone wouldnt it |
11:48.44 | tzanger | Krooks: I *think* so |
11:48.45 | creativx | full control of it |
11:48.54 | creativx | then integrate the crap outta it in your crm/whatever |
11:49.02 | Krooks | maybe its just Qt, |
11:49.09 | tzanger | no it does not, it's only qt |
11:49.13 | tzanger | I just ldd'd my twinkle app :-) |
11:49.20 | tzanger | I thought it could use kde as an option |
11:49.32 | tzanger | I'm actually more or less happy with twinkle |
11:49.37 | tzanger | qt-only means win32-able |
11:49.39 | Krooks | just Qt maybe bcuz for portibility sake |
11:49.41 | tzanger | and it's not too bad |
11:50.32 | tzanger | afctually |
11:50.35 | tzanger | it does have KDE integration |
11:50.38 | tzanger | 1.0.1 (released 19 may) |
11:50.39 | *** join/#asterisk matsk (n=mk@194.68.102.174) |
11:50.43 | tzanger | - Preload KAddressbook at startup. |
11:54.59 | Krooks | is there a asterisk-gui tarball ? |
11:58.45 | Krooks | ok done asterisk -vvvvvr |
11:58.50 | Krooks | done reload |
11:59.02 | Krooks | got the CLI> prompt |
11:59.10 | Krooks | now what ? |
11:59.27 | Krooks | I guess I got to read the manual sooner or later. |
12:04.27 | Krooks | whats the thing I have to configure so that a softphone can connect to it ? |
12:08.46 | Krooks | sip.conf is for sip and iax.conf is for iax. so if I'm using xlite. I edit sip.conf, right ? |
12:09.37 | creativx | yes |
12:09.48 | creativx | this is covered by asteriskdocs, wiki, etc |
12:13.58 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
12:15.22 | Krooks | ok I'm all set |
12:17.14 | *** join/#asterisk magikxx (n=tawandax@196.15.164.11) |
12:19.45 | *** join/#asterisk Greenbox (n=Brett@user-24-214-124-177.knology.net) |
12:28.33 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
12:30.59 | Krooks | fuck fuck fuck |
12:31.33 | Krooks | x-lite want libstdc++.so.5 and centos 5 has .6 |
12:31.51 | Krooks | soft linking it to .6 does not work. |
12:32.21 | ManxPower | many distros have older verisons of that available |
12:34.30 | Krooks | ah I knew its going to bite me for going wiht centos 5. |
12:34.37 | Krooks | someone in here did warn me. |
12:35.02 | Krooks | idedisk does not use sip ? |
12:35.05 | ManxPower | So Centos 5 does not have the older library avialable? |
12:35.14 | Krooks | good question |
12:35.46 | Krooks | will there be a conflict ? |
12:35.53 | ManxPower | no. |
12:36.01 | Krooks | oh |
12:36.08 | Krooks | lemme see |
12:36.39 | ManxPower | I had to use Mandriva's older C++ libs for an application and it worked just fine, not conflicts. |
12:36.42 | [TK]D-Fender | Krooks: Yes, idefisk does SIP, and on *nix, you should be using Ekiga anyways |
12:37.49 | Krooks | ManxPower: could I use centos4's libstdc++ rpm then ? |
12:38.06 | ManxPower | Krooks: No. |
12:38.11 | Krooks | [TK]D-Fender: let me see |
12:38.18 | Krooks | Ekiga |
12:38.44 | Krooks | I'm familair with xlite and idedisk. |
12:38.49 | ManxPower | I just said that many distros also include older stdlibc++. Do whatever you would do to search for all packages with libstdc++ in the package name. |
12:38.53 | Krooks | anyway, I'll give a try |
12:39.05 | ManxPower | on Mandriva the command would be "urpmi -y stdc++" |
12:39.37 | Krooks | I'm on centos 5. urpmi won't work here |
12:39.45 | [TK]D-Fender | Krooks: X-Lite is crap and limited. Idefisk is better & multi-protocol, Ekiga = 100% free and supports video, etc as well |
12:40.10 | ManxPower | Krooks: Mandriva has both libc++ 5 and 6 available. The default is to use 6, but you can also install 5 I think the Mandrive package is called libstdc++-compat or something like that. |
12:40.31 | ManxPower | Krooks: I know urpmi does not work on Centos. I was just giving an example. |
12:40.47 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
12:40.53 | Krooks | ManxPower: I'll search for it but [TK]D-Fender has kinda swayed my mind away from xlite |
12:41.25 | ManxPower | All softphones suck. (tm) (c) 2007 |
12:41.33 | Krooks | ManxPower: I know you know. I was just patronizing you, sorry. |
12:42.15 | Krooks | You can't trademark that. That was going to be my line. |
12:42.38 | ManxPower | Krooks: I've been using Asterisk much longer than you have 8-) |
12:43.09 | [TK]D-Fender | Krooks: True, all soft-phones suck, but X-lite is actually at the far end of the list :) |
12:43.13 | Krooks | can't argue with that |
12:43.33 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
12:45.29 | *** join/#asterisk cayorde (n=flexable@host121-81-dynamic.17-87-r.retail.telecomitalia.it) |
12:45.37 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
12:46.21 | Krooks | Wooo hoooo. ekiga is in the repository. |
12:46.34 | Krooks | Thanks [TK]D-Fender |
12:46.38 | [TK]D-Fender | Krooks: np |
12:47.10 | [TK]D-Fender | Krooks: Keep in mind the only uses you should have for a soft-phone is for a roaming laptop, or for basic internal testing of * |
12:47.36 | [TK]D-Fender | Krooks: Any fixed location past that deserves at least an ATA for the pittance they cost. |
12:47.46 | ReDNeQ | hey [TK]D-Fender how you doing.. |
12:47.57 | [TK]D-Fender | ReDNeQ: still breathing..... |
12:48.18 | ReDNeQ | you know the command to make the bots spit out the ports for nat or hte link to get me to the right doc |
12:48.50 | [TK]D-Fender | ~sipnat |
12:48.58 | jbot | it has been said that sipnat is for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
12:49.01 | [TK]D-Fender | ReDNeQ: 5060,10000-20000 all UDP |
12:49.05 | ReDNeQ | thanks... |
12:49.09 | ManxPower | [TK]D-Fender: I disagree about using a softphone for testing Asterisk. Softphones give Asterisk a bad reputation. |
12:50.08 | purserj | many times, a softphone is all we've got to test |
12:50.44 | [TK]D-Fender | ManxPower: This is a serious "don't bitch about it" item. People want to experience * and do so for free for a tiny bit at the start without having to worry about wiring, costs, etc while not knowing what to buy. And I cannot agree about it giving * a bad rep.... You don't see people in here complaining about them much. |
12:51.10 | basty | anyone got a CCBS Script (Callback on Busy) ? |
12:51.24 | [TK]D-Fender | ManxPower: Only points have been that "X-lite has no native transfer?!?!", "no G.729 for free?!", and even that is damn infrequent |
12:51.39 | [TK]D-Fender | basty: Don't bet on it. |
12:52.13 | [TK]D-Fender | ManxPower: Softphone lets you get right into * fast and easy. 3 parms no wiring, no running out of switch ports, etc. |
12:52.29 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
12:52.49 | Krooks | hmm, when I run ekiga for the first time. It says there an app thats already using the sip port. Could it be its refering to asterisk cause asterisk IS running on that box. |
12:52.51 | basty | [TK]D-Fender: so..there is no solution for that feature ? |
12:52.53 | [TK]D-Fender | ManxPower: BUT once they have used it through the end of that purpose they should go and get at least an ATA |
12:53.05 | ManxPower | Krooks: That is correct. |
12:53.17 | [TK]D-Fender | basty: Sure there is, more like don't bet on anyone HERE having it coded to hand to you. Get busy! |
12:53.20 | ManxPower | you don't run multiple applications using the same port on the same machine. |
12:53.42 | basty | [TK]D-Fender: So can you tell me where to get it ? =) |
12:53.48 | jkiff | I initially used couple softphones + my desk phone to test my first dialplan since I didn't have desk space/wiring/power for three phones at my desk. :-P |
12:53.49 | [TK]D-Fender | Krooks: Run Ekiga on 5061. in your soft-phones sip.conf entry you will specify "port=5061" |
12:53.50 | jkiff | It worked well. |
12:54.01 | Krooks | ManxPower: SO it mean generally I can't use a softphone on a asterisk machine. |
12:54.09 | ManxPower | jkiff: I did that too. I almost stopped using VoIP because of it. |
12:54.16 | [TK]D-Fender | jkiff: And did you get PAST that and get some hardware? |
12:54.25 | [TK]D-Fender | Krooks: Yes you can, see above |
12:54.41 | jkiff | [TK]D-Fender: Indeed. |
12:55.14 | Krooks | [TK]D-Fender: you mean in sip.conf for the user that gonna use Ekiga I put the line "port=5061" , is that right ? |
12:55.30 | jkiff | ManxPower: How come? My experience wasn't *that* bad. |
12:55.45 | Krooks | I mean in my case |
12:55.48 | ManxPower | jkiff: terrible audio quality, horrid user interface |
12:56.18 | [TK]D-Fender | Krooks: Correct. No issue |
12:56.43 | [TK]D-Fender | ManxPower: I've had flawless quality. It varies with super shitty sound cards and your headset. |
12:56.57 | ManxPower | [TK]D-Fender: Exactly |
12:57.08 | [TK]D-Fender | ManxPower: But no argument that any user interface IS cumbersomes compared to any hardware. |
12:57.13 | ManxPower | Other than the user interface issues, most softphone issues are hardware or OS issues. |
12:57.39 | ManxPower | [TK]D-Fender: Why? You do not need a massivly confusing interface to just dial a phone number |
12:57.41 | [TK]D-Fender | ManxPower: curmudgeon-- ;) |
12:57.57 | ManxPower | [TK]D-Fender: no. laxy. |
12:57.58 | ManxPower | ..er.. |
12:58.00 | [TK]D-Fender | ManxPower: C'mon..... |
12:58.01 | ManxPower | lazy that is |
12:58.20 | [TK]D-Fender | ManxPower: Ok, you're right. The IAXY is a flaming piece of shit ;) |
12:58.33 | ManxPower | [TK]D-Fender: at least we agree on that point 8-) |
12:59.17 | [TK]D-Fender | Only provisioned (?!) by * and only usable by it??!?! ISK |
12:59.20 | [TK]D-Fender | ICP* |
12:59.30 | [TK]D-Fender | ICK* |
12:59.33 | [TK]D-Fender | kldjhfasafasdsfdasfdaslfdhfiosewyr |
12:59.58 | ManxPower | [TK]D-Fender: I would have said that lack of codecs and DNS was the biggest issue |
13:00.09 | ManxPower | oh, and price |
13:00.20 | ManxPower | BBIAW |
13:00.29 | [TK]D-Fender | ManxPower: And look/feel/call features/number of ports |
13:00.42 | [TK]D-Fender | every/other/point/too~! |
13:02.19 | jkiff | ManxPower: Oh yeah, I've never used a soft phone who's interface didn't want to make me vomit. |
13:07.13 | Krooks | [TK]D-Fender: on ekiga , the registra is the ip on asterisk ? |
13:07.21 | *** join/#asterisk kannan (n=kannan@61.17.178.244) |
13:07.27 | [TK]D-Fender | Krooks: sounds about right |
13:07.48 | [TK]D-Fender | Krooks: Which would be "localhost" or the like in your case |
13:07.57 | Krooks | [TK]D-Fender: yep |
13:08.30 | Krooks | [TK]D-Fender: on themain screen after sip: |
13:08.46 | Krooks | I type the ip of the macghine also ? |
13:09.30 | [TK]D-Fender | Krooks: You mean to dial? |
13:09.44 | Krooks | yes |
13:10.01 | [TK]D-Fender | Krooks: No, you should be able to enter just teh # to dial and take its info from reg. |
13:10.10 | Krooks | [TK]D-Fender: I guess somewhere in Ekiga I have to tell it to use a different port too. |
13:10.26 | [TK]D-Fender | Krooks: shouldn't |
13:10.29 | Krooks | like 4321 for user test |
13:10.59 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
13:11.22 | kannan | hello , all . I want to use asterisk to connect outbound calls to an IVR of a bank that asks to enter account number followed by # key. I changed blinxfer in features.conf to ## in [featuremap] and re-started asterisk , but still the call gets blind transferred on a single # itself |
13:11.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:11.37 | Krooks | the number to dial |
13:12.11 | Krooks | [test] |
13:12.11 | Krooks | type=friend |
13:12.11 | Krooks | username=test |
13:12.11 | Krooks | secret=pwd_test |
13:12.11 | Krooks | host=dynamic |
13:12.12 | Krooks | context=tutorial |
13:12.14 | Krooks | port=5061 |
13:12.18 | Krooks | oops sorry |
13:12.27 | Krooks | whats the number to dial here ? |
13:13.18 | Krooks | I got user ivan already registered from one box using xlite. |
13:13.39 | Krooks | Bloody user test is on centos 5 and on the asterisk server itself |
13:14.36 | Krooks | I'm so close |
13:16.31 | [TK]D-Fender | Krooks: Thats only your SIP device DEFINITION. What you can DIAL is a whole other world. That is extensions.conf and is the HEART of Asterisk. |
13:17.48 | kannan | [TK]D=Fender -> any ideas on why i am not able to change blind xfer keys? |
13:18.08 | Krooks | exten => 4321,1,Dial(SIP/test) |
13:18.22 | Krooks | so what do I dial for this. |
13:19.29 | Krooks | but test is not registered yet |
13:19.34 | Krooks | ivan is registered |
13:20.11 | [TK]D-Fender | Krooks: Doesn't matter |
13:20.27 | [TK]D-Fender | Krooks: You should at least be able to dial that exten and watch it FAIL. |
13:20.44 | [TK]D-Fender | Krooks: Fialing to have the call acknowledged is something else entirely |
13:21.02 | [TK]D-Fender | Krooks: You'd dial "4321" |
13:21.13 | [TK]D-Fender | Krooks: Pastebin is your friend <---- |
13:21.15 | [TK]D-Fender | ~pb |
13:21.24 | jbot | well, 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 |
13:21.25 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
13:22.26 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:22.38 | Krooks | [TK]D-Fender: yes I can see it fail |
13:22.45 | Krooks | look |
13:23.01 | Krooks | Dial("SIP/ivan-0943a948", "SIP/test") in new stack |
13:23.01 | Krooks | [Jun 29 21:20:37] WARNING[20538]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
13:23.18 | Krooks | ivan is calling test |
13:23.24 | [TK]D-Fender | Thats absoluetly fine. You yourself said that you don't have test yet. |
13:23.42 | Krooks | I think its the port thing |
13:23.43 | [TK]D-Fender | Krooks: And * can't dial them accordingly. |
13:24.02 | [TK]D-Fender | Krooks: If its not registered * can't dial them. This is NORMAL. |
13:25.00 | pigpen | [TK]D-Fender, I am having issues with presence, polycom's and * 1.4.4. Would you have some time? |
13:25.08 | [TK]D-Fender | pigpen: Ask away |
13:25.21 | pigpen | k. gathering facts... |
13:27.02 | Krooks | why can't test register with asterisk ? is there somewhere I have to specify test to use port 5061 ? |
13:27.34 | pigpen | Here we go: |
13:28.07 | pigpen | sip.conf: limitonpeer=yes / call-limit=5 / busy-limit=1 |
13:28.32 | pigpen | Asterisk CLI: core show hints: "Watchers 1" |
13:28.45 | pigpen | sip.cfg: feature.1.name="presence" feature.1.enabled="1" |
13:29.00 | pigpen | polycom sip ver: 2.1.1 |
13:29.12 | pigpen | No buddy watch / presence. |
13:29.25 | [TK]D-Fender | pigpen: pastebin it all. |
13:29.28 | pigpen | k. |
13:29.30 | [TK]D-Fender | Krooks: You too. |
13:29.46 | [TK]D-Fender | Krooks: Along with add : "sip show peers" |
13:29.51 | *** join/#asterisk anthm (n=anthm@dhcp64-134-34-217.bwic.chi.wayport.net) |
13:29.51 | *** mode/#asterisk [+o anthm] by ChanServ |
13:30.33 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
13:30.33 | *** mode/#asterisk [+o mog] by ChanServ |
13:30.47 | rob0 | I thought port was a global setting in sip.conf ... can you set that per section? |
13:31.27 | [TK]D-Fender | rob0: Ye, it is phone specific. |
13:31.41 | [TK]D-Fender | rob0: If you've ever set up a dual port ATA you should know that. |
13:32.13 | *** part/#asterisk Andri[IS] (n=andri@hydrogen.calidris.com) |
13:32.19 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
13:32.31 | rob0 | I have a dual port ATA, but I didn't do anything special on the * side, they just register and tell * the port. |
13:33.37 | Krooks | http://pastebin.ca/595321 |
13:33.42 | [TK]D-Fender | rob0: You should tell it the port... |
13:33.49 | Krooks | Please take a look at http://pastebin.ca/595321 |
13:34.20 | [TK]D-Fender | Krooks: ok, test isn't registered. Thats your Ekiga, isn't it? |
13:35.01 | pigpen | http://pastebin.ca/595323 |
13:35.20 | *** join/#asterisk kvit123 (n=kvit123@203.209.31.86) |
13:35.23 | [TK]D-Fender | Krooks: Oh, and ditch the "username=" in your sip entries. Not needed |
13:35.30 | Krooks | yes |
13:35.31 | pigpen | I did not past the sip.cfg, phone.cfg, nor the <mac>.cfg, if needed, let me know. |
13:35.47 | rob0 | Hmmm, maybe that explains my 4-8 second delay when dialing ... |
13:35.52 | Krooks | is that the cause if the problem ? |
13:37.36 | [TK]D-Fender | pigpen: move "subscribecontext=from-sip", "call-limit=5" into each phone's entry and out of [general], and ditch "busy-limit=1", and "limitonpeer=yes". Also, your phones should now be "type=peer" |
13:37.59 | [TK]D-Fender | Krooks: Try dialing fred from your phone, and check your reg settings. |
13:38.07 | pigpen | k, did the type=peer change with * 1.4 ? |
13:38.20 | [TK]D-Fender | pigpen: Yeah... various silly things... |
13:38.20 | Krooks | fred ? |
13:38.25 | Krooks | who is fred ? |
13:38.33 | pigpen | k. kinda figured. |
13:38.38 | pigpen | fred is dead. |
13:38.49 | [TK]D-Fender | Krooks: Err..."ivan" ..... I have no explanation for the name shift ;) |
13:38.50 | tzanger | Dave's not here! |
13:39.05 | [TK]D-Fender | tzanger: ".... what are you doing, Dave?" |
13:39.12 | tzanger | wrong dave |
13:39.18 | tzanger | "No it's be, Dave, I got the stuff" |
13:39.21 | Krooks | [TK]D-Fender: security check error |
13:39.22 | tzanger | er it's me |
13:39.41 | Krooks | [TK]D-Fender: test is not regiostered. how can he call ivan . |
13:39.45 | pigpen | Dave is in line at the apple store, waiting for his iPhone.... |
13:40.05 | [TK]D-Fender | Krooks: You don't actually need to be registered to place a call... |
13:40.20 | Krooks | [TK]D-Fender: Does this all got to do anyting with the port thing earlier on? |
13:40.27 | [TK]D-Fender | Krooks: enable "sip debug" and pastebin your reg attempt. |
13:40.53 | [TK]D-Fender | Krooks: it might. You also didn't specify port 5061 in your entry, and I don't know if you screwed something else up in your Ekiga settings. |
13:41.03 | [TK]D-Fender | Krooks: I've used ti maybe twice... and don't have it here |
13:41.19 | [TK]D-Fender | pigpen: OpenMoko FTW |
13:41.24 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
13:41.26 | [TK]D-Fender | pigpen: I'd buy one. |
13:42.37 | Krooks | [TK]D-Fender: didn't specify where ? |
13:42.39 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
13:42.40 | [TK]D-Fender | Krooks: OH... and ditch THIS :" bindaddr = 192.168.100.1" for this : "bindaddr = 0.0.0.0" and completely restart * |
13:42.52 | [TK]D-Fender | Krooks: under [test] |
13:43.19 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
13:43.27 | pigpen | openmoko? |
13:43.32 | pigpen | ok...googleing. |
13:43.52 | Krooks | line 16 port=5061 |
13:43.52 | pigpen | shit...what did you just get me into. |
13:43.54 | [TK]D-Fender | pigpen: www.openmoko.org |
13:43.59 | [TK]D-Fender | pigpen: www.openmoko.com as well |
13:44.23 | [TK]D-Fender | Krooks: vnm I'm f'n blind today ;) |
13:44.29 | [TK]D-Fender | Krooks: But check out the rest! |
13:44.35 | [TK]D-Fender | nvm* |
13:44.37 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
13:44.57 | pigpen | [TK]D-Fender, any hardware out for it? |
13:45.29 | [TK]D-Fender | pigpen: it IS hardware. Arrives this fall |
13:45.49 | [TK]D-Fender | pigpen: One Smart-Phone to rule them all! |
13:45.57 | coppice | i wonder if it will ever really arrive. it seems rather late |
13:46.01 | pigpen | Yeah..as you want it, program it. |
13:46.14 | pigpen | it has to sync with my mac! haha |
13:46.15 | [TK]D-Fender | coppice: iPhone isn't entirely here yet either, big deal. |
13:46.32 | Krooks | oh man |
13:46.39 | [TK]D-Fender | ^%@# MAC. Its pretty and all but I refuse to let my devices own ME. |
13:46.57 | pigpen | looks sweet...I hope the openmoko comes with more storage. |
13:47.07 | pigpen | ok..back to presence... |
13:47.10 | [TK]D-Fender | pigpen: SD/microSD <- |
13:47.14 | coppice | isn't the iPhone supposed to be here a few hours from now? |
13:47.17 | [TK]D-Fender | pigpen: and more to come. |
13:47.32 | syzygyBSD | I have one right here |
13:47.36 | pigpen | coppice, yeah...6:00pm |
13:47.44 | syzygyBSD | ok.. maybe not |
13:47.49 | [TK]D-Fender | coppice: Oh... and like the iPhone doesn't own your ass with the bundled PLAN, lack of 3rd aprty apps, and do you trust their batter, or lack of SIM? :) |
13:47.52 | pigpen | [TK]D-Fender, thanks for the tip. |
13:47.56 | syzygyBSD | dirty $500 phones |
13:48.05 | [TK]D-Fender | batter* |
13:48.06 | syzygyBSD | think I will just go to japan and get one from there |
13:48.09 | [TK]D-Fender | battery* |
13:48.11 | [TK]D-Fender | askldsdhfafdshasf |
13:48.18 | coppice | I think the iPhone is more likely to cut your ass with broken glass |
13:48.29 | syzygyBSD | Iphone doesn't have a sim? |
13:48.37 | syzygyBSD | I thought it was gsm... |
13:48.39 | syzygyBSD | hmmm |
13:48.40 | coppice | it does have a sim |
13:48.41 | [TK]D-Fender | syzygyBSD: Built-in only. |
13:48.51 | [TK]D-Fender | syzygyBSD: Foget access or exchange :) |
13:48.54 | Krooks | I got it to work |
13:48.58 | coppice | no. a normal plug in sim |
13:49.02 | [TK]D-Fender | syzygyBSD: Your provider will OWN YOU < |
13:49.23 | [TK]D-Fender | syzygyBSD: And have you read about the nasty termination fee even iff you paid FULL PRICE for it? |
13:49.24 | syzygyBSD | which is why I switched from cingular shortly after they bought at&t |
13:49.32 | [TK]D-Fender | coppice: I've read differently on that. |
13:49.56 | coppice | its a software locked SIM like almost every other phone in north america |
13:49.58 | syzygyBSD | had the best plan I could imagine with at&t though |
13:50.23 | coppice | living where I do, I find it hard to imagine everyone tolerating locked phone, and tied packages |
13:50.43 | syzygyBSD | I haven't had a locked phone for 3 years |
13:51.20 | pigpen | [TK]D-Fender, buddy watch didn't work with a simple reload, so I am rebooting the polycom's, as this seemed to be a "thing" with previous ver's of *. |
13:51.49 | coppice | a locked phone would be about as popular as vista here |
13:52.40 | syzygyBSD | lol where you at? |
13:52.54 | coppice | HK |
13:52.58 | [TK]D-Fender | coppice: You shouldn't joke about Vista like that. They sold dozens of copies! |
13:53.02 | syzygyBSD | ahh, that makes sense |
13:53.09 | syzygyBSD | I have one of them! |
13:53.19 | syzygyBSD | course, it sits on my test box unused... |
13:53.55 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
13:53.56 | [TK]D-Fender | My phone is rather locked, but very nice, was heavily subsidized and my plan is SICK :) |
13:54.33 | naitram | [TK]D-Fender: yesterday, you were trying to help me fix a problem with automon from and active sip call. You suggested the Read() app to see if i got correct dtmf tone awareness from *. I do, it works. |
13:54.54 | [TK]D-Fender | naitram: Ok, well keep wiki-ing, I have no personaly experience with it. |
13:55.22 | naitram | The monitor app also works from the dial plan, I just can't get anything invoked from an active call to work |
13:56.26 | naitram | anyone, have experience setting up 1 touch monitoring with sip clients |
13:56.27 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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13:59.58 | anthm | I never set it up but I wrote that feature =p |
14:01.27 | naitram | anthm: any idea why I cant start any app on an active sip call via dtmf tones? |
14:02.03 | *** part/#asterisk scud (n=scud@c-68-62-219-34.hsd1.al.comcast.net) |
14:02.13 | anthm | all I can remember is you need the flag in your dial app D or d or something for allow dtmf and a key defined in the config file |
14:02.48 | anthm | if you choose * you sometimes end up out of luck cos that one is hard coded in a few places esp before the call is answered |
14:02.57 | BugKhaM | I got BPV count: 333 CRC4 error count: 4 E-bit error count: 986 FAS error count: 6 |
14:03.01 | BugKhaM | in my /proc/zaptel/3 |
14:04.10 | BugKhaM | I just made E1 Cross over cable to link btw span 3 & 4 |
14:05.01 | naitram | anthm: so, which sources files? |
14:05.13 | kannan | anthm -> maybe you can surely help me, i set blindxfer to ##, but it still transfers on a single # |
14:05.41 | kannan | instead of sending the # as a DTMF |
14:06.17 | kannan | i have defined tT flags in the Dial cmd |
14:07.36 | pigpen | [TK]D-Fender, thanks...that did the trick. |
14:08.12 | naitram | so, are these features broken, mis-documented or what? |
14:08.15 | torch | guys .. got a weird noise when dialing from my pstn to my asterisk extensions ... the integretion is mad using a Digium TE110P |
14:09.04 | torch | when in asterisk CLI I do 'zap show status' the IRQ is quite high ... ideas? |
14:09.10 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-164-154.buff.east.verizon.net) |
14:12.54 | tzanger | what... the... fuck |
14:12.58 | tzanger | I've got PRI DTMF issues again |
14:13.19 | tzanger | and my system hasn't changed or even bene restarted in what, 2 weeks |
14:14.58 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
14:17.13 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
14:17.19 | BSD_Tech | ok morning |
14:17.29 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
14:17.30 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
14:17.30 | BSD_Tech | I have a issue driving batty |
14:18.05 | BSD_Tech | everytime I hit the #key it says please enter the extension then press pound |
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14:19.01 | *** mode/#asterisk [+o blitzrage] by ChanServ |
14:19.01 | BSD_Tech | where is this defined I have nothing in my dialplan defining the # sign |
14:19.01 | IOscanner | You have transfer enabled in the gloabal settings |
14:19.01 | BugKhaM | is it possible to make a cross over E1 to link between two ports of my quad cards? |
14:19.01 | IOscanner | you using AH or Trixbox? |
14:19.01 | BugKhaM | I just tried and it gave me some errors |
14:19.07 | BSD_Tech | nope |
14:19.14 | BSD_Tech | just aasterisk 1.4.5 |
14:19.15 | _VoiceMeUp_COM | BSD_Tech .. featurs.conf ? |
14:19.39 | BSD_Tech | no # key enabled but double checking |
14:20.35 | IOscanner | I have a problem. I have a queue and when people call in it rings once then sends the callers back to the IVR |
14:21.08 | [TK]D-Fender | BSD_Tech: "tT" in youd dial statement.... |
14:21.41 | BSD_Tech | not # enables in feathers.conf |
14:22.11 | IOscanner | I even have ringall for the ring strategy. I do know that fewest calls does this. |
14:22.20 | [TK]D-Fender | BSD_Tech: Are you using either of those dial options? |
14:22.27 | BSD_Tech | no |
14:22.36 | [TK]D-Fender | IOscanner: pastebin is your FRIEND <---------------- |
14:22.42 | BSD_Tech | this is a standard out of the box asterisknow setup |
14:22.51 | [TK]D-Fender | BSD_Tech: show some CLI output. |
14:23.05 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
14:23.09 | [TK]D-Fender | BSD_Tech: GUI?! *shudder* Trust-- |
14:23.52 | kannan | BSD_Tech |
14:24.11 | kannan | BSD_Tech -> it is the default blindxfer defined in features.conf |
14:24.19 | kannan | lol, i am asking about this too |
14:24.32 | kannan | how to change the sequence in featuremap |
14:24.37 | kannan | it is not updating for me |
14:24.52 | [TK]D-Fender | pigpen: You're welcome |
14:25.09 | kannan | i need the tTo , as i am using vicidial |
14:25.20 | [TK]D-Fender | kannan: pastebin EVERYTHING after restarting * and retesting. |
14:25.25 | pigpen | OpenMoko is pretty cool. |
14:25.35 | [TK]D-Fender | pigpen: See? :) |
14:25.37 | kannan | ok sure thanks a lot |
14:25.50 | [TK]D-Fender | kannan: If you want help, we shouldn't even have to ASK <0 |
14:26.01 | BSD_Tech | nope |
14:26.27 | BSD_Tech | hold on brb doorbell |
14:26.34 | [TK]D-Fender | BSD_Tech: .... PASTEBIN :) |
14:28.22 | Krooks | what the ccommand to shutdown asterisk ? |
14:28.40 | Krooks | exit ? |
14:28.42 | [TK]D-Fender | Krooks: "stop now" |
14:29.17 | BugKhaM | [TK]D-Fender: sorry to interrupt, u know if I can make a cross over cable and link between two ports of a quad card? |
14:29.18 | Krooks | stop gracefully |
14:29.30 | [TK]D-Fender | BugKhaM: Sure |
14:29.35 | [TK]D-Fender | Krooks: |
14:29.42 | Krooks | [TK]D-Fender: I found a laptop with windows I tested it. |
14:29.47 | [TK]D-Fender | Krooks: "stop with extreme prejudice" :D |
14:29.47 | *** join/#asterisk luckyone (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
14:30.02 | Krooks | Wonderful . Its like a whole new world of stuffs to play with |
14:30.20 | [TK]D-Fender | Krooks: So you did a SIP>SIP call nice and happily? |
14:30.24 | Krooks | yes |
14:30.26 | luckyone | when tryin to connect to my running * process, I get Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
14:30.48 | [TK]D-Fender | Krooks: Congrats, not to embellish your setup, add VM, learn to make IVR's, funny prompts, conferencing, etc |
14:30.54 | *** join/#asterisk pogo123 (n=hh@buero-gw.dortmund.loca.net) |
14:30.56 | Krooks | One question . If I'm behind a firewall. The only port I have to open is 5060 ? |
14:31.01 | [TK]D-Fender | luckyone: You either lack rights, or its not running. |
14:31.11 | [TK]D-Fender | Krooks: Here : |
14:31.13 | [TK]D-Fender | ~sipnat |
14:31.13 | jbot | sipnat is probably for for more information about configurtion of Asterisk with SIP behind NAT, see http://voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
14:31.15 | BugKhaM | [TK]D-Fender: and /proc/zaptel/* should not have any errors right? |
14:31.22 | luckyone | this happens every time I reboot my machine - I recreated the directy /var/run/asterisk then chown'd it to user asterisk |
14:31.25 | penguinFunk | Krooks: 5060 for sip, 10000-20000 for RTP |
14:31.26 | Krooks | [TK]D-Fender: I will. Need to read the manual first. I did all this without reading the manual. |
14:31.32 | [TK]D-Fender | Krooks: 5060,10000-20000 all UDP and a whack of settings for sip.conf |
14:31.55 | Krooks | oh man |
14:32.11 | [TK]D-Fender | Krooks: not so bad. I'm going to prime JBOT for this shortly. |
14:32.30 | Krooks | ok thanks |
14:32.32 | Krooks | bye |
14:32.47 | kannan | hmm, i think i know the problem now, lol, lrt me try , brb |
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14:34.45 | BSD_Tech | I will pick up when I get back |
14:34.48 | BSD_Tech | thanks |
14:35.38 | luckyone | [TK]D-Fender: I think you helped me fix this a few weeks ago so I could reattach to it |
14:36.02 | luckyone | [TK]D-Fender: I know it is running, my sip clients can connect to is, calls work, voicemail is up, etc |
14:36.54 | pogo123 | Hi! I'm trying to use a Zaptel device driver which was originally released for Zaptel 1.2.6 with Zaptel 1.2.16. But the kernel Oops'es as soon as I pick up the phone. Did Digium make any internal changes to Zaptel so that I have to adjust my driver to be compatible with the 1.2.16? |
14:37.24 | luckyone | [TK]D-Fender: I don't understand why /var/run/asterisk gets blown away on restart. Further, I don't understand what mods I need to make to /etc/init.d/asterisk so that start runs the application so I can always attach to it.... |
14:37.42 | [TK]D-Fender | luckyone: Who are you calling it as? |
14:37.50 | luckyone | [TK]D-Fender: root |
14:37.53 | [TK]D-Fender | luckyone: And what distro? |
14:38.01 | luckyone | [TK]D-Fender: kubuntu |
14:38.38 | [TK]D-Fender | luckyone: EEK. They don't even use STD init anymore IIRC. That new bootup sequence thingy... |
14:40.21 | luckyone | [TK]D-Fender: it is somehow getting called automatically, I don't have to run asterisk -vvvc -U asterisk -G asterisk or whatever to start it after reboot |
14:41.13 | [TK]D-Fender | luckyone: This kind of item is out of my league.... maybe someone else will know (they'd have responded most-likely already). Get goolging.... |
14:41.36 | luckyone | [TK]D-Fender: hehe - yeah... =) |
14:47.20 | rob0 | luckyone, it's just bash. Read the script and see what it does. Or, don't use it at all. |
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14:48.23 | luckyone | rob0: I see what it does, it makes a call to safe_asterisk and when I look at that I don't understand why /var/run/asterisk/asterisk.crtl isn't being created |
14:48.25 | rob0 | You can disable asterisk in whatver you use to manage your services, and then just start it in rc.local as you need. |
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14:52.44 | magikxx | Cresl1n around ? |
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14:56.32 | *** join/#asterisk Traiana (n=simonc@12.39.229.194) |
15:01.54 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
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15:12.24 | Traiana | hi to all, i was wondering if ther is anybody that could help me with a issue with the conference room option? |
15:12.39 | Zeeek | ask away! |
15:13.02 | Traiana | does the set marked user option allow the room to start when the marked user join the room? |
15:13.28 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
15:14.23 | [TK]D-Fender | Traiana: ....HUH?! |
15:14.56 | Zeeek | what huh? |
15:15.03 | Traiana | i have turn the wait for marked user and set makred user on using the asterisknow gui but the room still will allow the conversation to start without a marked user joining. |
15:17.37 | [TK]D-Fender | Traiana: This is not the GUi support channel, please read the topic |
15:18.02 | Zeeek | [TK]D-Fender you're in a good mood :) |
15:18.45 | [TK]D-Fender | Zeeek: Under my usual quote for capitalization and swearing ;) |
15:18.46 | [TK]D-Fender | quota* |
15:19.06 | [TK]D-Fender | Zeeek: And I have not figured out mIRC style font control with Chatzilla ;) |
15:19.21 | Zeeek | there's font control? |
15:19.35 | Traiana | ok, no problem , thank you for the assistance. |
15:19.37 | [TK]D-Fender | Zeeek: Bold & colour, possibly italics. |
15:20.06 | Zeeek | I don't recall that, I only used zilla for a few times. It blew up too many times. |
15:20.48 | [TK]D-Fender | Zeeek: Very stable for me, and saves on installing too many extra apps. |
15:20.51 | Zeeek | krdian |
15:21.00 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
15:21.12 | Zeeek | I like extra apps |
15:21.19 | Zeeek | they introduce nice security holes |
15:21.45 | Zeeek | in addition to those furnished with the os |
15:22.04 | Zeeek | krdian hello |
15:22.30 | kannan | ok i can't get to send the # to the IVR. the call gets disconnected. I have pasted in http://pastebin.ca/595492 |
15:23.02 | kannan | [TK]D-Fender -> as i am very new to the chats , please excuse any mistakes in my part |
15:23.11 | BSD_Tech | I found the issue |
15:23.27 | BSD_Tech | its my macro-user-callerid |
15:23.29 | kannan | BSD_Tech , waht is it? |
15:23.29 | [TK]D-Fender | Zeeek: Gruyere++ |
15:23.50 | BSD_Tech | I ment my issue |
15:23.53 | BSD_Tech | sorry |
15:24.09 | BSD_Tech | back to writing and fixing my dial plan |
15:24.28 | anthony] | Anyone here own a street bike? |
15:24.45 | Zeeek | if it had a GUI... |
15:25.10 | [TK]D-Fender | kannan: What IVR, there IS no IVR in there. |
15:25.26 | Traiana | is it possible to setup a user to be a moderator of a conference room ?? |
15:26.01 | [TK]D-Fender | Traiana: the admin has a PIN or you can eter immediately as admin if you choose to do so in your dialplan. |
15:28.27 | kannan | i meant that i am making an outbounf to an IVR where i have to send the # as dtmf |
15:28.33 | kannan | but when i do it gets cut |
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15:28.59 | [TK]D-Fender | kannan: Show me the call's complete CLI output at verbose 10 |
15:29.09 | kannan | ok sure |
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15:34.00 | kannan | [TK]D-Fender - > http://pastebin.ca/595526 |
15:35.05 | kannan | very sorry pl ignore i will re-send |
15:35.10 | kannan | sorry |
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15:37.32 | kannan | http://pastebin.ca/595532 |
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15:40.52 | kannan | it just shows the call connected, when we send the digits all thew numerals are fine, when we hit # , it gets disconn |
15:44.20 | [TK]D-Fender | kannan: For all I know its the OTHER SIDE that's choosing to hang up on you. |
15:45.05 | kannan | :( , when i register an x-lite directly to the service provider , it goes fine |
15:45.26 | kannan | the callee is an automated IVR |
15:46.10 | kannan | i set the SIPtrunk, so that the tTo is not used in the Dial cmd, but still it gets disconnected on pressing the hash key |
15:46.53 | [TK]D-Fender | kannan: You'll need to include SIP debug info next and if that doesn't say why, then core debug |
15:47.19 | kannan | ok then , sure thanks alot |
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15:57.55 | BSD_Tech | Zeeek whats the irc channel for the talk shoe |
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15:58.21 | Zeeek | asterisk-users-conference |
15:58.45 | daveburr | does anyone know of an app for scheduling asterisk meet-me sessions? |
15:59.30 | [TK]D-Fender | daveburr: .... |
15:59.35 | [TK]D-Fender | ~toywy |
15:59.35 | jbot | i heard toywy is The one you write yourself. |
16:03.29 | IOscanner | I have an IAX trunk between two boxes. I need to pass a FROM_DID value. for now it is passing s@.... |
16:03.38 | IOscanner | how can I define this in the IAX2 trunk? |
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16:08.21 | javar | :O |
16:09.09 | [TK]D-Fender | IOscanner: Look at your DIAL...... |
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16:16.45 | irule | h there, if I run install_amp on freepbs installation for the second time, it will not ask me any questions! how can I change this? thanks! |
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16:16.48 | jer | urgh, i'm having a hellova time trying to turn off voicemail for a realtime user stored in a mysql database |
16:17.28 | Corydon76-work | Uh, "turn off voicemail"? |
16:18.29 | jer | yeah. the user has requested to have no voicemail, i can't turn it off (i didn't set up this system ,and i'm only acting as a fill-in until we hire an actual experienced asterisk guy) |
16:18.56 | jer | just driving me up the wall |
16:19.10 | Corydon76-work | Is your dialplan stored in the flat file? |
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16:19.54 | Corydon76-work | Basically, you just need to add a GotoIf prior to the call to Voicemail |
16:19.55 | jer | Corydon76-work, yes |
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16:20.23 | Corydon76-work | GotoIf($[${EXTEN} = 123]?hangup) |
16:20.32 | Corydon76-work | and then define a hangup label |
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16:20.54 | Zeeek | if anyone has anything to say today, #asterisk-users-conference or http://x2z.eu |
16:21.04 | jer | hrmm here i am trying to do it a complicated way |
16:21.06 | Zeeek | Beginning in 10 |
16:21.09 | jer | Corydon76-work, i'll give that a go |
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16:21.31 | Corydon76-work | or, if your Dial timeout is set to a variable, Set(${IF($[${EXTEN} = 123]?dialtimeout:foo)}=9999) |
16:21.32 | Zeeek | ~seen JerJer |
16:21.34 | jbot | jerjer <n=PhatJ@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk, 7d 20h 26m 49s ago, saying: 'dioedu: that sucks'. |
16:21.46 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-252-162.phlapa.fios.verizon.net) |
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16:22.14 | *** mode/#asterisk [+o Qwell[0]] by ChanServ |
16:22.40 | Corydon76-work | jer: and, if you want multiple people to be able to do that via a flag, you could create that flag in the db, then query it in the dialplan |
16:22.56 | quidpro | Hmm, anybody know how to stop the voicemail "alter tones" that pop up every couple minutes on Polycoms? |
16:23.09 | Corydon76-work | jer: I'd highly recommend that flag if a second person requests the same behavior |
16:23.11 | quidpro | oops, alert tones |
16:23.30 | jer | Corydon76-work, ok, thanks |
16:24.12 | bpiper | I am trying to create a PHP script to call a person & read some digits back to them with the manager API but when I try running the script, it just shows "manager logged in" and "manager logged off" and does nothing. Here is my script... would anyone like to give me a hand? |
16:24.12 | bpiper | http://pastebin.com/939055 |
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16:30.42 | matdon | hi |
16:32.06 | [TK]D-Fender | quidpro: In sip.cfg set the sound entry to "silence" |
16:32.32 | _VoiceMeUp_COM | <PROTECTED> |
16:32.37 | _VoiceMeUp_COM | aint that a nat issue ? |
16:32.52 | Traiana | does anybody have any experience in setting a conference room to wait for the moderator to arrive to start? |
16:33.35 | drako | anyone here doing call monitoring with TDMs card? |
16:33.59 | bpiper | Can anyone give me a hand with a php script calling to the manager API? http://pastebin.com/939055 |
16:34.09 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) |
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16:34.31 | *** mode/#asterisk [+o mog] by ChanServ |
16:35.48 | quidpro | TK: Hmm... I can't figure the exact tag... is se.appLocalEnabled="0" the one? |
16:37.07 | [TK]D-Fender | quidpro: <MSG_WAITING |
16:37.15 | quidpro | TK: Let me look... |
16:37.26 | *** join/#asterisk RazaMetaL (n=razameta@200.93.220.27) |
16:37.29 | RazaMetaL | hi guys |
16:37.47 | RazaMetaL | I've two questions regarding my new asterisk install |
16:38.06 | RazaMetaL | is better to use alphabetic extensions or numeric? |
16:38.11 | Strom_M | numeric |
16:38.17 | Strom_M | you can actually dial those |
16:38.30 | *** join/#asterisk perf3kt (i=perf3kt@149.166.34.171) |
16:38.37 | RazaMetaL | i'm planning to integrate my asterisk with jabber and want to use alphabetic usernames at jabber |
16:39.05 | [TK]D-Fender | bpiper: "fputs($socket, "Secret: *****\r\n");" <- Extra "\r\n" is called for IIRC to terminate the login. |
16:39.23 | torch | does anyone know any softphone to be used in a smartphone like a Palm Treo or P910 sony-ericsson? |
16:39.31 | RazaMetaL | I'm testing alphabetic usernames and giving one numeric extension for every username |
16:39.33 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
16:39.34 | RazaMetaL | it is correct? |
16:39.45 | twitchnln | good morning everyone |
16:39.48 | bpiper | TKD Fender, you're saying I need to make it \r\n\r\n? |
16:40.10 | [TK]D-Fender | bpiper: I believe so |
16:40.16 | bpiper | ok, thanks, I'll try that |
16:40.25 | RazaMetaL | I've another option that consist on install freepbx beta for 1.4 support |
16:40.32 | Strom_M | RazaMetaL: are you only going to be using jabber? |
16:40.33 | RazaMetaL | what do you think is better ... |
16:40.43 | Strom_M | ugh, don't touch freepbx |
16:40.46 | irule | h there, if I run install_amp on freepbs installation for the second time, it will not ask me any questions! how can I change this? thanks! |
16:40.52 | RazaMetaL | Strom_M, yes.. in the future |
16:41.37 | bpiper | that did the trick... thanks TDK-Fender |
16:41.38 | twitchnln | i was looking @ http://www.trixbox.org/forums/trixbox-forums/open-discussion/multiple-queues-agent-login-logout and am having difficulty getting the context setup so that extensions can dial them |
16:41.55 | [TK]D-Fender | irule: You're repeating yourself like a broken record again, and You already know its not supported here. |
16:42.33 | [TK]D-Fender | bpiper: Np |
16:42.48 | *** part/#asterisk bpiper (n=bpiper@ip-207-145-7-194.atl.megapath.net) |
16:42.57 | drako | irule, #freepbx |
16:43.03 | irule | [TK]D-Fender complete silence sounds like nobody was arround at the moment of the first message ;) |
16:43.20 | [TK]D-Fender | irule: You are being an ass and you KNOW it. |
16:43.36 | Strom_M | twitchnln: #trixbox |
16:44.39 | irule | thanks drako |
16:44.41 | RazaMetaL | it is recommendable to use freepbx? or you recommend to deal with the .conf files ? |
16:44.43 | twitchnln | thanx |
16:45.09 | perf3kt | razametal: its straight cli here |
16:45.39 | RazaMetaL | perf3kt, ok |
16:45.45 | drako | can you use variables like in extensions.conf in voicemail.conf? |
16:46.07 | *** part/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
16:46.45 | [TK]D-Fender | drako: Nope. |
16:47.06 | drako | damnit. |
16:48.12 | [TK]D-Fender | drako: Now trying thinking FORWARD. What do you want to DO? |
16:48.18 | Zeeek | http://www.dataevolution.com/dectop%20info%202.htm |
16:48.20 | Zeeek | oops |
16:49.12 | vn | ~book |
16:49.13 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
16:49.55 | drako | [TK]D-Fender, I don't want to repeat this line on every entry ,attach=yes|saycid=no|envelope=no|delete=no|maxmsg=2000 |
16:50.23 | [TK]D-Fender | drako: Copy&paste.. get over it! |
16:51.50 | Strom_M | drako: you can do it in the general settings too |
16:51.54 | Jingles | you know, that book is pretty decent for setup, laying out your dialplan, etc. |
16:52.05 | Jingles | however, there is one major bit of info it seems to lack. |
16:52.09 | *** join/#asterisk guillote_GNU (n=guillote@190.7.30.135) |
16:52.14 | Jingles | troubleshooting call quality issues. |
16:52.30 | vn | we can't get everything! |
16:52.31 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
16:52.42 | vn | go back helping Santa Jingles :p |
16:53.31 | Traiana | can anybody help me ? I am trying to setup a user to become a leader of a conference room ? any idea on how to do that ?? |
16:53.52 | *** join/#asterisk ManxPower (n=manxpowe@90.sub-70-220-188.myvzw.com) |
16:54.32 | [TK]D-Fender | Traiana: Go read the INSTRUCTIONS : "show application meetme" |
16:54.47 | drako | Strom_C, hmm good hint. |
16:54.59 | drako | Strom_M, i thought it was only a per line params |
16:55.40 | [TK]D-Fender | drako: You should occasionally read the SAMPLE files ;) |
16:56.21 | *** join/#asterisk techie (n=gus@antibala.net) |
16:56.43 | drako | yeah |
16:57.00 | drako | but most of the samples files sucks. |
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16:58.46 | Zeeek | http://www.lulu.tv/?p=11368 |
17:00.14 | [TK]D-Fender | drako: I never said "use as is", did I? Hell no, but the options are DOCUMENTED there. |
17:00.58 | drako | yes yes |
17:01.04 | drako | i get ur point, is a good one. |
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17:03.21 | [TK]D-Fender | Zeeek: PRICELESS |
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17:04.05 | Zeeek | needs a little trimming, but you get the idea |
17:05.29 | [TK]D-Fender | Zeeek: Indeed.... those wires were WAY too long ;) |
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17:16.51 | _VoiceMeUp_COM | when does cluecon end |
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17:23.44 | Zeeek | Karlito |
17:24.38 | neoalex | ok... now I know most of you guys hate grandstream phones but I'm having a problem with one |
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17:25.31 | neoalex | a gxp-2000 connected to my asterisk, it works fine accept for one thing, when receiving calls from one of my providers I can't here anything |
17:25.38 | [TK]D-Fender | neoalex: And you wonder WHY? |
17:25.40 | neoalex | even though it works fine with all others |
17:26.13 | neoalex | I know I know... bought it dirt cheap before I got a chance to ask if it's any good |
17:26.26 | *** join/#asterisk ixela (i=ixela@nat/digium/x-f665871c0ee1aaf5) |
17:27.29 | neoalex | so... any ideas... also if I receive the call from that provider on any other extension (softphone, ATA) it works fine |
17:27.57 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
17:29.20 | [TK]D-Fender | neoalex: Then it would clearly sound like a newtowork issue, and I would bet a NAT one as well. |
17:29.26 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
17:30.11 | neoalex | the phone is connected to the asterisk on the same network, local IP, no nat, just like all other clients |
17:30.44 | neoalex | and like I said, there's no problem receiving calls from any other provider as well on the grandstream |
17:30.57 | neoalex | so it's a pretty localized problem, which is what makes it weird |
17:31.55 | drzed | little ISDN question: is it correct that only for using a card as NT a HFC-S is necessary? |
17:32.17 | [TK]D-Fender | neoalex: This is the point where I tell you that naturally I don't trust your configs for one second and you should have had them pastebin'd up for use to see before you even MENTIONED what the problem is... |
17:32.18 | [TK]D-Fender | ~pb |
17:32.20 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
17:32.30 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
17:32.34 | IOscanner | I have a sip trunk and I can't get it to use the contact as the account or username. Instead it uses: Contact: <sip:s@69.18.209.xx> |
17:32.56 | [TK]D-Fender | IOscanner: Look. At. Your. DIAL. |
17:33.31 | IOscanner | This is NATED, normaly it is fine, but when I nat it I loose the Contact info |
17:33.54 | neoalex | [TK]D-Fender: uoook |
17:34.01 | neoalex | just a sec |
17:34.17 | IOscanner | I am just trying to registar not even call yet |
17:34.41 | [TK]D-Fender | God helps those who help themselves. *I* am considerably less forgiving ;) |
17:34.58 | [TK]D-Fender | IOscanner: Then look at your REGISTER. |
17:35.41 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
17:35.47 | blitzrage | [TK]D-Fender: I DON'T WANT TO KNOW YOUR NAME... I JUST WANT... |
17:35.54 | [TK]D-Fender | blitzrage: ! ! ! |
17:35.58 | IOscanner | The registar string has the account number, but it is not using it as the contact when asterisk is natted and using externip |
17:36.40 | blitzrage | IOscanner: correct... Jared Smith and I found this at Geek Week in KC about a year ago... when you register, it'll try to register with 's'... |
17:36.54 | blitzrage | let me see how I got around that in TFOT2 |
17:36.58 | IOscanner | correct |
17:37.01 | IOscanner | please... |
17:37.09 | IOscanner | I have to cases of this and it is driving me crazy |
17:37.10 | IOscanner | lol |
17:37.22 | blitzrage | ya, you have to do something kinda funky to register 2 asterisk boxes together |
17:37.36 | [TK]D-Fender | IOscanner: Stop talking about it and PASTEBIN <----------------- |
17:37.49 | [TK]D-Fender | blitzrage: Hardly :) |
17:37.56 | IOscanner | paste what: Contact: <sip:s@69.18.209.XX> |
17:37.59 | IOscanner | that is the problem |
17:38.19 | [TK]D-Fender | IOscanner: No, that is a sign of the SYMPTOM. Your CONFIG is the problem (duh) |
17:38.25 | IOscanner | blitzrage seems to knwo what is wrong |
17:38.50 | [TK]D-Fender | IOscanner: And I'm quite sure as well, I just want the incriminating evidence :) |
17:40.09 | neoalex | [TK]D-Fender: http://pastebin.ca/595788 |
17:40.33 | *** join/#asterisk basty (n=basty@dome-city-rockers.sunblast.de) |
17:40.36 | blitzrage | IOscanner: http://www.pastebin.ca/595790 |
17:40.39 | basty | Hi |
17:41.01 | blitzrage | IOscanner: fix it quickly -- that post will expire in 30 mins :) |
17:41.34 | basty | I have around 115 SIP-Phones connected to my Asterisk PBX..now if I do a "sip show channels" I usually see 30-40 - is that normal? |
17:41.43 | [TK]D-Fender | IOscanner: Paster your damn REGISTER LINE. Do I have to spell it out? |
17:42.09 | blitzrage | basty: depends how many NOTIFY's or OPTION's your Asterisk is doing -- you won't see channels just because a phone is registered |
17:42.09 | [TK]D-Fender | basty: You don't have channels if people aren't on calls. |
17:42.40 | *** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com) |
17:43.00 | basty | Mh, right now nobody is calling and i still see about 33 active SIP-channels... |
17:43.06 | [TK]D-Fender | IOscanner: Thats like pointing to the blood dripping on the floor as being the problem, and not the 14" craving kine that was plunged into your chest. |
17:43.15 | [TK]D-Fender | carving knife* |
17:43.33 | blitzrage | basty: what are those channels actually doing, and what version of Asterisk are you running? |
17:43.45 | IOscanner | just a sec trying what blitzrage sent |
17:43.54 | [TK]D-Fender | basty: Pastebin it so we can see what it MEANS. |
17:44.15 | basty | for example: 10.46.3.80 (None) 3c26700f704 00101/21794 unkn No Rx: REGISTER |
17:44.23 | [TK]D-Fender | IOscanner: Pasting that single line would have taken 1/2 second and look what we are going through for this... |
17:44.34 | [TK]D-Fender | basty: those linger for a little bit |
17:44.37 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
17:44.59 | neoalex | meanwhile, I'm waiting for [TK]D-Fender to get to me too :)) |
17:45.02 | [TK]D-Fender | basty: then disappear. You can try to reduce your register interval for that if you wish, but those aren't "calls" |
17:45.08 | *** join/#asterisk THX2000 (n=bob@netblock-208-127-94-59.dslextreme.com) |
17:45.23 | [TK]D-Fender | neoalex: canreinvite=no all around ([general] and phones, peers, EVERYWHERE) |
17:45.36 | IOscanner | register=2056000055:password@registrar.carrier.com |
17:45.41 | *** join/#asterisk CoffeeIV (i=rgr@rrcs-71-42-183-82.sw.biz.rr.com) |
17:45.47 | [TK]D-Fender | neoalex: and "alexgs" is local to your * box? |
17:45.55 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
17:45.57 | neoalex | no... it's the phone |
17:45.58 | [TK]D-Fender | io thats the problem, you didn't read the INSTRUCTIONS. |
17:46.03 | [TK]D-Fender | IOscanner: thats the problem, you didn't read the INSTRUCTIONS. |
17:46.10 | [TK]D-Fender | ; Format for the register statement is: register => user[:secret[:authuser]]@host[:port][/extension] |
17:46.23 | IOscanner | I typed it it is => |
17:46.29 | [TK]D-Fender | IOscanner: You didn't fill in the /1235532whevere , so * fills in "s" FOR YOU |
17:46.35 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
17:46.49 | [TK]D-Fender | IOscanner: No /extension <---------- |
17:46.50 | IOscanner | That is what I am trying |
17:46.57 | basty | Fender: ah okay - it just confuzed me because on voip-info it says "list all registed sip accounts". For right now I am trying to debug my sip stack, because after upgrading from 1.2.13 to 1.2.19, calls hangup during the conversation with aximum retries exceeded on transmission 52b24d6471faa4596be890cf624ec3ec@10.46.0.1 for seqno 104 |
17:46.59 | [TK]D-Fender | IOscanner: thatsw what your line doesn't HAVE |
17:47.01 | IOscanner | blitzrage sent that with it |
17:47.25 | IOscanner | no I didn't have the /.... at the end adding it now |
17:47.27 | blitzrage | IOscanner: try again |
17:47.36 | *** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net) |
17:47.40 | blitzrage | IOscanner: I did send it with a trailing /osaka |
17:47.52 | blitzrage | # |
17:47.52 | blitzrage | We have a pair of Asterisk boxes we're going to call Toronto and Osaka that we're going |
17:47.53 | blitzrage | # |
17:47.53 | blitzrage | to have register to each other. We're going to use the most basic sip.conf file that will |
17:47.53 | blitzrage | # |
17:47.53 | blitzrage | work in this scenario. Just like the SIP phone configuration earlier in this chapter, its |
17:47.55 | blitzrage | # |
17:47.57 | blitzrage | not necessarily the best way to do it, but it'll work. |
17:47.59 | blitzrage | # |
17:48.01 | blitzrage | Here is the configuration for the Toronto box: |
17:48.03 | blitzrage | # |
17:48.05 | blitzrage | <PROTECTED> |
17:48.09 | blitzrage | # |
17:48.11 | blitzrage | <PROTECTED> |
17:48.13 | blitzrage | oops :) |
17:48.14 | *** mode/#asterisk [+b %blitzrage!*@*] by Corydon76-work |
17:48.19 | IOscanner | correct I would then add the /accountnumber it should use that instead of s |
17:48.22 | *** mode/#asterisk [-b %blitzrage!*@*] by Corydon76-work |
17:48.44 | *** mode/#asterisk [-o blitzrage] by russellb |
17:48.52 | russellb | :-p |
17:49.01 | IOscanner | okay things now it is a firewall issue upstream |
17:49.02 | blitzrage | :) |
17:49.02 | IOscanner | thanks |
17:49.07 | basty | [TK]D-Fender: you have any idea to the "Maximum retries exceeded on transmission" Problem? |
17:49.17 | neoalex | [TK]D-Fender: I put canreinvite=no in general (the phone already had it), still no go |
17:49.17 | *** mode/#asterisk [+o blitzrage] by russellb |
17:49.36 | blitzrage | perfect... that was the last peice I needed in order to take over the world |
17:50.13 | blitzrage | man there are a lot of spiders outside my condo windows |
17:50.45 | *** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com) |
17:51.35 | *** join/#asterisk LoveHatePassion (n=Nwm@office.xanter.net) |
17:51.37 | LoveHatePassion | Hello |
17:52.02 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
17:52.14 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
17:52.15 | LoveHatePassion | anyone integrated skype to their asterisk pbx? |
17:54.09 | blitzrage | LoveHatePassion: some people have.... check google for how to do it (you can only do 1 channel, and you have to run some weird interface thing) |
17:55.24 | LoveHatePassion | as far as I know there are only 3 gateways that do skype to sip |
17:55.29 | matdon | can anyone point me at a simple example of a fast agi script wirtten in perl? |
17:55.33 | rob0 | What are you going to do with the world now that you're in charge of it? |
17:55.36 | [TK]D-Fender | IOscanner: Just time just paste it, k? :) |
17:56.14 | LoveHatePassion | but not sure which one to use so I wanted some advice |
17:58.55 | [TK]D-Fender | LoveHatePassion: What do you actually want Skype for? |
17:59.47 | *** join/#asterisk gardo (n=gardo@121.97.211.162) |
18:03.51 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-75-84-232-194.socal.res.rr.com) |
18:05.47 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
18:07.26 | kannan | matdon -> vicidial has a call logging script in perl |
18:07.31 | kannan | eflo.net |
18:07.54 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.20, 1.4.6 (June 29, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
18:11.20 | [TK]D-Fender | russellb: annnnnddd WHEE! |
18:11.36 | russellb | gettin' them bug fixes out there ... |
18:11.56 | russellb | 90-something changes to 1.4 in 2 weeks, 40-something to 1.2 |
18:12.16 | Strom_M | i should upgrade |
18:14.17 | LoveHatePassion | [TK]D-Fender I want to use skype for international calls |
18:14.40 | [TK]D-Fender | LoveHatePassion: Why not use a normal ITSP to terminate? |
18:14.47 | Hmmhesays | bah this sucks, it seems that channel variables are destroyed when asterisk redirects |
18:15.02 | Strom_M | because the normal ITSP isn't "cool" like skype is? |
18:15.10 | [TK]D-Fender | Hmmhesays: They should become INHERITED. |
18:15.14 | Hmmhesays | or when asterisk receives a redirect |
18:15.17 | Hmmhesays | but they aren't |
18:15.35 | Hmmhesays | I'm sending a call out to an ata, the ata sends a "302" back |
18:15.41 | Hmmhesays | asterisk redirects to a local channel |
18:15.46 | LoveHatePassion | well because Skype is free |
18:15.47 | [TK]D-Fender | Strom_M: Skype is the bastard child of telephony. |
18:15.54 | LoveHatePassion | ITSP does cost |
18:15.56 | Strom_M | [TK]D-Fender: i'm well aware of this |
18:15.59 | Hmmhesays | then my channel variables are gone |
18:16.01 | [TK]D-Fender | LoveHatePassion: Free to cal the PSTN everywhere? |
18:16.08 | Strom_M | hence why i put cool in quotes :D |
18:16.19 | LoveHatePassion | [TK]D-Fender: no |
18:17.04 | LoveHatePassion | but skype gave me free credits that would probably let me talk a very long time |
18:17.10 | *** join/#asterisk gardo (n=gardo@121.97.211.162) |
18:17.25 | BSD_Tech | ok I am back still have the issue |
18:17.43 | BSD_Tech | when I hit 3 it ask for the exten number |
18:17.56 | BSD_Tech | 3 /# |
18:18.04 | BSD_Tech | I dont have # defined |
18:18.14 | [TK]D-Fender | BSD_Tech: PASTEBIN <-------------- |
18:18.52 | [TK]D-Fender | LoveHatePassion: probably? You mean you're looking at this much pain and don't even know how long the ride is going to last? |
18:19.22 | BSD_Tech | http://www.pastebin.ca/595892 |
18:19.32 | BSD_Tech | thats what happens when I hit pound |
18:19.37 | LoveHatePassion | [TK]D-Fender |
18:19.53 | LoveHatePassion | the thing what I want to do is actually like this |
18:20.43 | BSD_Tech | and I do not have a exten =# defined anywhere |
18:20.43 | Hmmhesays | so this blows |
18:20.54 | [TK]D-Fender | BSD_Tech: pastebin your actual dialplan |
18:20.55 | LoveHatePassion | I will be at the other side of the world for a long time at least 10 months. and I want to call my home and my home country so I have a TDM11P attached to my home telephony system where I can dial my home from a softphone SIP |
18:21.19 | LoveHatePassion | also can call outside of my home using my existing phoneline. |
18:21.23 | Hmmhesays | so how the hell am I supposed to authenticate a redirect call |
18:21.35 | [TK]D-Fender | Hmmhesays: Whats to auth? |
18:21.38 | LoveHatePassion | but I also want to call to other 3 countries |
18:22.02 | LoveHatePassion | and I though that I could do that with my free skype credits |
18:22.15 | Hmmhesays | [TK]D-Fender: when my ata redirects back to asterisk, asterisk sends to call into local/${num}@default |
18:22.19 | BSD_Tech | http://www.pastebin.ca/595895 |
18:23.08 | [TK]D-Fender | Hmmhesays: Should send it to where the ATA is pointed to |
18:23.27 | Hmmhesays | [TK]D-Fender: the ATA redirects back to asterisk |
18:23.59 | Hmmhesays | thats my problem |
18:24.05 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-49-9.w81-251.abo.wanadoo.fr) |
18:24.41 | [TK]D-Fender | BSD_Tech: that is NOT funny |
18:24.58 | BSD_Tech | ? |
18:25.38 | Hmmhesays | so if asterisk sends the call back into the default context, that means anyone sending in a sip call can terminate to my gateway |
18:26.05 | BSD_Tech | not oll of it works |
18:26.10 | BSD_Tech | I am rewriting |
18:26.15 | Strom_M | BSD_Tech: what the hell are you doing using # as the beginning of an extension name? |
18:26.24 | Strom_M | # means "I am finished dialing; put the call through" |
18:26.32 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
18:26.48 | BSD_Tech | #6 is done by default inthe asterisk gui |
18:26.54 | BSD_Tech | I did not get that |
18:26.57 | [TK]D-Fender | Hmmhesays: The ATA is allowed to forward to anything it can dial |
18:27.02 | ramindia_ | hey some one can asists me, my call disconnecting every 5min http://www.pastebin.ca/595894 |
18:27.03 | BSD_Tech | and 3 is used for directory |
18:29.24 | Strom_M | BSD_Tech: it's still a bad idea :) |
18:29.26 | BSD_Tech | and some of th #XX are because they are used toi change functions from remote phones |
18:30.02 | BSD_Tech | ok |
18:30.09 | BSD_Tech | I will make changes |
18:30.34 | BSD_Tech | but I still does not explai why when I hit just # its asking for a exten and passwd |
18:30.57 | Hmmhesays | there has to be some variable that is set when a redirect is sent |
18:31.20 | [TK]D-Fender | Hmmhesays: ${BLINDTRANSFER} |
18:31.28 | [TK]D-Fender | Hmmhesays: Quite likely |
18:31.34 | BSD_Tech | I dont have bildxfer enabled |
18:31.44 | BSD_Tech | blind even |
18:31.53 | [TK]D-Fender | BSD_Tech: Got a licences for that name change? :) |
18:32.15 | BSD_Tech | what name change |
18:32.26 | BSD_Tech | I have fatfinger this week |
18:32.39 | BSD_Tech | been working on many projects |
18:33.33 | BSD_Tech | [featuremap] |
18:33.34 | BSD_Tech | ;blindxfer => #1 ; Blind transfer (default is #) |
18:33.34 | BSD_Tech | ;disconnect => *0 ; Disconnect (default is *) |
18:33.34 | BSD_Tech | ;automon => *1 ; One Touch Record a.k.a. Touch Monitor |
18:33.34 | BSD_Tech | ;atxfer => 2*2 ; Attended transfer |
18:33.36 | BSD_Tech | ;parkcall => #72 |
18:33.39 | BSD_Tech | all disabled |
18:33.53 | [TK]D-Fender | BSD_Tech: I wasn't talking to YOU :) |
18:33.57 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
18:34.13 | rob0 | Fat fingers mean you never have to say you're saoirrty |
18:34.15 | BSD_Tech | sorry |
18:34.38 | [TK]D-Fender | rob0: In lesbian terms that'd be "well hung" ;) |
18:35.22 | *** join/#asterisk waverly360 (n=waverly@209.12.249.243) |
18:35.37 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
18:36.09 | irule | how can I control in what order are extensions interpreted? |
18:36.26 | BSD_Tech | ? |
18:36.33 | BSD_Tech | for what function |
18:37.12 | waverly360 | Does anyone know whether the operator setting in the voicemail.conf file does anything in asterisk 1.2.12? |
18:37.28 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
18:37.41 | BSD_Tech | 1.2.12 is way old and I dont think it was in it |
18:37.45 | [TK]D-Fender | waverly360: Yes, just like every OTHER version. |
18:37.47 | BSD_Tech | try updating |
18:37.52 | ramindia_ | is there any meetme/zaptel issues with asterisk 1.2.18 or SVN ? |
18:38.26 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
18:38.39 | BSD_Tech | tk any other Ideas |
18:38.53 | BSD_Tech | where the issue might be |
18:39.07 | waverly360 | [TK]D-Fender: Are you sure about that? I've tried setting the review, operator, and contexts options...it still does nothing when I hit 0 |
18:39.22 | [TK]D-Fender | BSD_Tech: that is a psychotic overincluded mess of over 1000 lines. |
18:39.38 | [TK]D-Fender | waverly360: Then clearly you are not doing it right. |
18:39.50 | BSD_Tech | ? |
18:39.58 | BSD_Tech | explain |
18:40.17 | *** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com) |
18:40.20 | ramindia_ | is the meetme stabled in asterisk 1.2.18, does any have problem with meetme rooms |
18:40.30 | irule | I dial 321, and 1 is dialed, of course it is the only one valid, so, why is it skipping 3 and 2? |
18:41.04 | Strom_M | ramindia_: meetme should be fine in 1.2.18 |
18:41.32 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
18:41.33 | ramindia_ | Strom_M: i have low voice issue, |
18:41.45 | Strom_M | what's "low voice issue"? |
18:42.21 | [TK]D-Fender | BSD_Tech: pastebin the actual context I should care about. |
18:42.27 | jkiff | Oh my God... a grand total of 2,572 lines. |
18:42.29 | ramindia_ | Strom_M: when people in meetme room, they are not able to talk each other, very low voice not able to hear each otehr |
18:42.31 | sweeper | pendatry! |
18:43.13 | BSD_Tech | brb |
18:43.28 | Strom_M | ramindia_: what kind of phones? what kind of connection into the asterisk box? |
18:43.38 | Hmmhesays | well its kind of a hack, but it works |
18:43.46 | ramindia_ | SIPTrunks |
18:43.52 | ramindia_ | x-lite softphone |
18:44.04 | Strom_M | ramindia_: i'd blame the softphone |
18:44.14 | Strom_M | what happens if you use real phones? |
18:44.26 | ramindia_ | tried PAP2, same issue |
18:44.47 | ramindia_ | but same X-lite directly calling works great |
18:44.56 | Strom_M | pastebin your extensions.conf |
18:44.59 | [TK]D-Fender | ramindia_: You'r mics are all too low. |
18:45.24 | ramindia_ | no its adjusted as per the test |
18:45.27 | BSD_Tech | bbiab going to do some clean up and then retest |
18:45.34 | ramindia_ | even increasing that is the same problem |
18:45.43 | Strom_M | ramindia_: |
18:45.44 | Strom_M | pastebin your extensions.conf |
18:45.46 | ramindia_ | when i listen the call, i can hear low voice |
18:46.46 | _VoiceMeUp_COM | Jun 29 14:46:43 ERROR[7300]: pbx.c:5939 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! |
18:46.47 | BSD_Tech | olot of the file is documentation |
18:46.49 | _VoiceMeUp_COM | another nice one |
18:46.55 | ramindia_ | let me do that |
18:49.29 | ramindia_ | let me paste here |
18:49.53 | ramindia_ | http://www.pastebin.ca/595941 |
18:50.00 | ramindia_ | here is my extension.conf |
18:50.21 | ramindia_ | some time i get very choppy voice |
18:51.23 | Strom_M | ramindia_: ugh, haven't you heard of contexts? |
18:51.34 | _VoiceMeUp_COM | vicidial messes those up |
18:51.54 | ramindia_ | yes i heard contexts . |
18:52.06 | ramindia_ | why vicidial messup ? |
18:52.23 | _VoiceMeUp_COM | jsut saying that it looks like bits and pieces of the vicidial install procedure |
18:52.34 | _VoiceMeUp_COM | but like strom_m said .. without any contexts |
18:52.43 | Strom_M | ramindia_: your dialplan is a royal mess |
18:52.55 | _VoiceMeUp_COM | so like a quick get it up and runnign , no toughts at all config |
18:53.24 | ramindia_ | Strom_M: really ? |
18:53.30 | Strom_M | yes |
18:53.36 | _VoiceMeUp_COM | prboably security wholes in there too.., i imagine one getting intoa cofnerence and exiting to a zap |
18:53.47 | _VoiceMeUp_COM | sed/wholes/holes/ |
18:53.59 | waverly360 | [TK]D-Fender: Well, I've been digging through voip-info for awhile now regarding the 0 out to voicemail problem. When I'm in the person's voicemail box, hitting 0 does nothing..I don't even get any notification that I'm hitting it in asterisk. |
18:54.05 | ramindia_ | Strom_M: any suggestions |
18:54.14 | Strom_M | ramindia_: rewrite it from scratch? |
18:54.21 | [TK]D-Fender | waverly360: You are falling into the usual trap.... |
18:54.30 | waverly360 | What are you talking about? |
18:54.36 | _VoiceMeUp_COM | the 0 trap ;) |
18:54.45 | ramindia_ | Strom_M: thats working config of vicidial |
18:54.48 | [TK]D-Fender | waverly360: the one of complaining about your problems without showing all the EVIDENCE |
18:55.02 | Strom_M | ramindia_: well then vicidial is a load of crap |
18:55.03 | _VoiceMeUp_COM | does any other dmtf get passed ? |
18:55.04 | [TK]D-Fender | waverly360: PASTEBIN <------------------------ |
18:55.22 | _VoiceMeUp_COM | waverly360 , try replacing voicemailmain by READ( and NoOp(" |
18:55.32 | _VoiceMeUp_COM | and see if you at least have dtmf recog |
18:55.36 | [TK]D-Fender | _VoiceMeUp_COM: getting COLDER. |
18:55.40 | _VoiceMeUp_COM | hehe |
18:55.55 | _VoiceMeUp_COM | exten ,o,1, |
18:55.56 | kannan | ramindia -> does not vicidial provide a ui to boost volume? |
18:56.28 | ramindia_ | kannan: yes they does, but i have dialplan issue for that |
18:56.32 | ramindia_ | i need to fix that too also |
18:56.59 | Strom_M | overlapping extensions? variable-length numbering? godawful syntax? |
18:57.05 | Strom_M | i'd toss that software pronto |
18:57.09 | waverly360 | [TK]D-Fender: It makes more sense to me to just get a few suggestions before I go taking up everyone's time and energy with a wad of config files everytime. If nothing obvious comes to mind, I like to play around a little bit more. I don't like people holding my hand everytime I have a problem. |
18:57.36 | Strom_M | waverly360: do you have an "o" extension in the context from which you execute voicemail() ? |
18:57.51 | *** join/#asterisk mardum (n=mardum@71-81-73-42.dhcp.stls.mo.charter.com) |
18:58.06 | waverly360 | Strom_M: Yes. |
18:58.12 | ramindia_ | Strom_M: u want me to use context to build my own |
18:58.28 | Strom_M | waverly360: and you're pressing 0 while listening to the user's recorded greeting, right? |
18:59.03 | waverly360 | Strom_M: Yes. I don't get any indication that asterisk is reading the '0' when I press it while watching the CLI. |
18:59.30 | Strom_M | does asterisk get DTMF at any other point during the call? |
18:59.50 | Strom_M | (i.e. can you terminate voicemail by pressing #?) |
19:00.28 | waverly360 | Strom_M: Hmm...yeah I can. |
19:00.58 | [TK]D-Fender | waverly360: Just pastebin it all. your description is not helping at all. |
19:01.04 | waverly360 | Strom_M: Actually, # is the only thing it does recognize. |
19:01.38 | mishehu | sorry guys, I've got limited browsing from this location. is it possible in zapata.conf to do `channel => 1,3,4` instead of `channel => 1-4`, as I want to be able to remove a bad line from the zap trunk group |
19:01.48 | *** join/#asterisk sharp (n=sharp@pool-72-94-91-143.phlapa.east.verizon.net) |
19:01.53 | [TK]D-Fender | mishehu: Yesw |
19:01.55 | mishehu | I can't try it yet as there's folks on the phone. |
19:02.01 | mishehu | [TK]D-Fender: I'll give it a shot. |
19:02.09 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
19:02.19 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
19:02.33 | [TK]D-Fender | mishehu: You could also make multiple groups for your lines. |
19:02.49 | mishehu | just didn't want anybody going ballistic on me... it's bad enough that all their bitching at me for all 3 days of cluecon turns out to be at&t's problem. |
19:03.04 | kombi | is a variable defined in the dialplan also available in agi? |
19:03.15 | [TK]D-Fender | kombi: Yes |
19:03.16 | kombi | hi Fender! |
19:03.32 | [TK]D-Fender | kombi: y0 |
19:03.37 | Corydon76-work | kombi: see GET VARIABLE |
19:04.28 | [TK]D-Fender | I need to get off my ass and learn AGI, Realtime, etc.... |
19:05.11 | kombi | so like: say digits get variable(BluePill) ? |
19:05.28 | mishehu | [TK]D-Fender: if you were american I'd say that getting off your ass would be very unamerican, but at least according to your dsl, you're in canada instead... |
19:05.36 | mishehu | and likely a canadian |
19:05.38 | waverly360 | Hah |
19:05.52 | [TK]D-Fender | <- Clearly Superior :D |
19:05.58 | irule | is this valid? gotoif [${CALLERID(num)} = 5XX]?thisisanextension:thisisnotanextension I want to send an error message and new dial tone to extensions and sent callers from outside to the IVR |
19:05.58 | kombi | ..BluePill being some juicy digits of course |
19:06.07 | mishehu | waverly360: hey, you took the stretch hummer back to the hotel the other night, didn't walk back |
19:06.07 | [TK]D-Fender | waverly360: Gimmeh mah PASTEBIN! :) |
19:06.19 | [TK]D-Fender | irule: NO |
19:06.25 | mishehu | [TK]D-Fender: I'll let you believe that since it pleases you. |
19:06.26 | LoveHatePassion | why dont people use pastebin ? |
19:06.34 | waverly360 | mishehu: Yeah, I know....two beers and 3 glasses of wine..why walk? :) |
19:06.37 | [TK]D-Fender | irule: Go and completely reread there part of the book on EXPRESSIONS. |
19:06.44 | mishehu | waverly360: gotta walk it off... |
19:06.46 | waverly360 | [TK]D-Fender: actually, I think I may have found my problem. |
19:06.51 | mishehu | all that beer is lots of calories |
19:07.04 | [TK]D-Fender | mishehu: Not so much really... |
19:07.08 | Corydon76-work | kombi: No, you cannot embed one AGI command in another |
19:07.10 | mishehu | how many bottles of beer is equivalent to a loaf of bread? |
19:07.17 | kombi | irule: few more params required there |
19:07.27 | waverly360 | mishehu: less thinky, more drinky |
19:07.33 | mishehu | [TK]D-Fender: ok, so tehre was a lot of p izza eating too along with that wine and beer |
19:07.38 | [TK]D-Fender | mishehu: beer does cost a lof of 'bread' if you go to bars :) |
19:07.49 | kombi | Corydon76-work: so first write into an AGI var, then use that i take it? |
19:08.01 | Corydon76-work | kombi: correct |
19:08.03 | mishehu | [TK]D-Fender: indeed. that's why I go for the hard stuff. it's more economical. more booze for the buck. |
19:08.07 | mishehu | err alcohol |
19:08.15 | waverly360 | [TK]D-Fender: I had my operator extension defined in the wrong place. I thought I tried it this way once already..but must've neglected to reload or something. |
19:08.26 | [TK]D-Fender | mishehu: I'm a Grand Marnier stright kind of guy mayself |
19:08.39 | waverly360 | I prefer dirty martini's :) |
19:08.54 | [TK]D-Fender | waverly360: "Evidence has been obscured to protect the GUILTY" :) |
19:08.59 | Corydon76-work | Yick, too much salt |
19:09.09 | irule | kombi combi is the famous alias for the VW wagon in Mexico lol http://images.google.com.mx/images?q=combi |
19:09.26 | waverly360 | [TK]D-Fender: Hah..you just want to give me shit about how horrible my dialplan looks :P |
19:10.07 | [TK]D-Fender | waverly360: No, I'd be perfectly happy to assassinate you on this flw individual of your myriad other sins ;) |
19:10.10 | mishehu | [TK]D-Fender: gin & tonic, screwdriver, vodka & cranberry, straight vodka, straight rum... I used to do tequila but that does bad things to stomach and sense of balance... |
19:10.11 | [TK]D-Fender | flaw* |
19:10.18 | irule | waverly360 your dialplan is fine and dandy ....hmmm, can I see it? |
19:10.36 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
19:10.50 | [TK]D-Fender | mishehu: I'm good on most of that. Mind you I'm not really a drinker. That list alone accounts for more than my typical monthly consumption :) |
19:10.53 | JerJer | whoohoo v1.4.6 |
19:10.55 | waverly360 | irule: Now why would you want to do that? All my dialplan consists of now (well mostly) is a call to AGI |
19:10.59 | *** join/#asterisk Tako-san (n=Tako-san@216.232.147.102) |
19:11.16 | irule | jsut out of curiosity ;) |
19:11.40 | waverly360 | Corydon76-home: Martini's don't have much salt..the olive juice might have some in it..might you be thinking of margaritas? |
19:11.44 | irule | I am so far 420 working lines :s |
19:12.07 | JerJer | mmm 420 |
19:12.14 | waverly360 | irule: Mine used to be huge... [TK]D-Fender has already berated me on my lacking use of macros :) |
19:12.18 | kombi | Corydon76-work: is namespace completely separate in dialplan and AGI? |
19:12.26 | mishehu | [TK]D-Fender: heh, that's about my yearly consumption - one of each. just toss in a few glasses of wine on top of that. |
19:12.33 | irule | with beautyful formatting and ASCII art pointing all over for quick browsing throught the sections lol |
19:12.37 | kombi | irule: I like that car, I believe it is called that in south africa too |
19:12.52 | mishehu | [TK]D-Fender: we had somebody doing nose dive & face crashes at cluecon... |
19:12.55 | Corydon76-work | kombi: you're talking about two different processes, entirely, communicating through a pipe |
19:13.00 | mishehu | somebody couldn't handle the drinky |
19:13.23 | waverly360 | [TK]D-Fender: Why didn't we see you at ClueCon..I figured you'd be all about a geek convention like that :) |
19:13.33 | kombi | Corydon76-work: ok, understood. so, yes so to speak..;) |
19:13.40 | irule | kombi yes combis rule, when I was a teenager a friend of mine had one and was always seeding it like crazy, it was hillarious |
19:13.45 | [TK]D-Fender | waverly360: Too far, too expensive and I'd gain too little :) |
19:13.53 | irule | speeding |
19:13.58 | waverly360 | [TK]D-Fender: Ah..you're already near perfect then? :P |
19:14.18 | [TK]D-Fender | waverly360: And that is a border I do not want to cross while the DHS, GWB, are still in effect.... |
19:14.22 | mishehu | [TK]D-Fender: gain too little? what's not to gain from hanging out with all of us? |
19:14.31 | irule | [TK]D-Fender just enjoy and show love :* |
19:14.33 | [TK]D-Fender | waverly360: No, I'm not in-depth ENOUGH actually. |
19:14.52 | [TK]D-Fender | waverly360: I've mastered the basics, but the cool shit is out of my league. |
19:15.13 | kombi | irule: they're actually called "bus" in germany (not very imaginative really), my brother had one and drove it through the sahara desert |
19:15.15 | [TK]D-Fender | waverly360: I'd go if it was cheaper and quicker to get to. |
19:15.16 | irule | [TK]D-Fender what is _THE_ cool shit? |
19:15.26 | waverly360 | [TK]D-Fender: I'm still a newb no doubt..I've only been dealing with VoIP stuff for about a year and a half now. |
19:15.30 | irule | kombi awesome |
19:15.33 | [TK]D-Fender | irule: It's not singular. it's a collective term. |
19:15.55 | irule | [TK]D-Fender what are _THE_ cool shit for example then? |
19:15.57 | waverly360 | [TK]D-Fender: That's why I'm always leary of pastebinning my stuff...I don't do well with such harsh scrutiny :P |
19:16.15 | [TK]D-Fender | irule: and byt aht I mean Realtime, AGI, real AMI, SCCP/MGCP, billing channel dev, etc. |
19:16.38 | [TK]D-Fender | waverly360: my justice would be swift, and nearly painless ;) |
19:16.41 | irule | waverly360 I always pastebin, I only take the positive criticism, the rest is down the toilet |
19:17.24 | waverly360 | I really love the concept of AGI. It's what we're using for our dialplan now. My biggest problem is figuring out the difference in using certain commands from an AGI script, as opposed to using them in extensions.conf |
19:17.49 | waverly360 | Like the directory...calling the directory function from my AGI script causes the directory function to do nothing. |
19:18.11 | waverly360 | I had to create a special context in extensions.conf, and dial it locally to make it work. |
19:18.35 | irule | waverly360 http://sciencehack.com/videos/view/DKivdMAgdeA#searchkeywords I always intentionally ot out of the masses in a sense of individuality |
19:19.12 | waverly360 | Now I'm having issues with the Voicemail command. if I'm calling voicemail from extensions.conf, hitting 0 takes me to the operator. If I call it from AGI, hitting 0 hangs up the call...still workin on that one :P |
19:19.22 | *** join/#asterisk nDuff (n=ccd@fw2.isgenesis.com) |
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19:19.30 | waverly360 | [TK]D-Fender: Hah. nearly painless eh |
19:19.51 | xo8ox | guys anyone knows why the hell does my cisco 7960 keeps restarting ? |
19:19.52 | [TK]D-Fender | waverly360: No pain, no gain ;) |
19:20.27 | mishehu | xo8ox: why yes, I can think of about a million reasons to why your cisco 7960 keeps restarting. |
19:20.37 | waverly360 | [TK]D-Fender: Yeah I know..but I don't like getting slack for shit that I didn't write. And even then..both of us were brand new to the industry..it's like development. you always look back at your previous code and sigh at how pathetic it was. |
19:20.42 | xo8ox | how do I stop it from restarting so I can get into the configs to see whats wrong with the settings |
19:20.51 | nDuff | I've had a curious issue since moving to a new building and upgrading to Asterisk 1.4: On outgoing calls, two overlapping progress tones are audible. What might be the cause of this? |
19:20.53 | waverly360 | Doesn't really make us bad programmers does it? |
19:21.20 | xo8ox | well my question is how do u stop it from restarting so I can get into it |
19:22.03 | BSD_Tech | ok I found the issue and I am taking it up with the gui guis |
19:22.10 | BSD_Tech | its a bug inthe gui |
19:22.20 | BSD_Tech | part of it is |
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19:23.52 | [TK]D-Fender | waverly360>[TK]D-Fender: Yeah I know..but I don't like getting slack for shit that I didn't write. And even then..both of us were brand new to the industry..it's like development. you always look back at your previous code and sigh at how pathetic it was. <- I have no intention of cutting you any slack ;) |
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19:24.13 | [TK]D-Fender | BSD_Tech: GUI has bugz z0mg noes! |
19:24.48 | waverly360 | [TK]D-Fender: Y'know...if you weren't a punk like this to everyone, I might think you were....well wait..you're just a punk aren't you? :) |
19:24.56 | BSD_Tech | when it creates a user exten in users.conf its setting context= not usercontext= |
19:25.04 | BSD_Tech | so the dial fails |
19:25.13 | [TK]D-Fender | waverly360: Shup junior :) |
19:25.34 | waverly360 | [TK]D-Fender: I'm not young enough for you to call me junior ;) |
19:25.45 | [TK]D-Fender | waverly360: All you have to be is YOUNGER ;) |
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19:26.13 | waverly360 | [TK]D-Fender: You don't really have any basis for my age. I might have gray hair :) |
19:26.46 | [TK]D-Fender | waverly360: I do, but I have professional help in lying about it ;) |
19:27.03 | waverly360 | hah hah hah |
19:27.08 | waverly360 | excellent |
19:27.16 | kannan | ramindia_ , i cannot understand the dialplan issue you had mentioned |
19:27.42 | [TK]D-Fender | waverly360: NEVER think that you can do a decent job dying your own hair. |
19:28.03 | waverly360 | [TK]D-Fender: LMAO something tells me there's a great story there |
19:28.08 | mishehu | [TK]D-Fender: hahaha |
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19:28.21 | *** mode/#asterisk [+o mog] by ChanServ |
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19:28.49 | [TK]D-Fender | waverly360: Nothing "funny", just you miss stuff, and those box jobs rinse out at some point (they lie about that part. |
19:28.59 | *** part/#asterisk galeras (n=root@200.31.204.42) |
19:29.26 | [TK]D-Fender | waverly360: 2 weeks ago I deceded to go and have it done by a pro. Worth every penny, and it wasn't even that much. |
19:30.06 | [TK]D-Fender | I'v gotten comments on it from everyone so far... |
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19:30.23 | waverly360 | [TK]D-Fender: I've had mine highlighted before..but I won't be doing that stuff by myself. The most I've done is shave my head..that's pretty easy to do..and difficult to mess up. |
19:31.03 | [TK]D-Fender | waverly360: Highlights? Dear God that is so not on the lit of approved things to do unless you've been spending too much time with BKW ;) |
19:31.08 | [TK]D-Fender | list* |
19:31.37 | waverly360 | [TK]D-Fender: Hah. Perhaps :) |
19:31.46 | [TK]D-Fender | waverly360: TMI <- |
19:31.58 | Qwell[] | [TK]D-Fender: Why are you dying your hair to begin with? O.o |
19:32.00 | waverly360 | [TK]D-Fender: lmao. No worries...I'm about as straight as can be. |
19:32.39 | waverly360 | Qwell[]: He's workin his mojo for the ladies. |
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19:32.51 | ramindia_ | kannan: the dial plan is for working |
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19:34.50 | kannan | ramindia_ -> i wanted to ask whats the connection between the dialplan and the volume control in vicidial |
19:35.21 | ramindia_ | kannan: volume increase with dial plan only |
19:35.36 | ramindia_ | vicidial use app_meetme for the conference calls |
19:35.58 | ramindia_ | so vicidial have tool which increase the volume in conference |
19:36.55 | BSD_Tech | ok now I knwo there is a issue and its not my dial plan |
19:37.12 | BSD_Tech | I rm my dial plan and put the default asterisk dial plan back |
19:37.22 | kannan | BSD_Tech -> but what about ur featuremap? lol |
19:37.33 | [TK]D-Fender | Qwell[]: You missed the very clear statement that I HAVE grey hair. |
19:37.35 | BSD_Tech | exerything disabled |
19:37.41 | Qwell[] | [TK]D-Fender: and you're...how old? |
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19:38.02 | BSD_Tech | and yes I restaryed asterisk |
19:38.10 | [TK]D-Fender | Qwell[]: 32, but its genetic. But in the long run I still get to KEEP it unlike some people I know :) |
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19:38.23 | BSD_Tech | and yet its still playing please enter the exten themn press pound |
19:38.38 | BSD_Tech | when I hit just the # key |
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19:40.29 | waverly360 | [TK]D-Fender: Jesus man..you're not even that much older than I am...wtf |
19:40.29 | waverly360 | [TK]D-Fender: There's a 10 year requirement to call someone Junior and get away with it :) |
19:41.19 | kannan | ramindia_ -> did u try the s option strin in meetme? |
19:41.24 | kannan | string |
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19:41.36 | [TK]D-Fender | waverly360: .... shup Junior ;) |
19:41.49 | ramindia_ | |s u mean |
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19:43.21 | BSD_Tech | http://pastebin.ca/596038 this is all thats in the current dial plan |
19:43.33 | BSD_Tech | and the # key issue still happens |
19:43.51 | kannan | yes |
19:44.37 | waverly360 | [TK]D-Fender: hush old man :P |
19:44.49 | BSD_Tech | so in some other file the # key is set |
19:45.01 | BSD_Tech | kids goto seperate corners |
19:45.04 | BSD_Tech | <=== 39 |
19:45.15 | kannan | BSD_Tech -> it is features.conf i guess |
19:45.20 | kannan | 35 here |
19:45.43 | BSD_Tech | everythingg in features.conf is disabled |
19:46.18 | BSD_Tech | and this only started with 1.4.5 |
19:46.27 | BSD_Tech | did not do this in 1.4.4 |
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19:46.37 | BSD_Tech | so I think some one hardcoded something |
19:48.01 | BSD_Tech | its pissing me off |
19:48.09 | msetim | Hi guys |
19:48.27 | msetim | somebody already make a pre-provisioning with Linksys PAP2-NA |
19:48.54 | *** join/#asterisk gardo (n=gardo@121.97.211.162) |
19:49.01 | msetim | What is the parameter that I need to put in DHCP server to say to PAP2-NA the TFTP server |
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19:49.56 | [TK]D-Fender | msetim: Option 66 |
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19:51.29 | msetim | [TK]D-Fender: thanks TK but i already put it "option option-66 172.16.0.75;" but it doesn't work |
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19:52.50 | [TK]D-Fender | msetim: Well thats the #, now if you're talking dhcpd.conf SPECIFICALLY, then its "option tftp-server-name "ip-or-host"; |
19:53.25 | [TK]D-Fender | msetim: Next time be specific ;) |
19:54.14 | msetim | [TK]D-Fender: :-D O.k. Tks ;) |
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19:54.25 | diclophis-work | hello all |
19:56.12 | diclophis-work | has anyone ever worked with a F5-BigIP device? |
19:56.17 | diclophis-work | for load-balancing SIP? |
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20:07.23 | errr | if I am in the voicemail is there a way to hit some key to get back to the operator? |
20:07.26 | errr | Id like to hit 0 and have it take me back to the operator |
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20:08.27 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-33.dsl.irvnca.pacbell.net) |
20:08.58 | *** join/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
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20:09.51 | [TK]D-Fender | err : Yes, go read the instructions for VoiceMail and brush up on your Standard Extensions. |
20:09.59 | *** part/#asterisk ramindia_ (n=ramindia@202.63.96.9) |
20:10.19 | errr | [TK]D-Fender: ok thanks |
20:12.09 | claudiotainen | is there anyone who could give me some help with my first asterisk configuration ? |
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20:14.04 | Strom_M | claudiotainen: depends on what kind of help |
20:14.44 | kannan | they are categorized as paid and unpaid help, ? |
20:14.49 | claudiotainen | well I am a complete newbie of asterisk |
20:14.55 | claudiotainen | :D |
20:15.19 | Strom_M | claudiotainen: have you read the book? |
20:15.27 | Strom_M | ~book |
20:15.27 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
20:15.45 | claudiotainen | yes that's exactly the one I was reading |
20:16.01 | Strom_M | well, ask some specific questions and we can answer them |
20:16.05 | claudiotainen | it shows how to configure X-Lite |
20:16.14 | claudiotainen | I am trying to use Ekiga |
20:16.27 | Strom_M | hopefully not on the same machine as the asterisk box |
20:16.31 | claudiotainen | I have both asterisk and ekiga installed on the same pc |
20:16.37 | Strom_M | ugh no no no no no |
20:16.41 | claudiotainen | oh ... |
20:16.41 | Strom_M | don't do that |
20:16.48 | Strom_M | don't run asterisk on any box that runs X windows |
20:17.04 | claudiotainen | ah |
20:17.20 | claudiotainen | so I sould use a computer just to run an asterisk server ? |
20:17.25 | Strom_M | yes |
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20:19.23 | claudiotainen | so as an example |
20:19.29 | claudiotainen | I could install asterisk on a pc |
20:19.48 | claudiotainen | then connect this machine to the one where ekiga is |
20:19.56 | Strom_M | yes |
20:20.08 | claudiotainen | ok thank u |
20:20.13 | claudiotainen | er one more thing |
20:20.18 | Strom_M | sure |
20:22.06 | claudiotainen | when I am registering my account with ekiga I should use asterisk address as the registrar address |
20:22.10 | claudiotainen | is that correct ? |
20:22.14 | Strom_M | yes |
20:22.21 | claudiotainen | and asterisk addres |
20:22.49 | claudiotainen | is the IP address of the machine where asterisk server is running ? |
20:23.05 | Strom_M | yes |
20:23.12 | claudiotainen | oh ok |
20:23.58 | claudiotainen | maybe that's why it didn't work when I tried to use the same pc for both the server and the front end |
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20:25.27 | claudiotainen | thank you strom |
20:25.30 | claudiotainen | bye all |
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20:26.41 | [[blah]asfd | I keep getting these errors on my server: Jun 29 14:25:19 NOTICE[4926]: chan_zap.c:8362 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
20:26.44 | [[blah]asfd | version 1.2.19 |
20:26.58 | [[blah]asfd | what would cause that |
20:27.12 | Strom_M | timing issues? irq conflict? |
20:27.20 | Strom_M | what card are you using? |
20:27.37 | shido6 | good start |
20:27.37 | [[blah]asfd | te410p |
20:27.47 | [[blah]asfd | i checked the irqs. they look fine |
20:27.55 | [[blah]asfd | timing has been a discussion on this server. |
20:27.58 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
20:28.08 | [[blah]asfd | weather it should be provided by me, or the carrier. |
20:29.04 | Strom_M | are you running anything at all besides asterisk on the box? |
20:29.04 | *** join/#asterisk sabakas1 (n=solapus@66.90.121.129) |
20:29.04 | [[blah]asfd | only asterisk |
20:29.12 | Strom_M | no x windows or anything, right? |
20:29.50 | Jingles | what's wrong with running xwindows on an * box? |
20:29.57 | Strom_M | Jingles: timing issues |
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20:30.56 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
20:32.00 | [[blah]asfd | ok, here is my zapata.conf. have I made any mistakes there? I have 3 t1s in the card. the first has the dchan and the other two have none. they are in a trunk group. |
20:32.09 | [[blah]asfd | this has worked for a while... just now become an issue. |
20:32.12 | [[blah]asfd | http://pastebin.ca/596159 |
20:33.28 | Strom_M | why are you defining spanmaps? |
20:33.28 | Strom_M | NFAS? |
20:34.04 | Strom_M | also show me zaptel.conf |
20:34.32 | [[blah]asfd | yes, it is nfas |
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20:35.44 | [[blah]asfd | http://pastebin.ca/596170 |
20:35.48 | [[blah]asfd | there is my zaptel |
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20:37.53 | Strom_M | so you're expecting timing on all four spans |
20:37.53 | Strom_M | and the telco wants you to provide timing? |
20:38.32 | [[blah]asfd | i have it set to expect timing dont i? |
20:39.55 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
20:40.11 | [[blah]asfd | what is happening is i reboot my server and then it runs good for about 10 minutes. Then i start to see hdlc errors and then everything stops working all to gether |
20:40.36 | [[blah]asfd | i get errors like: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
20:40.38 | [[blah]asfd | when i dial outbound |
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20:40.50 | [[blah]asfd | if i reboot that all goes away for about another 5-10 minutes |
20:40.56 | [[blah]asfd | timing you think? |
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20:43.09 | Strom_M | [[blah]asfd: try setting it to not expect timing |
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20:46.30 | [[blah]asfd | Strom_M: ok, so would this be correct? |
20:46.34 | [[blah]asfd | http://pastebin.ca/596205 |
20:47.12 | Strom_M | no |
20:47.56 | [[blah]asfd | ok... so I am not sure how that works then. I missunderstood |
20:48.43 | Strom_M | span = span_number,timing,buildout,framing,coding |
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20:48.59 | Strom_M | so span number is self explanatory |
20:49.05 | [[blah]asfd | so i have the wrong numbers in timing... |
20:49.07 | Strom_M | timing should be 0 if you're not looking for timing |
20:49.18 | [[blah]asfd | i have 1,2,3,4 |
20:49.27 | [[blah]asfd | i should have either 0,0,0,0 or 1,1,1,1 |
20:49.28 | [[blah]asfd | right? |
20:50.07 | Strom_M | buildout should be 0 if your telco's CSU/DSU is less than 133 feet away |
20:50.09 | Strom_M | 0,0,0,0 |
20:50.15 | Strom_M | 1 means "primary timing source" |
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20:50.20 | Strom_M | 2 means "secondary timing source" |
20:50.21 | Strom_M | etc |
20:50.49 | [[blah]asfd | uhhh... ok. if the telco is the timing, then i want it to be what? |
20:50.58 | Strom_M | 1,2,3,4 |
20:51.15 | Strom_M | assuming you want to use span 1 as the primary timing source, et. al. |
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20:51.24 | [[blah]asfd | and if i am providing it then each one would be 0? |
20:51.29 | Strom_M | yes |
20:51.34 | koosroos | hi there |
20:51.45 | koosroos | am i alone in this asterisk jungle? |
20:51.50 | [[blah]asfd | ok |
20:51.53 | [[blah]asfd | let me try that. |
20:52.00 | Strom_M | koosroos: completely, utterly alone |
20:52.22 | koosroos | lol snom_m nice device name |
20:52.25 | shido6 | :) |
20:52.26 | Strom_M | snom? |
20:52.28 | Strom_M | it's strom |
20:52.29 | Strom_M | not snom |
20:52.32 | Strom_M | please read carefully |
20:52.46 | koosroos | i know just sounds familiar |
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20:53.03 | Strom_M | i picked the handle long before i knew about Snom |
20:53.23 | koosroos | k regular on asterisk? |
20:53.34 | Strom_M | just ask your question already |
20:53.49 | koosroos | i have all the answers |
20:54.16 | koosroos | i wish |
20:54.45 | koosroos | any asterisk and fax guru's around? |
20:55.50 | Strom_M | .... |
20:55.55 | Strom_M | ask your question |
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20:56.22 | aptura | Has there been a way to make the ip500s work behind nat? I know a discussion long ago saying that it cannot work with the soundpoint series. |
20:56.40 | Strom_M | aptura: my polycom works fine behind nat |
20:56.45 | aptura | ohh |
20:56.46 | nDuff | koosroos: like Strom_C said, ask your question. |
20:57.06 | nDuff | koosroos: some of us know faxing, but we won't volunteer ourselves until we know what we're volunteering ourselves for. |
20:57.24 | *** join/#asterisk sunsmasher (n=Beamer@rrcs-24-73-230-90.se.biz.rr.com) |
20:58.00 | koosroos | downloaded the bash script fax2mail but it does not work, and i don't know didlie about scripts, anyone know where i can find a bash script for just mailing the tiff file received with app_rxfax? |
20:58.07 | Strom_M | Qwell Qwell[] russellb Corydon76-home file can you please kick sunsmasher until he fixes his connection? |
20:58.25 | *** mode/#asterisk [+b *!*n=Beamer@*.se.biz.rr.com] by Qwell[] |
20:58.26 | *** kick/#asterisk [sunsmasher!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (Please fix your connection.) |
20:58.32 | nDuff | koosroos: not me -- I use HylaFAX+iaxmodem, not app_rxfax. |
20:58.40 | Strom_M | thanks Qwell[] :) |
20:58.54 | *** join/#asterisk obnauticus (n=obnautic@c-71-59-162-60.hsd1.wa.comcast.net) |
20:58.59 | koosroos | nDuff: how do you intergrate halafax into asterisk? |
20:59.03 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
20:59.11 | nDuff | koosroos: iaxmodem. |
20:59.28 | aptura | okay well then I guess I need to configure it so it can work. I am in the network part of the configuration and under Network Address Translation have the ip/signaling/media port setup. But wonder if both are 5060. Have RTP port setup with the range of 10k-20k but do not know what the forced port field is for. |
20:59.29 | koosroos | do you only do ip faxing? |
21:00.03 | nDuff | koosroos: no -- I don't do any. |
21:00.28 | nDuff | (don't do any IP faxing, that is; T.37 and T.38 are just too sparsely implemented). |
21:00.37 | aptura | My guess is with these fields not complete it cannot register and so the display on the ip500 is flashing. |
21:00.59 | Strom_M | aptura: how are you configuring the phone? |
21:01.25 | koosroos | nDuff: I can receive faxes no problem, i just have a problem mailing it, it works then it just stops, but I still receive the incoming faxes in var/spool/asterisk/fax/ folder |
21:01.33 | aptura | Right now though its web page. But done it in xml and on the phone in the past. |
21:01.44 | Strom_M | aptura: do it in the xml file |
21:01.48 | aptura | k |
21:01.50 | *** join/#asterisk sunsmasher (n=Beamer@66.251.47.154) |
21:01.56 | Strom_M | what version of the firmware are you using? |
21:02.03 | Strom_M | oh god, he's back |
21:02.34 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
21:03.22 | aptura | Strom looking up version. |
21:03.43 | aptura | 1.6.2.0041 |
21:03.48 | Strom_M | Qwell[]: it's happening again! halp!!! |
21:04.05 | Strom_M | aptura: that's an extremely old version |
21:04.07 | Qwell[] | ... |
21:04.09 | aptura | I know |
21:04.13 | *** mode/#asterisk [+b sunsmasher!*@*] by Qwell[] |
21:04.34 | *** mode/#asterisk [-b *!*n=Beamer@*.se.biz.rr.com] by Qwell[] |
21:04.58 | nDuff | koosroos: hrm. I don't trust app_rxfax -- it doesn't have the robustness of HylaFAX's Class 1 implementation -- so I don't use it, so I don't know anything about its hook scripts... and since I'm not ever going to use app_rxfax for work, I'm not too inclined to learn how its hook scripts work without someone paying me to do that. Sorry. |
21:05.02 | Strom_M | aptura: who is your reseller? |
21:05.04 | aptura | Not a licenced reseller so do not have access |
21:05.10 | aptura | to polycoms |
21:05.19 | aptura | attacom |
21:05.30 | aptura | polycoms web site. |
21:05.32 | Strom_M | they should be able to provide you with updated firmware |
21:05.38 | aptura | k |
21:06.12 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
21:06.12 | aptura | In the mean time for now want to see if I can make this phone register by passing out of my public ip and back in to my network. |
21:07.32 | mcab | aptura: what do you mean by "the display is flashing"? is it the entire display, or just the time? |
21:07.55 | aptura | sorry the time is flashing and the lines are not registered. |
21:08.32 | mcab | aptura: well, the time is flashing because the phone can't contact an SNTP server; this is probably a seperate issue from the registration problem :-) |
21:08.41 | aptura | I figure that is the case with a incomplete NAT fields setup in the phone. Phone works fine so wanted it to register though the public ip instead of internally. |
21:09.06 | mountainm2k | I'm trying to make a menu that executes some system commands (it momentarily unlocks various doors in the office in case somebody locks themselves out -- it works from the command line) |
21:09.21 | mountainm2k | http://www.pastebin.ca/596272 |
21:09.23 | aptura | mcab it works fine. I just made a minor change by adding nat info so while its incomplete it will not access the tftp server. |
21:09.49 | Strom_M | aptura: FWIW, my polycom uses the default files included with 2.1.2, and the only things i've added are the registration details and the SNTP server. works fine behind NAT. |
21:10.05 | aptura | okay |
21:10.19 | aptura | so this is unessesary fiels then. |
21:11.05 | aptura | will go back to change the servers ip to external. fw is configured |
21:12.30 | aptura | also anyone know of a west coast ntpd time server I can use? |
21:12.41 | Strom_M | tick.ucla.edu |
21:12.51 | Strom_M | is that west-coast enough? |
21:12.54 | *** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
21:12.57 | aptura | sure it |
21:13.01 | aptura | is :) |
21:13.06 | Strom_M | are you in los angeles? |
21:13.44 | aptura | bc |
21:14.10 | Strom_M | perhaps you'd better find an ntpd in BC then |
21:14.17 | Strom_M | los angeles is a bit of a schlep away |
21:14.18 | aptura | we are pacific time |
21:14.22 | [[blah]asfd | Strom_M: That did not fix my problem. I switch my timing to 0 and i still get the error |
21:14.25 | [[blah]asfd | what else can I try? |
21:14.27 | aptura | it does not matter |
21:14.28 | aptura | ;) |
21:15.31 | shido6 | :) |
21:16.24 | *** join/#asterisk boster (n=boster@38.98.147.68) |
21:16.56 | aptura | mmm just restarted ntpd and still says EDT |
21:17.10 | boster | is there a way to compile ztdummy for a newer (2.6.18) kernel with SMP and have it not lose clock seconds? |
21:17.16 | mountainm2k | are you saying you're in the incorrect time zone? |
21:17.20 | mountainm2k | or that your clock is wrong? |
21:17.24 | shido6 | use date |
21:17.29 | boster | I can compile it, and it loads, but conferences get progressively laggy |
21:18.56 | aptura | shido same thing |
21:19.04 | shido6 | erf? |
21:19.10 | shido6 | what did you put |
21:19.11 | aptura | Fri Jun 29 17:18:48 EDT 2007 |
21:19.18 | shido6 | oh |
21:19.19 | shido6 | zoneinfo.... |
21:19.21 | shido6 | let me think |
21:20.32 | shido6 | goog /etc/localtime |
21:21.27 | shido6 | http://www.linuxsa.org.au/tips/time.html |
21:23.10 | [[blah]asfd | ok, so i am loosing my zap channels after about 5 minutes of server uptime... regarding timing and interrupts, here are my files http://pastebin.ca/596308 |
21:23.21 | [[blah]asfd | could anyone possibly see where I could be getting this issue from? |
21:23.43 | aptura | done |
21:23.58 | Strom_M | [[blah]asfd: why not call digium support? |
21:24.07 | [[blah]asfd | hmmm good idea |
21:30.19 | aptura | interesting. VM light is not lighting up. |
21:31.07 | aptura | okay this is a little odd. its happened before leave a vm but the vm light does not illuminate. |
21:31.12 | Strom_M | aptura: did you set mailbox= in sip.conf? |
21:31.32 | aptura | your right |
21:31.41 | Strom_M | my right what |
21:31.45 | Strom_M | my right....arm? |
21:32.00 | aptura | You would think that would be built right into the code but in this case its not. |
21:32.25 | Strom_M | which code? |
21:32.41 | aptura | yea its setup mailbox=200@default |
21:33.00 | Strom_M | and is there a message in 200@default? |
21:33.03 | aptura | yes |
21:33.23 | aptura | left two test messages. shows two attempted calls on the LCD |
21:35.07 | Strom_M | is the MWI setting turned on in sip.cfg? |
21:35.32 | aptura | It should be have not messed with it. Has always worked but let me verify it. |
21:36.10 | Strom_M | is the phone set to host=dynamic in sip.conf? |
21:37.02 | aptura | There is now |
21:37.28 | Strom_M | ? |
21:37.56 | aptura | I just configured it. I think it was removed when doing some editing the other night |
21:38.04 | aptura | k made a test call |
21:38.04 | *** join/#asterisk Juxt (n=Juxt@c-71-196-42-107.hsd1.fl.comcast.net) |
21:38.08 | Juxt | hello guys |
21:38.12 | aptura | no mwi light yet. |
21:38.25 | Strom_M | is the phone actually registering>? |
21:38.45 | Juxt | quick question: if i am dialing a bunch of numbers at the same time e.g. SIP/exte1@carrier1&SIP/exte2@carrier2) how can i make asterisk write separate CDR records for all the calls? |
21:38.53 | aptura | yes |
21:38.58 | aptura | 200/200 192.168.10.50 D 5060 Unmonitored |
21:39.24 | Strom_M | wait - isn't your asterisk box on a public IP? |
21:39.40 | Dj_FlyBy | if I wanted to try asterisk would it be possible to try it in a Virtual Environment like VMWare? |
21:39.44 | aptura | both phone and * are internal network |
21:39.48 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
21:39.56 | Strom_M | Dj_FlyBy: sort of |
21:40.09 | aptura | this wmi mwi error preceded my attempt to make the phone access the gw externally |
21:40.20 | koosroos | what phone are you using aptura/ |
21:40.22 | koosroos | ? |
21:40.24 | aptura | ip500 |
21:40.28 | Dj_FlyBy | Strom_C: would there be some docs I could look at for this? |
21:40.36 | Dj_FlyBy | Strom_M: would there be some docs I could look at for this? |
21:40.42 | Strom_M | Dj_FlyBy: don't spam. |
21:40.50 | koosroos | what seems to be the problem? |
21:40.52 | Dj_FlyBy | not spam, typo |
21:41.00 | Strom_M | i can read, you know |
21:41.13 | Dj_FlyBy | then why accuse me of spam if you can see the typo |
21:41.17 | Dj_FlyBy | anyways |
21:41.26 | Strom_M | because you pasted essentially the same line twice |
21:41.28 | Strom_M | that's why |
21:41.32 | aptura | well i need to get other things done. |
21:42.02 | aptura | minor nusence that the mwi is not lighting up but will work on it another time. |
21:42.07 | Dj_FlyBy | well sorry I guess, never thought I'd have to apologize for fixing a typo |
21:42.14 | Strom_M | Dj_FlyBy: anyway, running asterisk in a VM is doable |
21:42.21 | Strom_M | but i wouldn't run it in production |
21:42.37 | Strom_M | you'll also have no access to hardware |
21:45.39 | Dj_FlyBy | not looking to run it in produfirstction *yet* just want to try it out |
21:45.41 | Dj_FlyBy | ack |
21:46.03 | Strom_M | lol |
21:46.03 | Dj_FlyBy | well, you get what I mean I am sure :) |
21:46.28 | Dj_FlyBy | <-- having troubles with the touchpad on the laptop running in *nix, lol |
21:46.47 | Strom_M | heh |
21:48.08 | Dj_FlyBy | well, thanks for the info, will search out some documentation for it, if there is any out there |
21:48.10 | Dj_FlyBy | :) |
21:48.49 | Strom_M | ~thebook |
21:48.50 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
21:49.02 | Strom_M | ^^^^^^^^^^^ read that |
21:49.41 | Dj_FlyBy | yea, I've been reading it *just started* but have yet to see anything related to setting it up in a Virtual Environment, I assume though it is probably the same. |
21:51.14 | Strom_M | yeah, just for dicking around with it, you should have no trouble |
21:52.08 | Dj_FlyBy | excellent, thanks again for the help. |
21:53.59 | *** mode/#asterisk [-bb brian!*@* #asterisk!*@*] by Qwell[] |
21:55.36 | *** join/#asterisk vn (n=vn@bas5-quebec14-1177599731.dsl.bell.ca) |
22:03.15 | *** join/#asterisk Primer (n=vi@sh.nu) |
22:03.43 | Primer | Is chan_sccp deprecated in favor of something else? or is it just dead? It won't compile against 1.4 |
22:03.46 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.67) |
22:04.00 | *** join/#asterisk swagger (n=Steve@ip68-227-26-15.lv.lv.cox.net) |
22:04.11 | mvanbaak | Primer: chan_sccp is dead |
22:04.16 | Primer | bogus |
22:04.26 | Primer | there's no sip firmware for my 7920, last I checked |
22:04.28 | Strom_M | Primer: see chan_skinny |
22:04.38 | mvanbaak | chan_skinny works ok |
22:04.42 | Primer | I thought chan_skinny was even older/less maintained |
22:04.43 | Strom_M | which, incidentally, is right there when you make menuselect |
22:04.47 | mvanbaak | I use it in production on a couple of sites |
22:04.52 | Strom_M | no, qwell fixed it up |
22:04.56 | Primer | ahh cool |
22:05.14 | mvanbaak | Strom_M: actually, chan_skinny is active again |
22:05.17 | Strom_M | i'm trapped in a universe factory |
22:05.19 | mvanbaak | look at the bugtracker |
22:05.22 | Qwell[] | quite active, actually |
22:05.22 | Strom_M | cocks |
22:05.36 | Strom_M | mvanbaak: thats what i said |
22:06.12 | *** join/#asterisk Wag3Slav3 (n=gregg@71-32-119-21.bsmr.qwest.net) |
22:06.33 | Wag3Slav3 | <PROTECTED> |
22:06.44 | *** join/#asterisk GothAlice (n=amcgrego@209.161.123.42) |
22:07.19 | GothAlice | How can I allow a user on a call to enter a command, say "112", and have asterisk perform an action, without disconnecting the call? |
22:07.36 | mvanbaak | GothAlice: look at features.conf |
22:07.40 | Strom_M | GothAlice: features.conf |
22:08.02 | Strom_M | and let us not forget features.conf while we're at it |
22:08.16 | mvanbaak | ;) |
22:08.52 | GothAlice | Aaah... guess I shouldn't have deleted that file when I first set up the basic server... XD |
22:08.53 | *** join/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net) |
22:09.00 | waverly360 | Any of you guys played around with the Aastra 57i phone? |
22:09.10 | __DAW | waverly360: yup |
22:09.14 | knarfly | can I record a call once it's started? |
22:09.25 | Strom_M | and let us not forget features.conf while we're at it |
22:09.29 | Strom_M | knarfly: features.conf |
22:09.30 | mvanbaak | knarfly: yeah |
22:09.51 | mvanbaak | knarfly: look at features.conf |
22:09.51 | mvanbaak | ;) |
22:09.52 | Defraz | when a caller calls into a DID on my PRI, it seems to work great when I ring an extention, but I would like an incoming DID to go straight to a fax tone, and then convert that to a pdf when it is done receiving. Is this possible or does anyone know of some howto's on doing this. |
22:10.00 | Strom_M | i should have an array of one-touch answers for this channel |
22:10.06 | Defraz | Like when someone dials 666 for a test fax tone. |
22:10.10 | waverly360 | __DAW: Well..I have one here..and it has two 'sidecars'...one's an lcd display, and the other looks like a couple of columns of buttons that you slip paper into. |
22:10.18 | mvanbaak | Strom_M: irssi has /alias |
22:10.18 | mountainm2k | Defraz: Yes, but not directly with Asterisk |
22:10.24 | waverly360 | __DAW: can they both be attached to the phone at the same time? |
22:10.29 | Strom_M | mvanbaak: eh, i'd like buttons |
22:10.38 | Defraz | mountainm2k: Really? |
22:10.44 | mountainm2k | Defraz: I have a solution that works using Hylafax -- check out "iaxmodem", read the docs... |
22:10.45 | mvanbaak | Strom_M: you can use F keys in irssi as well |
22:10.49 | __DAW | waverly360: not really sure, ive never used the side car. As with most aastra phone, im not terribly impressed. |
22:10.55 | mountainm2k | Defraz: Yup. Each of our people have their own fax DID. |
22:11.14 | *** part/#asterisk Wag3Slav3 (n=gregg@71-32-119-21.bsmr.qwest.net) |
22:11.25 | Defraz | can you set the pri to split up the traffic for asterisk and for your solution. |
22:11.51 | mountainm2k | Defraz: You don't set the pri to do anything -- Asterisk handles the call, passes it off to an IAX endpoint |
22:11.54 | Strom_M | dec: DNIS |
22:11.55 | Strom_M | er |
22:11.57 | Strom_M | Defraz: |
22:11.57 | Defraz | I am checking out Iaxmodem right now. |
22:12.01 | mountainm2k | Defraz: which happens to be a soft-modem |
22:12.08 | mountainm2k | Defraz: (iaxmodem) |
22:12.17 | Defraz | hmmmm cool cool. |
22:12.41 | mountainm2k | Defraz: The docs for iaxmodem are pretty good -- read it, and get back with us if you can't figure it out |
22:12.46 | Defraz | Because I have been using the fax detection deal and it works when the tone is provided but sometimes you get a 15 year old fax machine. |
22:12.53 | Defraz | Okay thanks a ton |
22:13.00 | mountainm2k | Defraz: (although I don't hang out here often -- maybe I should write a wiki) |
22:13.25 | mountainm2k | Defraz: I didn't try the fax detection stuff, but I knew I wanted actual DID faxing |
22:14.13 | mountainm2k | Defraz: I actually hacked the script in hylafax to do a database lookup (in the Asterisk Realtime DB) for the user's email address associated with their voice mailbox |
22:15.21 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
22:16.02 | mountainm2k | Defraz: http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem |
22:18.47 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
22:18.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
22:19.38 | Defraz | sorry mountainm2k I was away reading. |
22:20.27 | GothAlice | Last question, I'm trying to run some commands before a prefixed call (_99.) then place the call back into the call tree to continue on its way (e.g. dial a local extension, dial out, etc). Dial() isn't working for me. And googling on voip-info.org isn't helping... |
22:20.37 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
22:20.52 | sahafeez | question - is anyone running on ubuntu-server? |
22:20.53 | *** part/#asterisk Primer (n=vi@sh.nu) |
22:21.12 | sahafeez | i am upgrading a box on sunday and i was thinking about moving from slackware |
22:21.40 | mvanbaak | sahafeez: we use ubuntu-server only on the hardware |
22:22.01 | mvanbaak | it's because vmware has prebuild kernel modules for it |
22:22.17 | mvanbaak | so it's running on the steal, only to be the host for vmware |
22:22.26 | mvanbaak | all real stuff is running in a vm |
22:22.45 | mvanbaak | and all vm's run debian |
22:22.52 | mvanbaak | that's on our linux platform |
22:22.52 | sahafeez | hum, i do not want that. i just want a single ubuntu box running asterisk with a PRI card. |
22:23.16 | mvanbaak | ah, no idea about that sorry |
22:23.25 | sahafeez | thanks |
22:24.32 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
22:28.02 | *** join/#asterisk electus (i=electus@asphaleia.SuxOS.org) |
22:30.00 | waverly360 | Is there any way that a polycom phone and say an aastra phone can detect each other's presence? |
22:30.23 | sahafeez | wow. never seen it this slow |
22:32.25 | [TK]D-Fender | waverly360, Yes, |
22:33.12 | waverly360 | [TK]D-Fender: How does that work? The polycoms are specifically built to detect each other's presence using the buddy watch stuff....how can an aastra phone interact with that? |
22:33.33 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.68) |
22:39.11 | Strom_M | sahafeez: i run asterisk on debian, which is close enough |
22:39.40 | sahafeez | thanks. i need to read about how to get the zap kernel stuff in. |
22:39.51 | Strom_M | it's easy |
22:40.11 | Strom_M | just make install zaptel, assuming you have the kernel headers installed |
22:40.38 | mvanbaak | apt-get your kernel headers |
22:40.57 | mvanbaak | and then you can follow the simple: make install |
22:42.51 | *** part/#asterisk boster (n=boster@38.98.147.68) |
22:43.31 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
22:44.18 | sahafeez | ok. cool. i normal do this on slack or gentoo so it is all by hand. not quite used to the apt stuff |
22:45.33 | Strom_M | apt is sex on a stick |
22:46.14 | sahafeez | hum. i am a ports guy but ok. |
22:50.58 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
22:59.02 | *** join/#asterisk dflow (n=pch@chaos.addr-arpa.pl) |
23:02.58 | *** part/#asterisk dflow (n=pch@chaos.addr-arpa.pl) |
23:05.45 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
23:05.45 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
23:19.17 | *** part/#asterisk Juxt (n=Juxt@c-71-196-42-107.hsd1.fl.comcast.net) |
23:30.43 | *** part/#asterisk swagger (n=Steve@ip68-227-26-15.lv.lv.cox.net) |
23:35.16 | knarfly | thanks everyone...got the wW working... |
23:35.58 | *** part/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net) |
23:42.02 | aptura | Strom ever look though sundancecom? |
23:42.12 | aptura | sundance-com that is |
23:42.37 | aptura | Telcom guys commenting on asterisk and how it is not as serios as traditional PBX's |
23:44.58 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
23:45.57 | kombi | is the path to the voicemail file accessable in some way (maybe even in a variable)? |
23:46.43 | blitzrage | aptura: typically spouted by people who don't understand it |
23:48.26 | aptura | yea :) thay are putting down hosted PBxs which I can understand. The main gripe is support when its needed or a lack of. |
23:48.39 | Innatech | well, if your definition of serious is "complicated and expensive." Sure. |
23:49.22 | Innatech | Maybe those guys want to buy the abandoned Merlin switch I tore out earlier today. Heh. |
23:49.45 | aptura | BTW my IP-500 lines are now registered to the outside GW but the time is flashing. Sounds like a ntpd port needs to be opened. |
23:50.07 | aptura | Its a very good site check it out sundance communications. |
23:50.17 | blitzrage | aptura: well, that makes no sense to me actually... hosted PBXs actually would be easier to support because now you can have a team of engineers who support multiple PBXs in a single physical location. But everyone has their own requirements. Sometimes it makes sense, sometimes not. |
23:50.21 | aptura | Alot of vetran Telcom techs are registered in the site. |
23:50.47 | blitzrage | veteran means old, and old telecom techs typically don't understand the future |
23:50.53 | aptura | :) |
23:51.00 | aptura | some are running asterisk |
23:51.04 | blitzrage | they're living in the past... I'm the new hotness |
23:51.10 | blitzrage | I mean Asterisk is the new hotness.... :P |
23:51.14 | kombi | may I repeat my sophisticated, yet highly intersting question? |
23:51.29 | blitzrage | kombi: the question is still on my screen... wait a while |
23:51.38 | Innatech | ztdummy still requires secret tricks on virtualized linux hosts to make it go. (last I checked.) Only some of the hosting companies have it working right, which doesn't help the reputation of hosted * . |
23:51.47 | *** join/#asterisk ltd (i=foobar@nox.amused.net) |
23:52.13 | kombi | where is Fender? |
23:52.27 | blitzrage | Innatech: ahh... the way I've built hosted PBXs is not to virtualize anything |
23:52.37 | Innatech | blitz: yeah, that's the safe way. |
23:52.51 | blitzrage | guess it depends on your definition of hosted PBX |
23:54.06 | aptura | and if the dsl goes down then what? tanked unless the pbx is inhouse on a t1 or pots |
23:54.21 | Innatech | yup. Dedicated boxen or managed/shared large *'s for multiple organizations both work fine. But people want VPSes, which gets tricky. |
23:54.48 | blitzrage | aptura: your phones would not be functional, but all other services of the PBX would still function because it is hosted off site |
23:55.18 | Innatech | People could call in and get the IVR, they just wouldn't be able to get to your extensions. |
23:55.27 | blitzrage | VM would be functional |
23:55.42 | blitzrage | however, if you had the PBX on site, you would not have any of those services |
23:55.54 | blitzrage | and small companies can't afford a T1 (hence why they are using a vPBX) |
23:56.02 | Innatech | I prefer to have a front and back * for those who want to most of their call handling offsite for BW savings by colocation or whatever. You use an internal Asterisk and an external hosted Asterisk. Tie em together. |
23:56.30 | aptura | Or finace the cost of a onside pbc |
23:56.31 | aptura | pbx |
23:56.33 | Innatech | Thus if your WAN link goes down you can still call internally and clients can still get IVR and VM. |
23:57.07 | Innatech | If you're relying on a beefy hosted * to do most of the heavy lifting, a very modest * will do onsite for handling the extensions. |
23:57.16 | blitzrage | Innatech: yep, I've done that too so you can have 1 or 2 analog lines used when the Internet goes down |
23:57.29 | Innatech | yeah, typically the fax/DSL lines. |
23:57.32 | blitzrage | something like a soekris box |
23:57.34 | aptura | or setup the pbx with a backup pots line for outgoing calls. I recently installed some cisco routers with DDR and thought that was interesting. |
23:57.42 | blitzrage | or a shuttle PC, or whatever |
23:57.57 | Innatech | I like commel mini-ITX boards. |
23:57.59 | aptura | blitzrage how many channels can one of those soekris boxes handle with ulaw? |
23:58.09 | blitzrage | something like 10 |
23:58.23 | aptura | okay and that may include vm so mabey 8-9 |
23:58.30 | blitzrage | still 10 |
23:58.37 | blitzrage | VM is still a channel |
23:58.45 | aptura | thats what im saying |
23:58.46 | aptura | :) |
23:58.53 | aptura | 8-9 active conversations |
23:58.54 | Innatech | The little Commels can do pretty much whatever you need. |
23:59.11 | blitzrage | well, if you're sending the call to a phone, you have 2 chanenls in use (bridged) -- with VM, it's only 1 |
23:59.24 | aptura | good enough for small offices |
23:59.47 | Innatech | yup--very nice for small offices in fact. |