00:03.14 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
00:03.41 | wishes | has anyone ever had asterisk and mysql going ,and then had asterisk somehow just drop an entire table ? |
00:03.59 | Nugget | no, but I've had mysql do that on its own. |
00:04.06 | wishes | mm really? |
00:04.10 | Nugget | yes |
00:04.20 | nDuff | mysql ~= eeeevil |
00:04.25 | Nugget | indeed it is. |
00:04.47 | wishes | mm ahh shit |
00:04.48 | wishes | nm |
00:04.51 | wishes | i just realized i did it |
00:04.57 | wishes | what a fucknut i am :/ |
00:05.01 | Nugget | I guess you're evil, too! |
00:05.15 | wishes | i pasted the create new table, accidently pasted the 'delete if exists' on the line above |
00:05.18 | wishes | i guess i am :/ |
00:06.09 | Schreiber1337 | So, does anyone understand how MWI works with SIP.. I |
00:06.45 | Schreiber1337 | I'm trying to find out if my phones are turning off the MWI light after 5 seconds or if the server is sending a clear MWI call... |
00:08.55 | Schreiber1337 | crickets chirping... |
00:09.19 | Nugget | sounds like a job for sip debug. |
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00:31.22 | mistermocha | okay, I've been having a hell of a time trying to figure this out.... |
00:31.39 | TJNII | Bah. I keep getting "ztdummy: Unable to register zaptel rtc driver" whenever I try to load ztdummy |
00:31.40 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
00:31.47 | mistermocha | I can connect to manager, but I can't figure out how to actively listen for events |
00:32.27 | mistermocha | what would be the appropriate manager command to just see call events fly by? |
00:32.37 | mistermocha | (akin to watching the CLI) |
00:33.35 | elixer | at the end of your Action: Login block |
00:33.36 | elixer | add: |
00:33.38 | elixer | Events: on |
00:34.33 | mistermocha | oooh neato... |
00:35.15 | mistermocha | I'll give that a go... thanks |
00:35.15 | elixer | yup |
00:36.31 | mistermocha | hmm... no go |
00:37.19 | mistermocha | I added Events: on and placed a call... but nothing came through |
00:37.22 | elixer | paste your manager.conf to a pastebin |
00:37.24 | elixer | ~pb |
00:37.24 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
00:37.37 | elixer | mask out your passwords, please. |
00:38.42 | rob0 | no way!! I want 'em. |
00:39.29 | mistermocha | it's a trixbox pro... manager pwd's are mostly the same |
00:39.42 | JT | this isn't a trixbox channel |
00:40.02 | mistermocha | I know... but it's still manager |
00:40.16 | mistermocha | give me just a coupla mins before giving me the boot |
00:40.29 | elixer | mistermocha: manager.conf? |
00:41.38 | mistermocha | elixer: yah... as I look at it, I think I figured it out |
00:41.53 | mistermocha | the user I'm logging in as doesn't have any read perms |
00:42.18 | elixer | right |
00:42.22 | elixer | figured as much |
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00:45.21 | mistermocha | oh snaps! that did it |
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00:53.46 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
00:54.15 | tengulre | hi,all |
00:54.22 | tengulre | good morning everyone~! |
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01:04.03 | *** mode/#asterisk [+o mog] by ChanServ |
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01:21.56 | Fetch | bleh, wtf does make install in zaptel need to overwrite zaptel.conf |
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02:00.06 | WilliamK | usb radio ALOT of issues in the svn version |
02:00.13 | Zylkron | I've a question, I have a phone connected to my router, and we register at asterisk server, our users uses g729 codecs and we want asterisk to passthru the calls to Openser , its that possible |
02:00.51 | Zylkron | Im not sure technically how would the diagram looks like really :P |
02:01.02 | Zylkron | can anyone hekp |
02:01.04 | Zylkron | pls |
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02:23.35 | TJNII | Zylkron: You want asterisk to change protocols for you? |
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02:24.32 | JT | didn't see any mention of changing protocols, TJNII |
02:24.45 | TJNII | Yea, I'm dumb |
02:24.59 | TJNII | I read it wrong. |
02:33.19 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
02:33.57 | Zylkron | TJNII: nope |
02:34.19 | Zylkron | I just want asterisk to go into passthru mode, and let openser does the signal handling |
02:37.41 | *** join/#asterisk PepOSX (n=pepOSX@190.72.145.178) |
02:41.44 | [TK]D-Fender | Zylkron, What exactly is * DOING for you in this scenario? |
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02:44.51 | riddlebox | [TK]D-Fender, is there a good howto, on setting up a TDM card, with fx0? |
02:45.49 | [TK]D-Fender | Zylkron, I'm trying to figure out why you are shoving * BETWEEN your phones & SER in the first place. |
02:46.07 | JT | Zylkron: i thought that was crazy too |
02:46.11 | JT | [TK]D-Fender: even |
02:46.17 | [TK]D-Fender | riddlebox, Plenty of guides out there for setting up zapata. |
02:51.50 | Zylkron | uhm |
02:52.05 | Zylkron | okay well |
02:52.33 | Zylkron | we use * for registration |
02:52.43 | Zylkron | according to the diagram |
02:52.57 | [TK]D-Fender | Zylkron, so far thats like inserting a 5th wheen in your car. Whats to POINT? |
02:53.05 | Zylkron | the point is |
02:53.14 | Zylkron | we dont have g729 license :P |
02:53.19 | [TK]D-Fender | Zylkron, why not just say "route all of our calls through CHINA". that'd be just as productive. |
02:53.23 | Zylkron | so we use openser to handle the signal |
02:53.37 | [TK]D-Fender | Zylkron, well if * jsut a PASSTHROUGH then it isn't adding ANYTHING <- |
02:54.18 | Zylkron | uhm |
02:54.32 | Zylkron | this is definitely *not* helping me settle my problem :P |
02:54.37 | [TK]D-Fender | Zylkron, * sitting between your phones and OpenSER in passthrough mode does NOTHING, so why put it there? |
02:54.52 | JT | sounds like your problem is an insane setup |
02:55.06 | Zylkron | * does voice mail, registrar and stuff, but it wont handle g729 |
02:55.09 | Zylkron | instead |
02:55.09 | [TK]D-Fender | Zylkron, your description really sucks. You don't have a "need" in it. |
02:55.14 | Zylkron | it will ask for g711 |
02:55.32 | JT | stop being stingey and buy g.729 licenses? |
02:55.41 | Krurst | It'll work though - * can passthrough 729 without a license |
02:55.51 | Zylkron | I dont have to, our routers have it :P |
02:55.51 | [TK]D-Fender | Krurst, back up while the getting's good... |
02:56.08 | [TK]D-Fender | Krurst, leave this to trained masochists :p |
02:56.14 | iCEBrkr | : | |
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02:56.44 | Zylkron | man =p |
02:56.48 | Zylkron | fucking useless =p |
02:56.51 | Zylkron | wtfever =p |
02:56.55 | JT | =p |
02:56.57 | Zylkron | I thought you're engineers :P |
02:57.11 | JT | engineers in sanity |
02:57.15 | *** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir) |
02:57.17 | [TK]D-Fender | Zylkron, You have not made an end to end description of WTF * is actually DOING<----------- for you. |
02:57.18 | iCEBrkr | WORTHLESS!!! |
02:57.18 | JT | to get your proposed setup working |
02:57.32 | JT | it will cost more to make that work, than to buy g.729 licenses |
02:57.55 | Zylkron | .... |
02:57.56 | [TK]D-Fender | Zylkron, "yay, register to * to to PASS THE CALL AS IS to OpenSER." Translation : stis inbetween not converting ANYTHING or DOING anything. |
02:58.24 | JT | Zylkron: they're like $10 each |
02:58.38 | [TK]D-Fender | Zylkron, if all you're doing is doing G.729 from your phone to * and then right to OpenSER with G.729, then * has done NOTHING. |
02:59.06 | JT | [TK]D-Fender: he says it also does voicemail |
02:59.07 | [TK]D-Fender | Zylkron, Try starting from the beginning and show where you want * to actually do something productive. |
02:59.30 | [TK]D-Fender | JT : Fine, so record your prompts in G.729, do your RECORDINGS in G.729 and be done with it. |
02:59.34 | Qwell | you can only do voicemail if every user uses g729 |
02:59.44 | Zylkron | uhm |
02:59.57 | Qwell | unless you transcode for some |
03:00.27 | [TK]D-Fender | Qwell, So far all he's said was "G.729" so that seems consistent enough for me... |
03:00.40 | Qwell | fair assumption |
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03:00.55 | Qwell | so, why SER? |
03:01.34 | [TK]D-Fender | Qwell, we were just wondering "wht ASTERISK" all this time? Maybe the question should have been "WHY!?!?!??!!" in a "pleading to God" osrt of way :) |
03:01.54 | Qwell | well, that would be as futile |
03:02.24 | [TK]D-Fender | Qwell, yes, but we'd have just ignored it like God would and have made better use with our time :p |
03:02.25 | Zylkron | *sigh* =p |
03:02.37 | Zylkron | Thanks for the Hekp |
03:02.39 | Zylkron | bitch =p |
03:03.00 | Qwell | <[TK]D-Fender> Zylkron, Try starting from the beginning and show where you want * to actually do something productive. |
03:03.05 | Qwell | ^ a very fair request |
03:03.14 | [TK]D-Fender | Zylkron, So if you want * as an application server without G.729 licenses make sure all recordings are in G.729 and will be made in G.729 |
03:03.19 | Zylkron | so I dont get why they wont pay 10 bucks either, this one company has liek 200 phones waiting to be deployed :P |
03:03.28 | JT | liek fully |
03:03.29 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
03:03.33 | JT | don't call people bitches :P |
03:03.37 | JT | they are trying to help :P |
03:03.45 | [TK]D-Fender | JT : gnarly! |
03:03.45 | Qwell | [TK]D-Fender: and, that wasn't quite what I meant by futile.. |
03:03.47 | JT | you aren't exlaining the scenario fully :P |
03:03.51 | Qwell | take that as you will though :p |
03:04.27 | Zylkron | thats because I know its a stupid question which took me days to figure out =p, and I knew I'll be bitched at because of it :p |
03:04.44 | Qwell | Zylkron: so far, I haven't really even seen a question... |
03:05.05 | Qwell | "is <xyz> possible?" yes, it's just software |
03:05.08 | Qwell | anything possible |
03:05.21 | Zylkron | THATS MAYBE BECAUSE YOUR IQ IS 200% BETTER THAN ME O WIZARD OF QWELL |
03:05.27 | Zylkron | so I am DUmb |
03:05.32 | Zylkron | so what |
03:05.38 | Zylkron | zzzz |
03:05.50 | *** mode/#asterisk [+b %Zylkron!*@*] by Qwell |
03:05.52 | Qwell | moving on |
03:06.06 | JT | instead of acting like an immature fool, you could give the requested info |
03:06.07 | JT | oh well |
03:06.10 | JerJer | damnit - did i miss a flame war? |
03:06.24 | [TK]D-Fender | JerJer, No, the embers still burn bright |
03:06.28 | *** mode/#asterisk [+b Zylkron!*@*] by Qwell |
03:06.31 | *** kick/#asterisk [Zylkron!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
03:06.35 | [TK]D-Fender | JerJer, and trust me its the only "bright" thing going on ;) |
03:06.35 | *** mode/#asterisk [-b %Zylkron!*@*] by Qwell |
03:06.47 | JerJer | !op me |
03:06.55 | Qwell | !op JerJer |
03:06.58 | Qwell | didn't work :P |
03:07.00 | JerJer | :) |
03:18.01 | Juggie | holy shit |
03:18.09 | Juggie | is anyone else tired of the ast_frame_digital conversation? :P |
03:18.15 | Juggie | its been filling my inbox for like 2 weeks |
03:18.20 | Juggie | on asterisk-dev that is. |
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03:34.56 | dudes | busy busy busy in here tonight holy batman |
03:35.25 | [TK]D-Fender | dudes, SHH!!!! you'll wake the crickets! |
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03:35.34 | *** mode/#asterisk [+o d3wayne] by ChanServ |
03:35.38 | d3wayne | ~sipnat |
03:35.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
03:35.53 | dudes | If you're looking for crickets I could help you out |
03:35.58 | dudes | you don't mind if they are dead, eh? |
03:37.37 | dudes | what's NAT? |
03:38.09 | dudes | is that some bug that buzzes in your ear and pissed a homi off |
03:39.05 | [TK]D-Fender | dudes, hukt on fonix werkt 4 u! |
03:39.25 | dudes | I don't speak french |
03:42.04 | Nugget | S.O.C.K.S! |
03:42.30 | *** join/#asterisk dadicool (n=mrordaz@124.107.96.235) |
03:42.47 | dudes | I never got that hooked on phonics crap. If the person was that stupid, how do they expect them to remember the alphabet? |
03:43.56 | [TK]D-Fender | dudes, now you know your A-B-Q's! |
03:44.18 | dudes | yes I do |
03:44.24 | dadicool | Please help.... I have purchased G729 codec and am now having problems with the install network card. I have already tried re-installation of my box and got to ask digium for a reset on my G729 registration. Anybody here from digium? Please help. Thanks. |
03:46.23 | dudes | call them |
03:48.11 | dadicool | I would have but that would cost me... I'm from the Philippines. |
03:48.45 | dudes | you don't have termination to the US? |
03:49.24 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
03:49.40 | dudes | if you message me, I'll give you access to my trunk so you can call |
03:49.52 | Juggie | dadicool, doesnt the default asterisk extensions.conf provide termination into the digium pbx misery? |
03:49.53 | dadicool | how do i do that? thanks. |
03:50.58 | Juggie | IAX2/misery.digium.com |
03:51.05 | [TK]D-Fender | Juggie, You keep GIVING away all out Seek-Rats! |
03:51.20 | dadicool | pardon me for not being so techie... I am a newbie with VOIP. :-( |
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03:53.09 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
03:53.15 | dudes | you know how to get that to work eh? |
03:53.19 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
03:54.16 | dudes | wow, it's an outtolunc |
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03:54.55 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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04:06.51 | *** join/#asterisk dijungal (n=kdaniel@208.0.231.85) |
04:07.37 | dijungal | i just put a TE410P card in an asterisk install and i'm getting the following error "HDLC Abort (6) on Primary D-channel of span " 2/1 any ideas? |
04:08.02 | dijungal | i've only configured port 1 & 2, since i'm only using those for now |
04:10.20 | [TK]D-Fender | dijungal, pastebin "cat /proc/interrupts", "dmesg", your zaptel.conf and zapata.conf |
04:10.22 | [TK]D-Fender | ~pb |
04:10.22 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
04:10.53 | dijungal | :O |
04:10.58 | dijungal | ok hold |
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04:16.39 | dijungal | [TK]D-Fender: http://pastebin.com/d336f3ed0 |
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04:17.56 | [TK]D-Fender | dijungal, 2 T1 PRI's from the telco? |
04:18.41 | dijungal | from a cisco 3660 |
04:20.14 | [TK]D-Fender | dijungal, Ok, well set span 2 to 2,2,0 instead of 2,1,0. otherwise you have 1 & 2 fighting for "primary" which might cause clocking issues |
04:21.11 | dijungal | hmmm |
04:23.16 | dijungal | ok did that...., ran ztcfg -vvvv and restarted asterisk |
04:24.14 | dijungal | [TK]D-Fender: so far no errors |
04:24.27 | [TK]D-Fender | dijungal, Did you always get them immediately and constantly? |
04:24.42 | dijungal | yes on incoming calls |
04:26.01 | dijungal | [TK]D-Fender: so my third span should be 3,3,0 ? |
04:26.11 | [TK]D-Fender | dijungal, yup |
04:26.34 | dijungal | ok can u explain the timing column > |
04:26.35 | dijungal | ? |
04:26.52 | ManxPower | dijungal: think of timing as "sync source priority" |
04:27.06 | dijungal | i thought it meant 0 - timing from telco and 1 - time internally |
04:27.08 | ManxPower | where 0 means "never use this span as a source for sync" |
04:27.37 | [TK]D-Fender | dijungal, 0 = PROVIDE timing |
04:27.41 | dijungal | ok i just got the hdlc error again.... |
04:27.50 | ManxPower | or more correctly 0 means "never use this span as a source to get sync from" |
04:28.01 | ManxPower | dijungal: contact digium support |
04:28.27 | dijungal | hmm.. |
04:28.44 | mmlj4 | hey ManxPower |
04:28.48 | dijungal | what if i put all spans timing 0 |
04:28.51 | dijungal | ? |
04:28.52 | ManxPower | that error means "got corrupted data from the pci bus" It could be caused by many things |
04:29.16 | ManxPower | dijungal: then asterisk will not have a timing source and audio will be terrible |
04:29.35 | ManxPower | your faxes will fail. |
04:30.03 | [TK]D-Fender | calls may drop randomly and other craziness |
04:30.13 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
04:30.58 | dijungal | k |
04:32.27 | ManxPower | dijungal: basically you would get corrupted bits every once in a while. |
04:36.56 | outtolunc | i'd say take that usb device off |
04:37.33 | dijungal | ok thanks guys.. i've reverted to span=1,1,0,esf,b8zs |
04:37.33 | dijungal | span=2,2,0,esf,b8zs |
04:37.43 | dijungal | let's see how it goes in production tomorrow |
04:37.48 | dijungal | with 25 agents on :) |
04:39.06 | dijungal | I've also had an annoying issue where an agent will be on a call and might get another call on the phone, i've tried using call-limit=1, but that works for while, then seems to keep the agent in use, so no more calls go to that agent... |
04:39.15 | dijungal | this is on asterisk 1.2.19 |
04:39.34 | dijungal | is this a known problem? any work arounds, solutions? |
04:39.39 | dudes | dont send them a call then |
04:39.45 | [TK]D-Fender | dijungal, limit the calls on the phone if you can. |
04:41.25 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
04:41.44 | dijungal | how so? |
04:41.51 | [TK]D-Fender | dijungal, depends on the phone |
04:42.02 | dijungal | u mean on the softphone itself? |
04:42.02 | [TK]D-Fender | JunK-Y, I don't want to meet your mom! |
04:42.32 | ManxPower | on polycoms it's something like "max.calls.per-appearance=1" in the phone1.cfg |
04:42.49 | ManxPower | you'll have to figure it out on the softphone |
04:44.18 | dijungal | k, it's the eyebeam phone |
04:44.36 | ManxPower | ~softphones |
04:44.46 | dudes | then disable call waiting in sip maybe? |
04:45.01 | ManxPower | All Softphones Suck (tm)(c) 2007 ManxPower |
04:45.12 | [TK]D-Fender | ~softphone |
04:45.13 | jbot | something that should be drug out into the street and shot |
04:45.13 | dudes | you're telling me |
04:45.28 | dijungal | lol |
04:45.29 | ManxPower | [TK]D-Fender: you are such a jbot queen |
04:45.32 | dudes | their update sucks |
04:45.40 | [TK]D-Fender | ~[TK]D-Fender, |
04:45.45 | [TK]D-Fender | ~[TK]D-Fender |
04:45.46 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
04:45.51 | ManxPower | ~manxPower |
04:45.51 | jbot | you are, like, your God. someone you should send lots of money to because he helps so many people. |
04:45.51 | [TK]D-Fender | ^^^^^^ |
04:46.04 | JunK-Y | [TK]D-Fender: thats what you said after what you did with her,last night, huh! |
04:46.13 | ManxPower | I didn't know that said that, but I'm not complaining |
04:46.35 | [TK]D-Fender | ManxPower, keep whoring away there! |
04:46.37 | [TK]D-Fender | :p |
04:46.48 | dudes | then again eyebeam is done by Canadians |
04:46.50 | ManxPower | [TK]D-Fender: Everyone wants it for free! |
04:47.36 | [TK]D-Fender | dudes, Apparently.... |
04:47.51 | dijungal | lol!!! |
04:47.57 | dijungal | u guys crack me up... |
04:48.36 | dijungal | ok let me throw in another topic to gouge about "offshore call centers"... ok... and... GO!. |
04:48.36 | [TK]D-Fender | JunK-Y, Stacy's mom has got it going on! |
04:49.19 | outtolunc | look in the sky.. its a bird .. a plane.. no it's another phone being drop-kicked |
04:49.36 | JT | it's an iPhone |
04:49.41 | ManxPower | dijungal: more and more companies are "onshoring" where they go into a rural area, train the locals and import any additional required people and open a call center. |
04:50.04 | ManxPower | or a factory or a data center, etc |
04:50.26 | [TK]D-Fender | dijungal, http://video.google.com/videoplay?docid=7362895170514642646 |
04:50.32 | *** join/#asterisk cmwt (n=bit_frog@adsl-76-200-102-66.dsl.pltn13.sbcglobal.net) |
04:50.36 | dijungal | lol |
04:50.40 | JT | why would you put a datacentre in a rural area unless it was for disaster recovery? |
04:50.50 | ManxPower | JT: The iPhone is cool, but not hundreds of dollars cool. |
04:51.03 | JT | it's not that cool :P |
04:51.12 | ManxPower | JT: Why not? cheap power, cheap realestate, put it near a university town and bring inthe bandwidth |
04:51.22 | dudes | I hate the iphone |
04:51.28 | ManxPower | cheap labor too |
04:51.29 | JT | dunno how cheap the power would be |
04:51.35 | JT | rural power is also shit power |
04:51.37 | dijungal | i have a nokia e61i with SIP capabilities... kicks @@SS |
04:51.46 | dudes | just another thing to pre-occupy peoples minds and crash into shit |
04:51.47 | ManxPower | JT: put in your own power plant |
04:51.55 | JT | and there's no bandwidth in the middle of nowhere |
04:52.00 | ManxPower | all these cool phones are GSM |
04:52.33 | ManxPower | JT: I don't mean in the middle of nowhere, I mean like 20 or 40 miles outside of a larger city. |
04:52.40 | ManxPower | google has been doing that |
04:53.01 | JT | well i debate the ruralness then ;) |
04:53.12 | JT | but still, near cities is the best for connectivity |
04:53.37 | ManxPower | http://seattlepi.nwsource.com/business/280581_datacenter09.html |
04:53.50 | dijungal | ahhh.. is "call-limit" flawed in *1.2 ? |
04:53.56 | dudes | yes it is |
04:54.00 | dudes | hahaha |
04:54.23 | dudes | I love beer |
04:54.47 | dijungal | riiighttt.... |
04:55.00 | dijungal | and the price of rice in china just spiked! |
04:55.13 | dudes | the US sent Rice to China |
04:55.19 | ManxPower | http://www.thedalleschronicle.com/news/2007/08/news08-05-07-01.shtml |
04:56.43 | *** join/#asterisk watchy (n=watchy@c-68-51-54-72.hsd1.ar.comcast.net) |
04:56.56 | watchy | anyone here use a solid state drive with linux? |
04:56.56 | dudes | http://www.usatoday.com/news/world/2006-10-19-rice-south-korea_x.htm |
04:57.12 | dudes | That's so funny. They sent Rice to China hahaha |
04:57.16 | watchy | im trying to install trixbox on a solidstate drive and getting mad problems |
04:57.37 | ManxPower | watchy: there was a discussion about that this week on the asterisk -users mailing list. |
04:57.46 | watchy | what did they say? |
04:58.48 | dijungal | dudes... lol "US sends Rice to china"!!! that' as a good one! |
04:59.15 | dudes | I get a kick out of that everytime I think about it |
04:59.17 | dudes | the Irony |
05:00.59 | dijungal | me tooo |
05:01.00 | dijungal | lol |
05:01.24 | dijungal | i'll email that one to the IT guys... "US sends Rice to China, due to shortage" :) |
05:01.52 | dudes | I like Condy though |
05:02.04 | dudes | she a smart lady --- but her name takes the cake |
05:02.53 | dudes | anyone here from the Toronto area? |
05:03.38 | dudes | where you from dijungal? |
05:03.53 | watchy | im trying to install trixbox on a solidstate drive and getting mad problems |
05:04.37 | *** join/#asterisk Joneser (n=Joneser@pool-71-170-201-50.dllstx.fios.verizon.net) |
05:05.44 | [TK]D-Fender | watchy, and you're asking HERE :p |
05:05.57 | watchy | well its a general question to all your smart losers :) |
05:06.07 | watchy | i think its linux issue personaly not trixbox |
05:06.14 | watchy | getting wierd hd erros |
05:06.23 | watchy | you ever use a solid state drive with linux> |
05:06.35 | [TK]D-Fender | watchy, nope. |
05:06.59 | [TK]D-Fender | watchy, try another distro and see if its the kernel version |
05:07.03 | watchy | you think they are a good idea for a * box? |
05:07.39 | dudes | I hate trixbox |
05:07.48 | [TK]D-Fender | watchy, SSD's should have the same long term writing wear out issues as flash>IDE dongles, so I'd consider thingfs carefully |
05:08.00 | dijungal | dudes: i'm in st. lucia right now |
05:08.34 | watchy | tk: really? |
05:08.34 | dudes | a lot of babes imagine |
05:08.37 | dijungal | watchy: there's a #trixbox |
05:08.45 | *** join/#asterisk Rsaman2 (i=Alain@c1-169-15.tpr.isadsl.co.za) |
05:08.46 | Rsaman2 | hello |
05:09.08 | Rsaman2 | quick quiz : I am in asterisknow the shit gui version, stuck in some menu... how do i terminate it ? |
05:09.26 | fujin | ctrl+alt+f1 |
05:09.28 | fujin | login |
05:09.33 | fujin | killall shitguiversion |
05:10.20 | [TK]D-Fender | watchy, its been discussed before due to CDR writes, VM, logging, etc. you know flash has a limited write-life |
05:12.10 | watchy | well i dont think solid state disk are considered flash drives |
05:12.19 | WilliamK | hey [TK]D-Fender, flash cards are essentially re-programable eeproms right? |
05:12.38 | dudes | does it matter in the end |
05:12.41 | [TK]D-Fender | WilliamK, I don't know the real tech nitty gritty... |
05:12.50 | watchy | this drives failure thing is 1,3 million hours |
05:12.56 | watchy | the mtbf is |
05:12.59 | dudes | no moving parts --- you loss the mystery |
05:13.19 | WilliamK | I've often wondered about taking 2 compact flash cards using an IDE adapter |
05:13.22 | JT | dudes: yes, it matters |
05:13.23 | watchy | i mean they put solid state in laptops for windows now |
05:13.29 | dudes | psst |
05:13.33 | WilliamK | same way Global Technology Associates does for their firewalls |
05:13.37 | watchy | if solid stateonly supported so many writes you couldnt evfen use windows |
05:13.43 | watchy | it would kill the drive in a week |
05:14.01 | JT | would depend on the solid state technology |
05:14.10 | dudes | I have drives from when I was 16 that are still rock solid |
05:14.26 | JT | what type of drives? |
05:14.28 | WilliamK | It'd be interesting to load windows on an 8gb flash, and run a burn in util to see how long it'll run for |
05:14.34 | JT | and you could still be 16 |
05:14.34 | watchy | jt: hmm hold |
05:14.37 | dudes | Maxtor drives |
05:14.43 | JT | hard drives |
05:14.44 | dudes | I'm 23 |
05:14.59 | JT | i'm sure everyone has drives that have lasted that long |
05:15.03 | JT | a lot fail too |
05:15.08 | [TK]D-Fender | WilliamK, not the same as firewalls.... the store their settings on flash, but load and run them from RAM. Writting is very sparse. |
05:15.09 | dudes | maybe |
05:15.09 | watchy | http://www.newegg.com/Product/Product.aspx?Item=N82E16820208316 |
05:15.12 | watchy | that card jt |
05:15.21 | watchy | i mean that drivce |
05:15.35 | dudes | but they have sat in a POS Athlon box for years and they still rock out |
05:15.44 | watchy | i cant get linux on it though |
05:15.44 | JT | solid state is also less heat and less power |
05:15.48 | WilliamK | true, just thinking about the concept as a whole though |
05:15.53 | JT | i'm talking about for server/appliance use |
05:15.54 | dudes | powers cheap |
05:15.58 | JT | more important than desktop |
05:15.58 | JT | ... |
05:16.05 | dudes | and air conditioning fixes the head issues |
05:16.10 | dudes | or the window in the winter |
05:16.28 | WilliamK | power is cheap depending on which part of the US you're in |
05:16.30 | JT | wtf are you going on about |
05:16.49 | dudes | this that that |
05:16.54 | WilliamK | I don't know that it's cheap overseas - alot of places have power issues and asking for AC you might as well wish upon a star |
05:17.01 | JT | dudes: are you drunk? |
05:17.18 | dudes | no I'm quite sober |
05:17.18 | WilliamK | JT, I can only claim to have taken my meds |
05:17.20 | WilliamK | :) |
05:17.46 | JT | watchy: says flash, wonder if it's write sensitive |
05:18.02 | dudes | <WilliamK> - where might one wish upon a star for AC? |
05:18.02 | [TK]D-Fender | JT : Looks like a duck.... |
05:18.17 | [TK]D-Fender | dudes, Walmart :) |
05:18.19 | dudes | I can imagine a few places -- but enlighten me |
05:18.30 | dudes | Wal-Mart has a kick ass AC here |
05:18.36 | dudes | heater too |
05:19.11 | dudes | it's refreshing in the summer to walk into wal-mart |
05:19.29 | dudes | I really like when the lassies walk in heh |
05:22.50 | watchy | hey tk you tried a sangoma PCI-E card yet? |
05:22.53 | [TK]D-Fender | ok, time to hit the sack... later all |
05:23.03 | [TK]D-Fender | watchy, Nope, no need.... |
05:23.08 | watchy | we bought a t1 PCI-E sangoma card to play with |
05:23.26 | [TK]D-Fender | watchy, have fun and don't let it run out of smoke... |
05:23.44 | watchy | yea, its for a customer were installing tommorow |
05:24.16 | watchy | if we can get this stupid linux on this solid state working |
05:24.39 | JT | 1 day lead time? nice |
05:24.53 | watchy | jt: well we can always use a normal drive |
05:25.05 | watchy | we planned on being up all night doing it anyways |
05:25.12 | watchy | so we have plenty of time |
05:25.20 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
05:25.35 | JT | heh ok |
05:29.32 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
05:30.04 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@h460eb882.area3.spcsdns.net) |
05:30.58 | *** join/#asterisk appelza (n=b@dsl-242-33-145.telkomadsl.co.za) |
05:32.22 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
05:34.16 | appelza | Wow |
05:34.21 | appelza | The xorcom distro is AWESOME |
05:34.22 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
05:34.51 | dlynes | mudirc |
05:38.37 | remmo | yawn |
05:47.09 | Rsaman2 | arg |
05:47.17 | dlynes | ? |
05:47.17 | Rsaman2 | i am still stuck in that stupid menu |
05:47.21 | Rsaman2 | for asterisknow |
05:47.25 | Rsaman2 | when the linux boots up |
05:47.29 | Rsaman2 | how do i terminate it ? |
05:47.35 | dlynes | what's the menu look like? |
05:47.38 | Rsaman2 | and get a linux terminal |
05:47.39 | dudes | you tell it to |
05:47.45 | dudes | then you rub it and it does it |
05:47.48 | dlynes | alt-f2 maybe? |
05:48.00 | *** part/#asterisk Shazoo (n=feng_me@60.216.14.2) |
05:48.02 | Rsaman2 | still in |
05:48.07 | Rsaman2 | i know its a hotkey |
05:48.08 | dlynes | are you in a console, or a gui? |
05:48.09 | Rsaman2 | but i forget it ... |
05:48.13 | Rsaman2 | well,, |
05:48.23 | Rsaman2 | its the asterisknow console menus |
05:48.27 | Rsaman2 | its the asterisknow console menu |
05:48.35 | dlynes | alt-f1, alt-f2, alt-f3, alt-f4, ... |
05:48.52 | JT | Rsaman2: wrong channel, this isn't #asterisk-gui |
05:49.04 | Rsaman2 | i know |
05:49.07 | Rsaman2 | get it thanks |
05:49.11 | dlynes | or maybe you haven't gone far enough yet, to have the vt's loaded up |
05:49.28 | dlynes | Rsaman2: you could also try #asterisknow |
05:49.42 | dlynes | or maybe it's #asterisk-now |
05:49.49 | dlynes | list *asterisk* |
05:49.51 | dlynes | erm |
05:50.30 | dlynes | yeah...#asterisknow |
05:51.04 | dlynes | erm #asteriskNOW....don't know if it makes any difference that the 'now' is in capslock or not |
05:51.35 | JT | no |
05:51.39 | JT | irc is case insensitive |
05:51.40 | *** part/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net) |
05:51.57 | JT | but people with good irc clients can tell how you capitalised it when you join |
05:52.17 | JT | user and chan modes are case sensitive, however |
05:53.52 | dlynes | ic |
05:53.53 | dlynes | brb |
05:54.39 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
05:57.44 | *** join/#asterisk RsaMan (n=aa@196.210.155.3) |
06:05.25 | awk | hmm, is it vital for txgain to be upped in order to use hylafax/iaxmodem, etc |
06:05.36 | awk | i'm getting a fax tone, but nothing as strong as it normally gets pushed out |
06:06.00 | awk | hence its not passing the fax through, only problem is I don't want it that people hear this intense sound on the other ed when we make a phone call |
06:06.04 | awk | so how do you balance it |
06:06.15 | awk | sangona sugests you put the gain up to 8 |
06:06.18 | awk | but thats focking load |
06:07.12 | Strom_M | awk: you call the telco's milliwatt test |
06:07.32 | Strom_M | and you adjust your gain that way |
06:11.19 | *** join/#asterisk _10nix_ (n=hyjnx@user-160u96o.cable.mindspring.com) |
06:21.57 | mistermocha | awk: are you using a dedicated fax line? |
06:22.13 | mistermocha | you can always just pump the gain on the one FXS channel |
06:22.44 | appelza | whats the debian etch package for amportal called? |
06:24.30 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
06:28.53 | watchy | anyone ever put linux in a vlan> |
06:29.38 | [hC] | yeah |
06:29.43 | [hC] | its pretty easy |
06:29.48 | [hC] | easier depending on your distro |
06:31.36 | watchy | centos |
06:31.38 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
06:31.44 | watchy | what do i do to say put linux on vlan 1? |
06:32.42 | [hC] | vlan 1 is the default |
06:32.49 | watchy | ok vlan 2 |
06:32.54 | *** join/#asterisk Corydon76-home (i=eleven@pdpc/supporter/sustaining/Corydon76-home) |
06:32.54 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
06:32.54 | watchy | i just wanna know how to change it |
06:33.12 | [hC] | im not sure on centos, i use debian.. but the way it does it, its just a wrapper to vconfig |
06:33.16 | [hC] | search google for vlan settings centos |
06:33.20 | [hC] | im sure you'll find an easy howto |
06:33.25 | watchy | thanks |
06:33.49 | watchy | hey hc you get ringing the base to work instead of call waiting beep? |
06:33.59 | [hC] | havent tried yet nope |
06:34.18 | watchy | you ever use a solid state disk with asterisk |
06:37.33 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
06:39.02 | *** join/#asterisk cayorde (n=flexable@host201-100-dynamic.17-87-r.retail.telecomitalia.it) |
06:41.08 | *** join/#asterisk RsaMan (n=aa@196.210.154.3) |
06:41.10 | RsaMan | hello all |
06:41.15 | RsaMan | i have a slight problem |
06:41.19 | RsaMan | my asterisk is buggered... |
06:41.28 | RsaMan | [root@localhost ~]# asterisk |
06:41.28 | RsaMan | [root@localhost ~]# asterisk -r |
06:41.28 | RsaMan | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
06:41.34 | RsaMan | wont load,, |
06:41.48 | RsaMan | how would i uninstall it ? |
06:42.35 | RsaMan | running asterisk 1.2 |
06:42.39 | RsaMan | i dont just want to reinstall |
06:42.46 | RsaMan | surely that will break something ? |
06:43.12 | Strom_M | how about... |
06:43.43 | RsaMan | :) |
06:43.43 | Strom_M | run asterisk -cvvvvvvvg and see why it crashes on startup |
06:43.58 | RsaMan | asterisk -cvg? |
06:44.18 | Strom_M | -cvvvvvvvg |
06:44.52 | RsaMan | oh crap |
06:45.18 | RsaMan | http://pastebin.com/d403ba263 |
06:45.22 | RsaMan | zap problems |
06:45.37 | RsaMan | i dont get it |
06:46.02 | Strom_M | are your drivers loaded? |
06:46.04 | JT | make sure zaptel is running |
06:46.16 | Strom_M | did you run ztcfg -vv? |
06:46.45 | RsaMan | oh |
06:46.46 | RsaMan | ran it now |
06:46.47 | RsaMan | working |
06:46.54 | RsaMan | i ran zaptel make config |
06:47.04 | RsaMan | but didnt seems to load zaptel on start :( |
06:47.19 | JT | you make some weird assumptions there |
06:47.33 | RsaMan | :( |
06:47.35 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
06:47.41 | RsaMan | what does ztcfg -vv do then ? |
06:47.51 | JT | configures zaptel |
06:48.01 | JT | reads /etc/zaptel.conf |
06:48.12 | RsaMan | do i only need to run it once ? |
06:48.31 | JT | per boot, unless you unload zaptel stuff |
06:49.23 | RsaMan | how can i make it read each time i boot then ? |
06:50.38 | watchy | can i come vlan polycoms by provision? |
06:50.54 | RsaMan | thats why i though make config in the zaptel source directory would do that |
06:51.10 | JT | RsaMan: do you know anything about the linux boot process? |
06:51.16 | JT | watchy: err, say what? |
06:51.40 | mistermocha | watchy: I believe you might be able to... |
06:51.44 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
06:51.56 | watchy | can i setup polycoms on a vlan |
06:52.03 | JT | sure |
06:52.04 | watchy | using ftp provisioning |
06:52.49 | RsaMan | JT : does not seem like it :( |
06:54.11 | JT | each distribution has a particular method of calling init scripts |
06:54.55 | mistermocha | watchy: the Polycom manual says you can from sip.cfg, but it doesn't say how |
06:55.05 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
06:59.12 | watchy | if i setup say 2 linux boxes on Vlan 2 |
06:59.21 | watchy | do i have to hook them to a switch that supports vlans? |
07:00.21 | [hC] | it either has to support vlans, or be so dumb that it just passes whatever it gets along |
07:00.34 | [hC] | if it does support vlans, chances are you have to enable those ports to work on the new vlan |
07:00.42 | [hC] | otherwise it should just pass data if its a dumb switch |
07:02.10 | mistermocha | the vlan setting in the phone won't mean anything unless the networking equipment understands it |
07:04.37 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
07:07.49 | watchy | thanks' |
07:08.24 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
07:11.37 | Aurs | is this normal: |
07:12.21 | Aurs | [root@sipgw01 /usr/src/zaptel-1.4.5.1]# /etc/init.d/zaptel start |
07:12.22 | Aurs | [root@sipgw01 /usr/src/zaptel-1.4.5.1]# |
07:12.41 | Aurs | (centos5). |
07:15.14 | litage|w | why would asterisk write CDRs to /var/log/asterisk/cdr-custom/ but not /var/log/asterisk/cdr-csv/ ? |
07:23.32 | WilliamK | hey Aurs, shouldn't you be using /etc/rc.d/init.d/zaptel start |
07:23.33 | WilliamK | ? |
07:24.34 | WilliamK | neat, looks like someone Alias'd it |
07:25.23 | WilliamK | oh by the way, if someone's paying any thoughts to the SVN...someone tacked some comments into the zaptel.init file that need to come out so it'll exec correctly |
07:25.54 | litage|w | n/m |
07:33.33 | Aurs | WilliamK: zaptel.conf was missing, so it just got to a exit 0 line |
07:46.27 | *** join/#asterisk dlynes_ (n=dlynes@d154-20-9-152.bchsia.telus.net) |
07:47.57 | *** join/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
07:49.09 | n3glv | can someone tell me, I see sip/2.0 on some things, I have an old zyxel wifi phone that will not connect, could it be running sip/1.0 (if there is such a thing) |
07:49.17 | n3glv | would this cause 401 errors? |
08:01.10 | penguinFunk | ~pb |
08:01.11 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
08:01.47 | penguinFunk | anyone know what would cause this: http://pastebin.com/m645dbea0 ?? |
08:01.56 | dan__t | Anyone rockin' LTSP today by chance? |
08:02.02 | penguinFunk | something looks flakey |
08:02.05 | penguinFunk | :( |
08:02.37 | *** join/#asterisk appelza (n=d@dsl-240-171-43.telkomadsl.co.za) |
08:12.04 | *** join/#asterisk Shazoo (n=feng_me@60.216.14.2) |
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09:01.15 | *** part/#asterisk Shazoo (n=feng_me@60.216.14.2) |
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09:12.54 | yxa | if my digium pri card has onboard ec, do I need to do anything else other than compiling and modprobing zaptel? |
09:16.41 | JT | what does the hardware ec have to do with zaptel? |
09:16.57 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
09:18.02 | Nugget | You still need echocancel=yes in zapata.conf to enable the hardware echo cancellation |
09:18.08 | awk | hm does asterisk record history by default for registered extensions, if so where? |
09:18.14 | Nugget | all the other echo-related settings in zapata.conf are ignored in that case |
09:18.17 | awk | I need to get an ip for a phone that was registred and is now off |
09:22.24 | ai-a[off] | awk: turn it back on. |
09:25.12 | Uatec | hi |
09:25.12 | *** join/#asterisk Loke-pt (n=kumar@mail.netvita.com) |
09:25.16 | Uatec | i reinstalled my asterisk box |
09:25.36 | Uatec | but now the CLI is not showing the steps of my dialplan as a calls steps through it |
09:25.52 | Uatec | it used to say the line of the extensions.conf that it was on in purple |
09:25.57 | Uatec | so i could trace how my callplans were going |
09:26.01 | Uatec | but it's gone |
09:26.06 | Uatec | does anybody know how i can get it back? |
09:27.57 | ai-a | whats your exact command line you use to run * Uatec ? |
09:29.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
09:34.45 | Uatec | umm |
09:34.46 | Uatec | i don't konw |
09:34.49 | Uatec | it starts on boot |
09:34.51 | Uatec | always has |
09:34.52 | Uatec | lol |
09:35.27 | Uatec | it's there in chkconfig thoug |
09:35.28 | Uatec | h |
09:41.01 | Uatec | ok |
09:41.09 | Uatec | well i've just restarted it with "asterisk -vvvgc" |
09:41.18 | Uatec | but i can't exit the CLI without terminating asterisk |
09:43.19 | Uatec | ahah |
09:43.23 | Uatec | "service asterisk start" |
09:43.25 | Uatec | w00t |
09:48.02 | enioh | Uatec: no, you have to run asterisk without -c |
09:48.12 | enioh | and then connecting to the console with asterisk -r |
09:48.29 | enioh | so you will be able to leave with quit |
09:52.28 | Uatec | enioh, when i ran asterisk with the -c it just didn't give me the console |
09:52.41 | Uatec | but i was still running it in that terminal |
09:54.08 | enioh | with -c, you should have the console |
09:54.22 | enioh | typing something doesn't do anything ? |
09:54.55 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
09:58.46 | *** join/#asterisk c1|freaky (i=alpha@team.code-1.de) |
10:02.06 | c1|freaky | hi all. can asterisk be used for only voIP talks - f.e.: I want to put asterisk on a server and let people talk with each other using some software ... |
10:02.26 | c1|freaky | private use |
10:02.28 | c1|freaky | nothing else |
10:03.06 | c1|freaky | ? |
10:04.34 | Teln1100A | yea for sure... make sip accounts for them and assign extensions |
10:05.00 | Teln1100A | xlite is a good softphone but any other sip phone should work |
10:05.12 | awk | I asked earlier with no luck, when using hylafax is it vital to set your txgains up drmatically |
10:05.15 | c1|freaky | ok, thank you :) |
10:05.16 | awk | deegan: to 8db |
10:05.23 | awk | errr eg! |
10:07.14 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
10:07.51 | c1|freaky | if i run an asterisk server, can ppl from other servers be reached? |
10:08.02 | c1|freaky | Teln1100A: thank you :) |
10:08.22 | hwt | hey, i have an asterisk/meetme server that crashed hard the other day. i am using ztdummy on 2.6 as timing source. we have maybe 50-60 users on peak hours. could it be that ztdummy does not scale up to this number of users? |
10:08.47 | hwt | or is the server hardware unstable? the box just halted completely. |
10:08.54 | awk | what was your server load? |
10:09.05 | awk | whre you using sip or iax peers |
10:09.18 | hwt | awk: 0.3 maybe, when we had ~20 users. |
10:09.20 | hwt | awk: SIP only. |
10:09.33 | hwt | how can i confirm that ztdummy actually works? |
10:09.53 | hwt | i got reports about choppy sounds with ~20 users. |
10:10.19 | awk | coppy sound |
10:10.21 | hwt | ztdummy/zaptel is loaded without warnings. |
10:10.27 | awk | hmm, whats up with your link |
10:10.31 | awk | could be a b/w issue |
10:10.44 | hwt | awk: but that shouldn't cause the box to die. |
10:10.50 | awk | no it shouldn't |
10:10.54 | hwt | we have 100mbit |
10:10.54 | awk | but could cause choppy sound |
10:11.00 | hwt | ah, yeah. |
10:11.17 | awk | the box dying would be hard without running a log |
10:11.24 | awk | you can set asterisk to dump its core if there is a problem |
10:11.28 | awk | then monitor that core log |
10:11.41 | awk | the next time that happens |
10:11.47 | hwt | but we had another box running there, with a zaptel card, running 1.2cvs whiched coped just fine |
10:12.13 | hwt | i could not find anything in either /var/log/messages or in asterisk/messages |
10:12.18 | awk | hmm, i have had not problems using ztdummy, but that could just be me |
10:12.25 | hwt | how many users? |
10:12.27 | hwt | how much mem? |
10:12.29 | hwt | cpu? |
10:12.29 | awk | naaa, if it core dumps you wont find it in those logs |
10:12.41 | awk | p4 3ghz, 1 gig ram / 30users |
10:12.41 | hwt | brb |
10:12.48 | awk | started getting choppy around there |
10:12.56 | awk | but that our link i believe |
10:13.12 | awk | also it takes strain on a box to do transcoding |
10:13.16 | awk | what codec where yo usuing |
10:23.19 | *** join/#asterisk nohup- (i=hmmmph@203.81.206.134) |
10:26.39 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
10:27.33 | c1|freaky | do I neen an voip provider if i run asterisk? |
10:28.20 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
10:30.53 | penguinFunk | no |
10:31.58 | penguinFunk | only if you don't plan to use an FXO card or digital card and still want to make calls to everywhere |
10:32.15 | penguinFunk | i.e internet only based system |
10:34.51 | c1|freaky | i would like to run it for me and some friends on an dedicated internet server |
10:35.07 | penguinFunk | to be able to call anyone? |
10:35.30 | penguinFunk | are you going to install some hardware to inferface with a telco? |
10:35.39 | penguinFunk | interface* |
10:36.03 | penguinFunk | if not then you need a voip provider |
10:36.33 | c1|freaky | no not to be able to call anyone - just each other |
10:36.42 | nohup- | if you're just wanting to run it for yourself and a couple of your friends on a dedicated server then no you dont need a voip provider |
10:36.49 | penguinFunk | ah then you dont need a voip provider |
10:37.09 | penguinFunk | just two sip clients both connecting to asterisk |
10:37.16 | c1|freaky | ok very nice ^^ thank you for your help :) |
10:39.54 | c1|freaky | if i run my own server, can it communicate with other servers like f.e. skype servers? |
10:40.03 | shido6 | muahaha |
10:40.15 | shido6 | there are wrappers |
10:40.16 | c1|freaky | just curious |
10:40.17 | shido6 | u can use |
10:40.40 | c1|freaky | ok, thx |
10:40.45 | c1|freaky | y r u laughing? |
10:40.46 | shido6 | yes you can use it with skype (difficulty level 4) 0 ......10 |
10:41.03 | shido6 | 0 = beginner 10 = expert |
10:41.16 | shido6 | if u can read instructions its a 0 |
10:41.27 | nohup- | yes you can use it with skype and fwd and other services |
10:41.30 | c1|freaky | i can ^^ |
10:41.35 | shido6 | but if you read instructions you wouldnt be in here :) |
10:42.18 | c1|freaky | nah i've just heard about it half an hour ago |
10:42.33 | shido6 | well, good morning to you. |
10:42.37 | c1|freaky | and i wanted some information ;D |
10:43.09 | shido6 | theres all kinds of people here. Even really crazy ones |
10:44.04 | c1|freaky | well ive heard from it before, but ... some guy in #jabber asked for a possibility to let his users use software to talk to each other, than i remembered the asterisk plugin for openfire (jabber server) then i became curious and looked at the configuration options of the plugin which confused me, so i went to the asterisk site to find some information |
10:44.08 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
10:44.28 | c1|freaky | as the documentation site with the book is down and so i dont have too much to read i first ask the essential questions for me to even consider running a asterisk server in here ;D |
10:44.49 | McDouglas | i need help sos: i ordered some dlink voip phones |
10:44.53 | McDouglas | and i cant transfer a call |
10:45.01 | McDouglas | is it possible at all with sip phones? |
10:46.13 | ai-a | McDouglas: connected to an asterisk pbx - yes. |
10:46.35 | McDouglas | and if there is no transferbutton on the phone? |
10:46.39 | McDouglas | (cant find any) |
10:46.51 | ai-a | try #<ext># |
10:46.54 | McDouglas | can i configure asterisk to use the * (asterisk) button to initiate a transfer? |
10:47.35 | McDouglas | ai-a: nothing happens |
10:48.00 | c1|freaky | shido6: what do the really crazy people do with asterisk? |
10:50.50 | Uatec | damn |
10:51.46 | ectospasm | c1|freaky: crazy people use stuff like NFAS to share D-channels across multiple PRIs |
10:52.04 | c1|freaky | allright ;D i dont know what that means ;p |
10:52.11 | c1|freaky | NFAS, D-Channels, PRIs |
10:52.19 | ectospasm | in essence giving you extra channels on a T1/E1 |
10:52.33 | ectospasm | Normally, with every PRI line you need to have one D-channel |
10:53.17 | ectospasm | NFAS, or Non-Facilities Associated Signalling, you share a D-channel or two across multiple T1/E1 lines. |
10:54.22 | ai-a | McDouglas... watch... |
10:54.23 | ectospasm | This means that what would normally be used as the D-Channel on some lines can be used as B-channels. B-channel is a voice line, whereas the D-Channel is a signalling channel |
10:54.52 | ectospasm | (I'm taking the DCAP this afternoon) |
10:54.55 | ai-a | McDouglas: what model are they ? |
10:55.04 | McDouglas | dph-300s |
10:55.33 | McDouglas | ai-a: http://www.dlink.hu/?go=jN7uAYLx/oIJaWVUDLYZU93ygJVYLelXSNvhLPG3yV3oWIl5jqltbNlwaaRp7jgkFz2onGQTo48EBtfhzKHkK0gRse3Ya48= |
10:55.48 | ai-a | Russian ? |
10:56.06 | McDouglas | well, yes thats the strange thing... couldnt find any manual in english |
10:56.15 | McDouglas | btw, i'm in Hungary |
10:56.22 | ai-a | im English. |
10:57.05 | ai-a | McDouglas: cant read it.. ask dlink how a sip transfer is done on this phone. |
10:57.20 | McDouglas | i did ask them |
10:57.26 | McDouglas | waiting for their ansfer |
10:57.30 | ai-a | asterisk uses sip, which uses a standard protocol that supports transfer. If the phone doesnt support that, then its not really an asterisk or sip issue. |
10:57.37 | McDouglas | inly problem is.. i already ordered 30 of these phones :\ |
10:57.46 | McDouglas | *o |
10:57.48 | c1|freaky | ectospasm: nice ^^ |
10:57.58 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
10:58.49 | shido6 | if u cannot return them |
10:58.58 | shido6 | then u can use features.conf |
10:59.07 | shido6 | and use like a *# or something |
10:59.13 | ectospasm | yeah, features.conf will allow you to set it up |
10:59.16 | shido6 | or add a speed dial for "Transfer" |
10:59.18 | ai-a | yes, as shido6 said,, enable a feature. |
10:59.27 | shido6 | so dont freak out..... yet |
10:59.45 | JT | still |
10:59.47 | JT | d-link |
10:59.47 | ectospasm | I believe blindxfer is configured to be just # by default |
10:59.48 | JT | eww |
10:59.52 | shido6 | who knows there might be a firmware update for your phone if it uses SIP |
11:00.04 | ai-a | McDouglas: future reference... research and test before you spend your money.. I've been deciding about a new laptop for 3 years so far :) |
11:00.19 | ai-a | ectospasm: he tried #ext, didnt work. |
11:00.48 | ai-a | #ext is for ATA phones, not sip right ? |
11:01.03 | JT | and never buy any d-link voip product, ever |
11:01.20 | McDouglas | well, if i rpess the # key and the extension nubmer all that happens is that i hear the tones in the phone |
11:01.22 | McDouglas | *press |
11:01.34 | ectospasm | ai-a: some ATAs are SIP |
11:01.34 | ai-a | McDouglas: feature not enabled ? |
11:01.44 | McDouglas | how do i enable it? |
11:02.21 | c1|freaky | can users have something like an answering machine on a asterisk server? |
11:02.21 | ai-a | so we saying his dlink is an ata box with a cordless phone ? |
11:02.37 | ectospasm | I would highly suggest you remap the blindxfer to be something other than #, because if someone is in an IVR and they hit # at the end of a digit string, the phone will try to xfer them |
11:02.55 | ectospasm | c1|freaky: called Voicemail, sure |
11:02.56 | ai-a | McDouglas: http://www.voip-info.org/wiki-Asterisk+config+features.conf |
11:03.23 | c1|freaky | cool |
11:03.24 | c1|freaky | :DD |
11:04.27 | McDouglas | ai-a: yes, this is a cordless dict phone with analog port, but it also has a voip port |
11:04.42 | c1|freaky | can they let asterisk speak their own text if they're not online f.e. "im currently not online please leave me a message" ... and can they put something like an own ringtone like modern mobile phone "networks" provide? |
11:04.48 | McDouglas | so if you press the first dial button it uses the analog line, but if you use the 2nd dial button you dial on voio |
11:04.49 | ai-a | McDouglas: hence sip protocol isnt working. |
11:05.05 | McDouglas | well, i could register the sip account |
11:05.17 | McDouglas | and call other sip hpnes |
11:05.32 | ai-a | yes, [asterisk] --- (sip) --- [ata device] -- analogu -- [dict phone] |
11:05.41 | ai-a | your dict phone isnt sip,, its the base unit. |
11:06.19 | ai-a | however our corless phones have a 'recall' button that support the sip transfer vai the ata boxes. |
11:06.23 | ai-a | *cordless. |
11:06.26 | ectospasm | c1|freaky: yeah, there's unavailable and busy messages |
11:07.00 | ectospasm | You can have distinctive rings, but I'm pretty sure they're governed by the phones in question, and asterisk doesn't have anything to do with ringtones |
11:07.12 | c1|freaky | ok ;) |
11:07.23 | c1|freaky | thank you :) |
11:08.56 | McDouglas | ai-a: i put "blindxfer => #" in features.conf |
11:08.59 | McDouglas | but nothing happens |
11:09.10 | ai-a | reset asterisk ? |
11:09.11 | McDouglas | how can i check if it actually works? |
11:09.16 | McDouglas | i restarted it |
11:09.24 | ai-a | press # :) |
11:09.36 | McDouglas | i can hear the dtmf sound, nothing else |
11:09.40 | ai-a | McDouglas: get a softphone, and check it works on the computer. |
11:09.48 | ectospasm | Are you bridged to anything you can transfer? |
11:10.02 | ectospasm | You can't transfer if all you've got is a dial tone, there's nothing to transfer |
11:10.15 | McDouglas | i call phone B from phone A |
11:10.22 | nohup- | try this: just dial the extension number and then press # ? |
11:10.25 | McDouglas | and try transfering from phone B to phone C |
11:10.36 | ectospasm | In the Dial string, do you have t or T as an option? |
11:10.43 | McDouglas | i have t |
11:11.00 | ectospasm | Who's trying to do the transfer? |
11:11.07 | McDouglas | phone b |
11:11.18 | ectospasm | ah, then you probably need T as well |
11:11.34 | ectospasm | IIRC, t means A can transfer, T means B can transfer |
11:11.51 | McDouglas | lemme try it |
11:15.16 | McDouglas | ai-a: i installed x-lite |
11:15.30 | McDouglas | if i call that extension and press X in xlite |
11:15.34 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3062ffe84f40f3ec) |
11:15.41 | McDouglas | asterisk says "transfer" and i can transfer |
11:15.49 | ai-a | what about # |
11:15.50 | McDouglas | but if i call a dlink extension |
11:15.58 | kaldemar | t is for called party, T is for calling party. |
11:15.59 | McDouglas | err, i meant # |
11:16.02 | McDouglas | sry |
11:16.13 | McDouglas | so, if i call the dlink and press the # on it |
11:16.16 | McDouglas | nothing happens |
11:16.20 | ai-a | ok, so transfer works.. just your phone doesnt allow it.. |
11:18.03 | ai-a | McDouglas: check the docs for the device... must be some way of doing it. Transfer is a high importance of a phone on a PBX. unless they just expected you to use the phone as a home phone connected to a sip service for calls. |
11:18.18 | McDouglas | lol, there was NO documents in the box |
11:18.25 | McDouglas | hunted one from some russian ftp server |
11:18.31 | McDouglas | and it says nothing about call transfering :\ |
11:18.53 | JT | send them back |
11:20.48 | ai-a | McDouglas: how much did you pay for each phone ? |
11:21.11 | McDouglas | $50 |
11:21.34 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
11:21.55 | ai-a | us$ ? |
11:22.35 | McDouglas | yes |
11:22.45 | ai-a | well, they are cheap. what do you expect ? |
11:22.58 | McDouglas | they are not cheap |
11:23.07 | McDouglas | this is 1/3 of the normal price |
11:23.12 | JT | they are |
11:23.17 | JT | and they are still a waste of money |
11:23.19 | McDouglas | because we got some demo prices |
11:23.22 | ai-a | 1/3 of normal price = cheaper than normal price. |
11:23.23 | JT | $85 buys you a polycom |
11:23.27 | McDouglas | being a reseller |
11:24.00 | McDouglas | ai-a: i meant, if you wanted to buy this phone froma retail you had to pay $150 |
11:24.13 | JT | what a rip off |
11:24.17 | JT | sounds like a pile of junk |
11:25.41 | *** join/#asterisk sniper[FOO] (i=Snip3r@217.27.214.111) |
11:25.53 | sniper[FOO] | hi there |
11:26.12 | sniper[FOO] | anyone having a couple of minutes to help me out? |
11:26.21 | JT | ~question |
11:26.22 | jbot | hmm... question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
11:26.22 | ectospasm | sniper[FOO]: don't ask to ask, just ask |
11:26.26 | JT | ~ask |
11:26.27 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
11:26.41 | Wonka | why not have transfer handled by the asterisk the phone is registered to? |
11:26.50 | sniper[FOO] | OK, got it :) |
11:26.54 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582526.dsl.bell.ca) |
11:27.09 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
11:28.46 | sniper[FOO] | my main concern is that my application has to detect a ringing tone in the media stream (SIP/g.711) and I ended up with no acceptable results using AMD and BackgroundDetect, both are giving a huge number of false positives |
11:29.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
11:29.41 | c1|freaky | what are asterisk extensions? can i find a list somewhere? |
11:29.44 | sniper[FOO] | the point is to look for a regular ringtone in an already connected audio stream |
11:30.00 | ai-a | c1|freaky: you mean addons ? |
11:30.03 | sniper[FOO] | and do something if it succeeds |
11:30.09 | c1|freaky | ai-a: yea, also ... |
11:30.20 | ai-a | extensions would refer to a phone extension :) |
11:30.21 | c1|freaky | i just want to find out more about the possibilities with asterisk |
11:31.10 | sniper[FOO] | I found an app called nvlinedetect |
11:31.42 | sniper[FOO] | but I gotta use the 1.4 branch and I saw several posts reporting it won't compile |
11:31.47 | ai-a | c1|freaky: http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons |
11:31.56 | c1|freaky | thank you :) |
11:32.30 | ai-a | c1|freaky: or... http://www.voip-info.org/wiki/view/Asterisk+addons |
11:32.36 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
11:32.50 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
11:33.02 | ectospasm | ~thebook |
11:33.02 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
11:33.31 | ectospasm | yeah, asteriskdocs appears to be down |
11:33.31 | ai-a | ectospasm: url failed :) |
11:33.32 | c1|freaky | thanks |
11:33.38 | JT | sniper[FOO]: wow, detecting tones, sounds like you like to get frustrated |
11:33.59 | sniper[FOO] | JT: I'm already frustrated :) |
11:34.14 | JT | give up? |
11:34.17 | sniper[FOO] | nope |
11:34.30 | JT | why are you detecting tones? |
11:34.36 | ectospasm | http://209.85.165.104/search?q=cache:X5x-an-2KkcJ:www.asteriskdocs.org/modules/tinycontent/index.php%3Fid%3D11+asterisk+the+future+of+telephony+mirrors&hl=en&ct=clnk&cd=1&gl=us |
11:34.38 | ai-a | isnt there a Ring cadence detection ? |
11:34.49 | sniper[FOO] | if I can't find a suitable solution in a couple of days, I'll write one |
11:34.53 | ectospasm | Some of those mirrors might still work. Yay Google Cache! |
11:34.55 | ai-a | guess that cant detect tones ? |
11:35.07 | ai-a | ectospasm ;) |
11:35.10 | JT | i have the book mirrored |
11:35.23 | sniper[FOO] | It's not that difficult to detect a known waveform in HQ PCM audio |
11:35.44 | JT | http://210.14.110.96/~jon/asterisktfot.zip |
11:35.52 | JT | it is in asterisk |
11:35.53 | sniper[FOO] | but I hoped I won't have to get into * inside out |
11:35.54 | ai-a | i downloaded a copy of the Asterisk book and now its used as a door stopper for my virtual house. |
11:36.16 | JT | and it depends how clean you expect the waveform to be |
11:36.33 | ai-a | sniper[FOO]: detecting tones,,, when, and why ? |
11:36.56 | sniper[FOO] | JT: the purpose is to OK the INVITE on a dumb gateway in the right time |
11:37.29 | *** join/#asterisk klictel (n=klictel@atelka.info) |
11:37.38 | sniper[FOO] | eg. set the status to Ringing till ringtone comes |
11:37.58 | JT | what sort of piece of junk gateway is this |
11:38.04 | JT | doesn't it have sip messages? |
11:38.04 | sniper[FOO] | and screen out the providers' message |
11:38.11 | JT | oh right |
11:38.14 | JT | toll evasion |
11:38.16 | JT | gsm gateway |
11:38.27 | sniper[FOO] | indeed |
11:38.40 | sniper[FOO] | toll evasion? |
11:38.41 | JT | not a fan of gsm gateways |
11:39.00 | JT | it's at least evading terminating your calls properly :) |
11:39.04 | sniper[FOO] | this process is called toll evasion? |
11:39.10 | sniper[FOO] | sure :) |
11:39.24 | JT | high density gsm gateways are a bad idea |
11:39.35 | ai-a | toll evasion doesnt sound legal. |
11:39.46 | sniper[FOO] | I'm pretty familiar with the issues :) |
11:40.04 | JT | sucky audio |
11:40.09 | JT | lack of cell capacity |
11:40.20 | sniper[FOO] | anyway, it's not my business, a friend runs the business and asked me to help |
11:40.51 | sniper[FOO] | audio is really decent |
11:41.07 | JT | not compared to pri |
11:41.13 | JT | and transients can screw it up |
11:41.15 | sniper[FOO] | yep, right |
11:41.23 | *** part/#asterisk nohup- (i=hmmmph@203.81.206.134) |
11:41.31 | sniper[FOO] | but it's somewhere near |
11:42.35 | sniper[FOO] | anyway, do you have a clue how to accomplish this? |
11:43.10 | JT | write c, or don't use asterisk :) |
11:44.56 | sniper[FOO] | OK, then I'm gonna tweak the nvlinedetect source to get it compile in the 1.4 branch |
11:45.15 | sniper[FOO] | ...is it something people don't really want to share? |
11:45.32 | JT | no, it's just something people really don't do |
11:46.13 | ai-a | sniper[FOO]: tried #phonefreaking |
11:46.20 | JT | what exactly does nvlinedetect detect? |
11:47.19 | sniper[FOO] | tones (ringing, congestion, busy/ignore, anything) |
11:47.37 | sniper[FOO] | nvfaxdetect does the same for fax tones |
11:47.38 | JT | you can specify tones to detect? |
11:47.56 | sniper[FOO] | BT ringing tone |
11:48.01 | sniper[FOO] | sure I can |
11:48.16 | *** join/#asterisk michael-i (n=michael-@Wb85d.w.pppool.de) |
11:48.18 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
11:48.47 | JT | freeswitch can detect tones you specify out of the box |
11:48.59 | sniper[FOO] | freeswitch? |
11:49.06 | sniper[FOO] | some * distribution? |
11:49.07 | JT | google :) |
11:49.09 | JT | no |
11:49.10 | sniper[FOO] | IC |
11:49.12 | JT | nothing to do with * |
11:49.15 | sniper[FOO] | lame question |
11:49.38 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
11:51.44 | michael-i | Does anyone have any tips to prevent the first 100-200ms of my outgoing voicemail greeting from being cutoff? Calling in on a zaptel channel and reaching voicemail cuts a bit off. |
11:56.47 | deegan | do a Wait(2) first? |
11:57.01 | dan__t | hrm hrm hrm.... ALMOST have inbound calling working |
11:59.06 | michael-i | ah yes, had a wait(1) in there but not in the right spot... (INSIDE the macro would be better) |
11:59.15 | michael-i | thanks for the jumpstart |
12:04.00 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
12:04.55 | cjk | hi, since a certain time i have problems to play wav49 files from my voicemail system under linux and windows. when i convert them using sox everything is ok. how do you play such files? |
12:06.42 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
12:07.52 | dan__t | Alright, I got incoming IAX2 calls to work, kindof |
12:08.01 | dan__t | I think my.. uh... dialplan is hosed |
12:11.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
12:11.30 | drako | morning |
12:11.50 | *** join/#asterisk klictel (n=klictel@atelka.info) |
12:13.45 | dan__t | sup, dood. |
12:13.50 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
12:16.19 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
12:16.58 | dan__t | This is so bad-ass. |
12:17.45 | shido6 | must be on the right track, dan__t :) |
12:17.46 | dan__t | Ok, so, it looks like the documentation over at Teliax suggests that I can use the phone number as the name or number of the extension |
12:17.52 | dan__t | Getting there man, I really am! |
12:18.22 | dan__t | I mean their examples for adding an extension included the actual phone number as the extension name |
12:18.25 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
12:18.43 | dan__t | So I'd just keep using that "extension name" (really, the phone number) and build my dialplan out that way? |
12:19.37 | McDouglas | ai-a: i got it working |
12:19.54 | McDouglas | the dtmf signaling was at fault |
12:20.13 | ai-a | settings on the ata device ? |
12:20.24 | McDouglas | yes |
12:20.30 | McDouglas | it was set to inband |
12:20.31 | ai-a | thats $50 * 30 saved in your pocket now :) |
12:20.35 | McDouglas | had to change it to rfc |
12:20.41 | McDouglas | lol, yes ;) |
12:21.08 | [TK]D-Fender | dan__t: you will typically receive calls against an exten registered with your ITSP. Once it arrives on that # you can then simply GOTO wherever else you want in your dialplan to actually begin processing the call. |
12:21.14 | *** join/#asterisk coppice (n=chatzill@234.155.17.210.dyn.pacific.net.hk) |
12:22.17 | *** join/#asterisk snk00sj (n=gnelisse@apollo.digitalbase.be) |
12:22.19 | snk00sj | hi all |
12:22.24 | dan__t | oh ok. |
12:22.33 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
12:22.37 | dan__t | Very cool. |
12:22.51 | snk00sj | i am using asterisk 1.4.11 for the first time, i created a SIP user, and now want to connect to phone to it |
12:23.10 | snk00sj | after setting auth userid & pw, the asterisk log gives "no matching peer found" |
12:23.28 | snk00sj | i am using the gui (web interface) to change settings |
12:29.23 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
12:30.49 | dan__t | Anyone know of a .gsm format player for X, so I can mow through a few of these recordings? |
12:31.35 | dan__t | heh, loading one just segfaulted audacious |
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12:32.56 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
12:33.34 | *** join/#asterisk jhiver_ (i=jhiver@164-242.206-83.static-ip.oleane.fr) |
12:33.58 | jhiver_ | hi all, do you know how I could convert .wav files into .g729 to avoid transcoding when doing music on hold? |
12:35.06 | jhiver_ | http://www.asteriskguru.com/audio_conversion.php thank you voip-info.org =) |
12:35.27 | *** join/#asterisk minkus (n=minkus@pool-72-84-53-31.clrkwv.east.verizon.net) |
12:36.20 | [TK]D-Fender | snk00sj: This is not the GUI support channel. |
12:36.47 | snk00sj | [TK]D-Fender, i know, i just don't think it has anything todo with the gui |
12:37.16 | [TK]D-Fender | snk00sj: Of course you don't, that would imply some sort of personal responsibility...... |
12:37.32 | [TK]D-Fender | snk00sj: but you see until you can show us something solid, its jsut you... |
12:37.45 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
12:37.47 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
12:37.47 | *** join/#asterisk PepOSX (n=pepOSX@190.72.145.178) |
12:37.54 | lirakis | morning |
12:37.54 | [TK]D-Fender | snk00sj: PASTEBIN is your friend... |
12:37.56 | [TK]D-Fender | ~pb |
12:37.56 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
12:38.45 | [TK]D-Fender | snk00sj: So go show us what you've got configured for that SIP device and lets see how * reacts with SIP debug enabled |
12:43.42 | *** part/#asterisk Strom_M (n=strom@216.64.24.250) |
12:43.57 | arekm | if I drop E1 part of that config and leave Zhone part then asterisk starts and zhone works fine |
12:45.29 | arekm | http://pastebin.com/m7293149e - added zapata.conf here |
12:45.31 | arekm | any ideas? |
12:45.42 | arekm | (I mean zaptel.conf) |
12:47.59 | *** join/#asterisk ManxPower (n=manxpowe@38.sub-75-203-142.myvzw.com) |
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12:53.55 | [TK]D-Fender | arekm: I don't think I've ever heard of E! & T1 mixed on the same card. I recall there being a jumper you had to set to use E1. To me that might devalidate the mix. Maybe somebody can correct me on this... |
12:54.15 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.132.243) |
12:54.23 | coppice | you can mix E1 and T1 on many boards |
12:55.00 | [TK]D-Fender | coppice: I should have specified my assumption of Digium there.. |
12:55.15 | coppice | you can mix on digium boards |
12:55.19 | arekm | Tormenta 2 (PCI) Quad T1 Card 0 Span 1 |
12:55.19 | arekm | Tormenta 2 (PCI) Quad T1 Card 0 Span 2 |
12:55.19 | arekm | Tormenta 2 (PCI) Quad E1 Card 0 Span 3 |
12:55.19 | arekm | Tormenta 2 (PCI) Quad E1 Card 0 Span 4 |
12:55.27 | arekm | to 2xT1 and 2xE1 here |
12:55.33 | coppice | expect that one :-) |
12:55.40 | *** join/#asterisk Corydon76-dig (i=green@pdpc/supporter/sustaining/Corydon76-home) |
12:55.40 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
12:55.42 | ManxPower | Learn to use pastebin.ca |
12:55.54 | [TK]D-Fender | thats 4 lines... I wouldn't panic... |
12:56.06 | arekm | ManxPower: learn to read (earlier ;) |
12:56.07 | coppice | the tormenta 2 card has different chips fitted for T1 oe E1 operation |
12:56.43 | file | [TK]D-Fender: !?!!!! |
12:56.53 | [TK]D-Fender | file: !?!??! |
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13:03.16 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
13:09.56 | *** join/#asterisk saftsack (n=saftsack@pD9E042FB.dip.t-dialin.net) |
13:13.24 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:19.11 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
13:21.53 | henkoegema | <PROTECTED> |
13:25.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:32.40 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
13:32.44 | kippi | hey |
13:32.49 | [TK]D-Fender | ho |
13:33.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
13:33.17 | *** mode/#asterisk [+o anthm] by ChanServ |
13:33.43 | kippi | really simple question, I have this config http://www.pastebin.ca/697508 but when the line is busy it goes to voicemail, how can get it to go to 5 then 6 etc? |
13:36.36 | [TK]D-Fender | kippi: the line isn't busy as far as # can see. it got answered by the telco |
13:37.03 | [TK]D-Fender | * |
13:37.20 | kippi | what happends is, asterisk puts it to the voicemail because the handset is already busy |
13:38.00 | ai-a | your not using your return 'r' |
13:38.04 | [TK]D-Fender | oops.... ok, 5 should get called. Show me a call where it isn't |
13:38.16 | ai-a | oh ignore me. :)_ |
13:40.31 | *** join/#asterisk lucky7 (n=Adam@207.200.28.175) |
13:40.55 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
13:41.24 | kippi | how do you mean called? |
13:42.17 | ai-a | kippi: CLi log. |
13:42.38 | ai-a | prove 5 & 6 is not being called by showing us the log of the call. |
13:42.59 | kippi | ok |
13:43.08 | ai-a | (side note, dont paste in here, use http://pastebin.ca ) |
13:43.24 | ManxPower | kippi: voicemail IS priority 5 |
13:43.38 | ManxPower | sorry, I misread that. |
13:43.46 | ManxPower | kippi: it should work as you expect. |
13:43.47 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-a4695c49f2ee5a82) |
13:44.05 | ManxPower | kippi: what version of Asterisk are you using? |
13:44.10 | kippi | http://www.pastebin.ca/697520 |
13:44.30 | kippi | 1.2.20 |
13:44.33 | ManxPower | kippi: the phone IS NOT BUSY! |
13:44.49 | ManxPower | The phone is forwarded to extension 1050 |
13:44.54 | ManxPower | turn off call forwarding on the phone |
13:45.00 | kippi | ah ha |
13:45.18 | ManxPower | also remove the "r" from the Dial line. It makes you look like an asterisk retard |
13:45.49 | ai-a | learn to read your logs. |
13:46.04 | ManxPower | kippi: the phone MAY be configured to FORWARD calls to a voicemail extension you have configured for it when the line is busy. Don't know. |
13:46.08 | ManxPower | But it is not a dialplan issue. |
13:46.08 | *** join/#asterisk Penggu (n=me@203-213-102-59-nme-ts7-2600.tpgi.com.au) |
13:46.52 | Penggu | hi all. is there a 'camp' feature? eg you call someone, so you 'camp' on their phone.. then hang up. as soon as the person gets off the phone, your phone rings, you pick up, and it calls that phone |
13:47.18 | Penggu | sorry i mised putting in there they the called party is initially busy on another call |
13:47.22 | ManxPower | Penggu: you can write one in the dialplan. |
13:48.10 | ManxPower | "show applications like dial" |
13:48.13 | *** part/#asterisk pointer (n=pointer@aj.catt.com) |
13:48.15 | ManxPower | notice the retry version of Dial |
13:48.33 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-12bdf5e231dc4a8e) |
13:49.00 | ManxPower | So if you Dial a device and it is busy (as determed by the value of DIALSTATUS or HANGUPCAUSE) then you can use the version of Dial that retries. |
13:49.01 | Penggu | ta, i;ll look into it |
13:49.14 | Penggu | i c |
13:49.20 | Penggu | so the called party can hang up in the meantime? |
13:49.23 | ai-a | but hes hanging up on the call, it needs bind transfering to a holding ext, then bringing back when the calling party is free. |
13:49.35 | ManxPower | I dunno. read the docs for the dial retry |
13:50.03 | Penggu | ai-a: you mean call-parking style? |
13:50.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:50.39 | ai-a | yep |
13:50.41 | ai-a | park the call. |
13:51.03 | Penggu | hmm |
13:52.13 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
13:54.02 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
13:55.26 | Penggu | i guess for the 'camping' that i described above, it'd probably be more desirable to use chanisavail() rather then dail()ing the extension, to avoid annoying the person too much |
13:56.31 | Penggu | could may be have an h extension to allow the hanging up to check for registred campers... and trigger a call back |
13:56.42 | Penggu | would need somewhere to store campers |
13:58.19 | ai-a | Penggu: http://www.voip-info.org/wiki/view/Asterisk+call+parking |
14:02.40 | *** join/#asterisk yangvnc (i=yang@static-ip-62-75-255-125.inaddr.intergenia.de) |
14:02.58 | Zeeek | [NEWS] in 90 minutes, there's the #asterisk-users-conference on the IRC channel of that name and info here: http://VoipUsersConference.org |
14:04.29 | [TK]D-Fender | yangvnc: This isn't #Asterix |
14:05.02 | elixer | or #Asstricks |
14:05.08 | elixer | thats fun |
14:05.11 | [TK]D-Fender | kippi: if the phone is forwarded like that you're screwed |
14:05.33 | [TK]D-Fender | elixer: no, thats supposed tobe "That's hawt" |
14:05.51 | ai-a | yangvnc: oh, that was a great comic. |
14:05.52 | elixer | heh |
14:07.08 | Zeeek | Robin Williams is a great comic |
14:08.33 | yangvnc | It was ment as joke |
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14:18.11 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
14:18.55 | elriah | Hi all. How stable is the Asterisk Manager interface in 1.2.x, I have an app that's going to poll every 3 seconds but that app may be in the hands of say 50 different customers. Am I shooting myself in the foot doing this? |
14:21.30 | dijungal | http://asteriskdocs.com |
14:21.42 | elriah | dijungal: Is that to me? |
14:21.52 | dijungal | oops sorry..... |
14:22.03 | dijungal | what's wrong with http://asteriskdocs.com??? |
14:23.05 | lirakis | dijungal: .. not sure.. but its been down for a while ( a few weeks at least i think ) |
14:23.09 | Nugget | Just a guess, but I don't think the URL is supposed to have three question marks at the end. |
14:23.14 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.148.235) |
14:23.27 | dijungal | ohooo |
14:23.36 | Qwell | Nugget: RFC-8457 allows it |
14:24.32 | [TK]D-Fender | elriah: What are they polling for? |
14:24.35 | Nugget | heh |
14:25.33 | elriah | [TK]D-Fender: Calls (Status) and Peers (SIPpeers) and possible Queue info... In a Flex 2 based operator console hitting a php back end... |
14:25.44 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
14:26.15 | elriah | The app works great, but before I release it I need to test under load and do some research.. this channel is the first step.. |
14:26.19 | [TK]D-Fender | elriah: What I'd suggest : Make a SINGLE polling app that will retrieve & do the initial parsing and have it get picked up from THAT server. |
14:27.06 | elriah | [TK]D-Fender: Ahh... Such as you suggested yesterday, a daemon... Have my app poll every few seconds and write out XML files.. yep, that's works and should be easy to do on my end.. |
14:27.23 | elriah | [TK]D-Fender: Where do I send your consulting fee, lol |
14:27.52 | ManxPower | elriah: you send it to eric@fnords.org via Paypal |
14:28.43 | [TK]D-Fender | Hide and Zeeek! |
14:28.59 | elriah | [TK]D-Fender: lol!!!! |
14:29.43 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
14:29.46 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
14:30.38 | [TK]D-Fender | elriah: For my Polycom MicroBrowser Idle live queue status I sued to have 5 phones polling every 10s with 2 access each. my CLI would get spammed even though it wasn't a huge load. Instead I made a master poller taht created a STATIC XHTML page which would get reloaded instead of creating extra AMI calls. |
14:31.02 | elriah | ManxPower: Money sent. |
14:31.03 | [TK]D-Fender | elriah: While this adds a few seconds of net lag to the reporting "currentness" its more than acceptable |
14:33.13 | [TK]D-Fender | elriah: if you need something a little more live and want to reduce the frequency a bit you could also try the AMI Proxy so that its a single persistant connection. |
14:33.52 | ManxPower | elriah: Uh, I was joking. Also $1 is not a consulting fee, it is an insult. |
14:34.40 | elriah | ManxPower: lol, sorry couldn't resist. I'll get you next time. |
14:35.00 | shido6 | wow |
14:35.34 | kippi | thanks guys |
14:35.36 | kippi | that worked |
14:35.50 | *** join/#asterisk juliux (n=juliux@ubuntu/member/juliux) |
14:35.55 | outtolunc | i want $1 where do i get one |
14:36.14 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
14:36.41 | juliux | hi all, i setup a new client and now i get this error dsp.c:1426 ast_dsp_process: Inband DTMF is not supported on codec gsm. Use RFC2833 all the time in my asterisk log, what did i wrong? |
14:37.09 | ManxPower | elriah: you are the first person to send me money is at least 6 months |
14:37.28 | ManxPower | juliux: what you did wrong is configure inband dtmf when using GSM |
14:37.34 | ManxPower | don't do that. it won't work |
14:38.06 | elriah | Seriously, this channel is incredibly helpful and open source doesn't = free. I wish I could give back more and will as our products and services are successful. |
14:38.46 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
14:39.16 | [TK]D-Fender | elriah: Share code where you can as well then. |
14:39.27 | *** join/#asterisk ChrisN (i=ccn@72.46.131.18) |
14:39.38 | [TK]D-Fender | elriah: Public document "how-to's" for the tricky stuff you got to work. |
14:39.44 | [TK]D-Fender | ~sipnat |
14:39.45 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
14:39.54 | [TK]D-Fender | ^^^ thats one of my contributions |
14:40.24 | [TK]D-Fender | elriah: Alons with countless JBOT trainings. |
14:40.55 | ChrisN | Any comments on how to fix this error from Asterisk 1.2.13? channel.c: Avoided initial deadlock for '0x8121d00', 10 retries! |
14:42.11 | ManxPower | ChrisN: upgrade helps with many deadlock issues. |
14:42.25 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
14:43.23 | ChrisN | I wonder why Debian is still using 1.2.13 in their provided asterisk package. |
14:43.25 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
14:43.30 | ManxPower | Using Monitor, MixMonitor, ChanSpy, and ZapScan can make this errors happen much more often. |
14:43.33 | juliux | ManxPower, thxs |
14:43.41 | ManxPower | ChrisN: We don't care. Compile from source. |
14:43.58 | [TK]D-Fender | ChrisN: Because glaciers are stable too.... onlyl thats too fast for Debian's liking ;) |
14:44.15 | Wonka | debian has 1.4.11 already... |
14:44.21 | Zeeek | here we go... |
14:45.17 | CaT[tm] | chrisn: if you don't want to compile it yourself check with beckports.org or apt-get.org to see if anyone else has done a newer version and made it available already |
14:45.42 | ChrisN | Cool. Thanks for the help. |
14:45.44 | *** part/#asterisk ChrisN (i=ccn@72.46.131.18) |
14:46.10 | ManxPower | the problem with using packages is that most of the packagers of Asterisk, Zaptel, libPRI have no idea what the requirements of the various pieces of software and how they interact. |
14:46.18 | jcanfield | What is the secret to make calls ring down to the next line on a Polycom 550? |
14:46.21 | ManxPower | And more importantly, WE have no idea how it is packages. |
14:46.27 | ManxPower | jcanfield: there are at least two. |
14:47.10 | ManxPower | The one *I* use is that each line appearance is registered to Asterisk as a seperate SIP account, then I check the return value of Dial (using DIALSTATUS) and decide where to route the call based on that. |
14:47.20 | CaT[tm] | manx: well it might work out for him or it might not. if not he can always try and compile it himself. |
14:47.30 | ManxPower | You also have to set the max calls per line appearance in the sip.cfg or phone1.cfg config file for the polycoms |
14:49.23 | jcanfield | ManxPower, hmmm, so the polycom can't manage the line traffic? I was hoping it would work like the softphones; Setup one line and it rings down automagically . |
14:49.43 | mattboll | does anyone know something about create_addr: No such host: free/0247503054 ? |
14:49.51 | ManxPower | I didn't say you can't do that, I said this is how *I* do it, as it give me TOTAL control of how calls are routed on the phone. |
14:50.00 | [TK]D-Fender | jcanfield: setup reg1 to use X calls @ 1 call per line-key |
14:50.11 | ManxPower | mattboll: Your Dial line is screwed up. |
14:50.12 | [TK]D-Fender | jcanfield: And its will ring down naturally |
14:50.38 | [TK]D-Fender | mattboll: pastebin your failed call attempt |
14:50.40 | [TK]D-Fender | ~pb |
14:50.41 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
14:51.51 | jcanfield | [TK]D-Fender, I'll check it out. Forgive me for being the FNG, but is reg1 a phone setting? |
14:51.57 | ManxPower | mattboll: chances are you have Dial(SIP/free/0247503054) instead of Dial(SIP/0247503054@free) |
14:51.57 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
14:52.08 | dijungal | i know call-limit is buggy in *1.2, has it been fixed in 1.4? |
14:52.18 | mattboll | http://pastebin.com/d302d7014 |
14:52.25 | ManxPower | And I am assuming you have a [free] section/peer/user/friend in sip.conf |
14:52.26 | [TK]D-Fender | jcanfield: By what means have you configured your phone? |
14:52.39 | ManxPower | dijungal: call limit has been buggy since day 1 |
14:52.53 | linagee | ManxPower: call limit? |
14:52.53 | dijungal | k |
14:53.02 | jcanfield | [TK]D-Fender, web ui only so far....jsut started my dive into all this. |
14:53.04 | [TK]D-Fender | mattboll: Executing Dial("SIP/bmatthieu-0817f328", "SIP/0247503054@free/0247503054|300|TtW") <- indeed this is not a valid format |
14:53.09 | dijungal | is the agent module more stable in *1.4 ? |
14:53.22 | mattboll | ManxPower: ManxPower yes I have |
14:53.41 | ManxPower | dijungal: last I heard the calllimit is not reset when an agent transfers a call, until the call is hungup. |
14:53.51 | [TK]D-Fender | mattboll: What GUI created that dialplan? |
14:54.01 | mattboll | [TK]D-Fender: everything is configured with destar, I'll tell them ^^ |
14:54.18 | [TK]D-Fender | mattboll: Verify how you filled in the blanks when configuring it. |
14:54.19 | dijungal | k |
14:54.25 | ManxPower | mattboll: we can't help you with GUIs. I already told you how your Dial line should be. |
14:54.30 | dijungal | ManxPower: and the work around is? |
14:54.32 | [TK]D-Fender | mattboll: Dial("SIP/bmatthieu-0817f328", "SIP/free/0247503054|300|TtW") <- this should be valid. |
14:54.50 | [TK]D-Fender | mattSee if the way you filled things in was the source and not the GUI itself. |
14:54.51 | mattboll | ok thanks a lot |
14:54.54 | ManxPower | dijungal: no idea. I hate queues and don't normally use them. I simulate simple queues using dialplan logic |
14:55.40 | ManxPower | I had massive problems with queues - most of them were USER issues, but I had a few Asterisk issues too. |
14:55.47 | [TK]D-Fender | dijungal: By & large works fine. Was that an open-ended question or do you have an actual issue? |
14:56.05 | ManxPower | [TK]D-Fender: he has the classic call-limit=1 problem when using queues. |
14:56.14 | ManxPower | and agents transfer calls |
14:56.29 | ManxPower | I have no idea if/when that was fixed. |
14:57.06 | [TK]D-Fender | ManxPower: I've seen the one where app_queue thinks the agent is busy, I've never head of chan_sip's counter being off because of it though... |
14:59.25 | ManxPower | dijungal: describe to [TK]D-Fender what your issue is with queue and calllimit |
14:59.41 | Uatec | hello, is there any way of making a queue wait 30seconds or so after an agent becomes available before their phone rings again? |
15:00.06 | errr | does _. match any incoming number? |
15:00.36 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
15:00.38 | [TK]D-Fender | Uatec: thats what wrapuptime is for |
15:00.48 | [TK]D-Fender | errr: Dangerously so. |
15:01.00 | errr | [TK]D-Fender: thanks |
15:01.02 | [TK]D-Fender | errr: _X. is safer |
15:01.16 | errr | ok |
15:01.22 | [TK]D-Fender | errr: but restricts you to 2+ digit #'s |
15:01.31 | [TK]D-Fender | errr: actually I think _X! would be best |
15:02.02 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
15:02.06 | ZaVoid | hey guys |
15:02.07 | errr | [TK]D-Fender: what is the _ for.. does that mean its going to be an incoming call? |
15:02.24 | [TK]D-Fender | errr: No it means that what follows is a PATTERN MATCH. |
15:02.34 | errr | ah |
15:02.40 | ZaVoid | can someone point me to a spot in the wiki or somehwere else where i can understand how to break out numbers in a dialplan from a variable? |
15:02.47 | *** join/#asterisk zeppelin_ (n=zeppelin@201-66-150-155.paemt700.dsl.brasiltelecom.net.br) |
15:02.47 | errr | [TK]D-Fender: thanks |
15:02.48 | [TK]D-Fender | errr: You really should have know that part..... its beyond dialplan 101 |
15:02.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
15:03.09 | errr | :( |
15:03.19 | [TK]D-Fender | ZaVoid: lookup "asterisk variables" on the WIKI and it will show all sorts of ways |
15:03.43 | [TK]D-Fender | ZaVoid: And after review "show function CUT" for some funkier stuff |
15:03.44 | ZaVoid | tyeah i was reading http://www.voip-info.org/wiki/index.php?page=Asterisk+variables |
15:03.44 | errr | [TK]D-Fender: looks like I need to brush up on http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
15:03.53 | ZaVoid | but it wasn't helping |
15:04.19 | [TK]D-Fender | errr: not knowing that "_" at the start INDICATES that what follows is a pattern is jsut a little disturbing :) |
15:04.29 | ZaVoid | i got variable called bignumber.. and "345" is in the variable... exten => s,n,SayDigits(${bignumber}) and i want to say 3 first.. then play a sound file.. then 45 |
15:04.36 | ZaVoid | let me search on the show function cut |
15:04.43 | [TK]D-Fender | ZaVoid: Be specific with your example of what you'd like to accomplish then. |
15:05.23 | [TK]D-Fender | ZaVoid: Saydigits(${bignumber:0:1}) |
15:05.29 | [TK]D-Fender | ZaVoid: Saydigits(something) |
15:05.38 | [TK]D-Fender | ZaVoid: Saydigits(${bignumber:1}) |
15:05.48 | ZaVoid | ahhhh |
15:05.49 | ZaVoid | i see |
15:05.52 | [TK]D-Fender | ZaVoid: And yes, that WAS all in the first link |
15:06.00 | ZaVoid | my link or yours? |
15:06.10 | [TK]D-Fender | ZaVoid: yours |
15:06.16 | ZaVoid | err ok let me go look again |
15:06.46 | [TK]D-Fender | ZaVoid: Substrings ${foo:offset:length} |
15:06.56 | ZaVoid | substring i see |
15:07.00 | ZaVoid | thanks man |
15:07.03 | [TK]D-Fender | Returns a substring of the string foo, beginning at offset offset and returning the next length characters. |
15:07.05 | [TK]D-Fender | <PROTECTED> |
15:07.06 | [TK]D-Fender | <PROTECTED> |
15:07.08 | ZaVoid | i won't ask that ever again :) |
15:07.22 | [TK]D-Fender | ZaVoid: s'ok |
15:07.37 | [TK]D-Fender | ZaVoid: At least it wasn't something we had to beat into your head :) |
15:07.59 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
15:08.01 | ZaVoid | lol |
15:08.02 | ZaVoid | thanks man |
15:08.30 | Dr-Linux | is there any link from where i can see/listen MP3 ads? |
15:08.32 | dlynes | [TK]D-Fender: I got that parking feature so that it rings multiple phones when it coms off of park done much easier |
15:08.38 | *** join/#asterisk melbert (n=IceChat7@66.179.79.70) |
15:08.41 | dlynes | [TK]D-Fender: app_valetparking |
15:08.41 | Dr-Linux | actually i wanna download one |
15:09.58 | [TK]D-Fender | dlynes : Funny, my way was MUCH easier ;) |
15:10.12 | dlynes | [TK]D-Fender: your way to not do it at all? |
15:10.24 | dlynes | [TK]D-Fender: that's not a workable solution, though |
15:10.28 | [TK]D-Fender | dlynesthat compiling that addon... but if you need to actually PICK IT UP, versus leaving them in limbo a bit, yeah ValetParking rocks :) |
15:11.19 | [TK]D-Fender | dlynes : I told you a way to transfer them out, let them sit & circulate, and then call back after a timeout. Pure dialplan, 4-5 lines tops, and no app to compile. What could be easier than that? :p |
15:11.46 | [TK]D-Fender | Dr-Linux>is there any link from where i can see/listen MP3 ads? <----- HUH!?!??!?!? |
15:11.52 | dlynes | [TK]D-Fender: yeah, but you said your solution would have issues with more than one caller |
15:12.07 | melbert | I am having trouble with the CLID changing after doing a match. After doing a: "exten => 18885555555/4444444444,n,Hangup" to block a specific incoming number the caller ID for everything after that shows up as the 888 number and not the caller ID |
15:12.56 | [TK]D-Fender | dlynes : No, I said YOUR idea of trying to compensate for an idiot user not doing a BLIND TRANFER wouldn't work. And I challenged you to prove otherwise :p |
15:13.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
15:13.18 | [TK]D-Fender | dlynes : Valet Parking with an attended transfer sucks jsut the same :) |
15:13.20 | puzzled | hi |
15:13.23 | dlynes | [TK]D-Fender: ah...anyways..whether they do a blind transfer or not, anthony's method will work |
15:13.39 | linagee | [TK]D-Fender: valet parking? do they keep the key? lol |
15:13.51 | ZaVoid | ok stupid question what if my variable is 3.45 instead of 3.45 i treat "." i could "." as a field space.. i'll test.. just thinking out lound here |
15:14.04 | ZaVoid | ignore me |
15:14.21 | [TK]D-Fender | melbert: pastebin the entire context and we'll show you where you went wrong |
15:14.32 | Uatec | hey, i'm using GotoIf(X = 1?3) |
15:14.39 | Uatec | but whatever X equals, it's always going to three |
15:14.53 | Uatec | eg |
15:14.53 | Uatec | <PROTECTED> |
15:14.54 | Uatec | <PROTECTED> |
15:14.55 | [TK]D-Fender | ZaVoid: You'd use CUT to grab the left & right halves so you could say "point" in between. |
15:14.58 | linagee | whatever X equals, it's always going to be 42 |
15:15.03 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
15:15.05 | Uatec | oh |
15:15.08 | dlynes | [TK]D-Fender: btw...that page works wonderfully...thanks again |
15:15.09 | Uatec | ? |
15:15.12 | ZaVoid | ok let me go lookup cut |
15:15.13 | [TK]D-Fender | Uatec: because that is not a valid EXPRESSION at all. |
15:15.25 | [TK]D-Fender | Uatec: Go lookup "asterisk expressions" on the WIKI |
15:15.30 | Uatec | ok |
15:15.35 | Uatec | let me rewrite the example |
15:15.36 | [TK]D-Fender | dlynes : glad to help. |
15:15.53 | Uatec | GotoIf(${command} = 1?6) |
15:16.06 | Uatec | where command is set by the Read() command |
15:16.09 | [TK]D-Fender | Uatec: STILL not an expression at all. Go read the WIKI page on them |
15:16.14 | jcanfield | Okay i got a very natural ring down working with the Polycom500 by setting NumLineKeys to 3 and Calls per line to 1. Can you see any problems with this setup? |
15:16.18 | dlynes | [TK]D-Fender: Just need to get chan_alsa up and running now so I can do external paging and radio for music on hold |
15:16.27 | jcanfield | *Polycom550 |
15:16.46 | [TK]D-Fender | jcanfield: Works doesn't it? |
15:16.55 | [TK]D-Fender | jcanfield: so you left 1 key for speed-dial? |
15:17.18 | [TK]D-Fender | dlynes : For external paging I'd far sooner get an ATA +Amp. |
15:17.22 | jcanfield | [TK]D-Fender, no i was hoping to make that a call park key. |
15:17.38 | [TK]D-Fender | jcanfield: Never going to happen. Find something else to do with it. |
15:17.55 | melbert | Here is the context that it is in: http://pastebin.org/2423 |
15:18.11 | dlynes | [TK]D-Fender: Well, my boss wants me to get radio working as input for moh as well |
15:18.27 | dlynes | [TK]D-Fender: so getting the soundcard working on these machines is still important |
15:18.35 | jcanfield | [TK]D-Fender, crap! so can't to call parking on BLF keys? |
15:18.35 | dlynes | [TK]D-Fender: I cringe at the thought, myself |
15:18.36 | [TK]D-Fender | melbert: exten => 188855555555/4444444444,n,Hangup <- this won't work because you don't have a "1" priority for it. |
15:18.46 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
15:18.55 | jcanfield | *do |
15:19.02 | dlynes | [TK]D-Fender: I really really hate trying to get the latest and greatest hardware to work under linux...and soundcards are one of those things that never seem to catch up |
15:19.04 | [TK]D-Fender | melbert: exten => 188855555555/4444444444,1,Hangup <- this will work. It does not INHERIT anything from the exten => 188855555555,1 version |
15:19.13 | Uatec | ahhh |
15:19.15 | Uatec | $[ ] |
15:19.16 | Uatec | that's odd |
15:19.20 | Uatec | i've not used it else where but it works |
15:19.23 | Uatec | that's quite annoying |
15:19.34 | [TK]D-Fender | jcanfield: Nope |
15:19.50 | [TK]D-Fender | Uatec: Its been like this in * since its creation. |
15:19.52 | dlynes | jcanfield: you should be able to call pickup on blf, but ont call park |
15:20.24 | jcanfield | [That's a bit of a step back. ...so how do you park calls? |
15:20.42 | dlynes | jcanfield: just call transfer to 700, by default, unless you've changed it |
15:21.20 | jcanfield | Hmmm...kinda like the old nortels. :P |
15:21.23 | [TK]D-Fender | jcanfield: Go read up on "call parking" on the WIKI to see how |
15:21.29 | dlynes | jcanfield: then you can blf: exten => 701,hint,park:701@parkedcalls (I think)...check the wiki for sure |
15:21.43 | dlynes | jcanfield: and then exten => 701,1,ParkedCall(701) |
15:22.10 | dlynes | jcanfield: and that WILL NOT work in asterisk 1.2 |
15:22.15 | [TK]D-Fender | dlynesYou never have to do that, it gets auto included if you use the context that gets dynamically created. |
15:22.28 | dlynes | [TK]D-Fender: including the hint? |
15:22.48 | [TK]D-Fender | dlynes : the hint you will have to make, but not the pickup mechanism... |
15:22.54 | dlynes | ok |
15:23.03 | [TK]D-Fender | dlynes : thats get generated automatically. |
15:23.26 | jcanfield | [TK]D-Fender, k will do. I know a lot of my questions expose my ignorance, but i do appreciate the direction. |
15:23.54 | dlynes | jcanfield: actually, nortel is *74 |
15:24.22 | dlynes | jcanfield: which is what i program my systems for...most of my customers are nortel converts |
15:24.42 | dlynes | jcanfield: so even voicemail is *980 :) |
15:25.08 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-212-135.dsl.irvnca.pacbell.net) |
15:25.10 | jcanfield | dlynes, it's still a braindead code, that's why i switched to the Panasonic DBS systems years ago because they handled call park very nicely and users don't have to think. |
15:25.38 | JerJer | users can think? |
15:25.47 | Zeeek | JerJer ! |
15:25.50 | dlynes | JerJer: not really |
15:26.03 | UnixDog | hell I know some admins who cant think |
15:26.18 | dlynes | jcanfield: once the user's get used to a certain system, it's a real pain to get them used to a new system |
15:26.26 | dlynes | users, even |
15:26.35 | JerJer | hehehehe |
15:27.02 | jcanfield | dlynes, that is what a lot of PBX guys say. :P |
15:27.04 | dlynes | jcanfield: I get so many damned users saying...nortel did it this way...how come your phone system doesn't do it that way? and blah blah blah |
15:27.16 | anonymouz666 | JerJer: did you fix your nat problem with openser+ast? |
15:27.28 | Zeeek | has anyone hooked up GrandCentral to * ? |
15:27.59 | JerJer | anonymouz666: not specifically proven fixed, no |
15:28.29 | jcanfield | dlynes, true, but if you set a system up properly the transition can actually be a good thing. iswitch many nortels systems over and never had a complaint. |
15:28.33 | JerJer | that project has all kinds of various anonying network situations - like 4 year old routers and first gen SIP gear |
15:29.26 | JerJer | anonymouz666: we think some of the devices don't support symmetric rtp and others have very very old nat routers, who are not stateful |
15:30.28 | dlynes | jcanfield: yes, but you're in Atlanta; I'm in Canada...Nortel is king, here :) |
15:30.36 | *** join/#asterisk melbert (n=IceChat7@66.179.79.70) |
15:30.58 | anonymouz666 | JerJer: yeap, some devices aren't smart enough to be symmetric |
15:31.05 | anonymouz666 | they suck |
15:31.12 | JerJer | yes they do :( |
15:31.15 | jcanfield | dlynes, I'm in Tulsa. ...you have a point though. |
15:31.21 | melbert | [TK]D-Fender - I lost my connection but I did want to say thanks for helping me. That worked. |
15:31.29 | JerJer | so we had to swap out those bad customers with 3102s |
15:31.50 | JerJer | once they swapped, everything worked as expected |
15:32.03 | dlynes | jcanfield: oh...you've got an atlanta ip block...so figured you were in atlanta |
15:32.26 | dlynes | jcanfield: oh...nvm...read that wrong...cox is in atlanta...not you |
15:32.50 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
15:32.51 | jcanfield | dlynes, ya it's cox, they hand out ip's like candy. |
15:33.24 | dlynes | jcanfield: but it'd be like trying to switch over someone from avaya in the US |
15:33.26 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
15:33.52 | dlynes | jcanfield: well...maybe...i don't know if americans pride avaya like canadians pride nortel |
15:34.08 | Qwell | I've never met a canadian who "prided" nortel |
15:34.13 | agx | hi, i'm getting some like one-way-audio with chan_misdn and HFCPCI. The ougoing audio is ok, the incoming audio is distorted and noised. any idea? |
15:34.25 | *** part/#asterisk melbert (n=IceChat7@66.179.79.70) |
15:34.29 | dlynes | Qwell: you were talking to the telecom guys though, not the end users :) |
15:34.38 | Echinos | Qwell: not these days...(canadian here) |
15:34.52 | Echinos | Although their head office is in my home town |
15:35.06 | jcanfield | dlynes, done, it. Thing about us americans...not very loyal to a product. |
15:35.31 | ZaVoid | hey fender.. http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut this page right? |
15:36.03 | dlynes | anyways...gotta run |
15:36.07 | dlynes | nice chatting with you all |
15:37.11 | *** join/#asterisk `paul (n=vina@124.107.13.212) |
15:40.36 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
15:41.35 | `paul | hi. i have a prob hope someone has a solution for it. here is the set up.... local_number <calls>---> another_local_number --callforward--> toll free number --> enters asterisk -->queue(agents) the problem is we have lots of drop calls the reason being the original caller puts down his phone before an agent answers. testing this setup we noticed that it takes around 6 rings before the agents phone rings. is the problem with asterisk or before the call ent |
15:41.42 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
15:41.48 | drako | whats the way to tell queue.conf when using mixmonitor to use option b so it merge the files. |
15:44.10 | thewiizle | hi |
15:44.17 | thewiizle | whats the config file that stores the trunk settings |
15:46.23 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
15:47.10 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
15:48.18 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
15:49.00 | WilliamK | Can someone with SVN access please remove these lines from the zaptel init file.... it's causing zaptel not to start because the lines aren't commented... <<<<<<< .mine , ======= , >>>>>>> .r3017 |
15:49.12 | Qwell | WilliamK: what branch? |
15:49.18 | WilliamK | 1.4 |
15:49.22 | WilliamK | latest SVN |
15:49.27 | Qwell | looking |
15:49.32 | WilliamK | this just happened like yesterday or so |
15:49.56 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
15:49.59 | WilliamK | thanks :) |
15:50.01 | Qwell | WilliamK: that's all you |
15:50.26 | WilliamK | file I took them out of was /etc/rc.d/init.d/zaptel |
15:50.43 | Qwell | Your local copy is modified, so when you updated, it added those |
15:50.51 | [TK]D-Fender | ZaVoid: Just read the CLI page on it |
15:51.09 | Qwell | do an svn diff, and you'll see it |
15:51.09 | WilliamK | if [ -z "${MODULES}" ]; then (is the line of code between it) |
15:53.08 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
15:53.42 | WilliamK | nice.... it's me |
15:53.55 | WilliamK | I rm'd the directory and pulled a new fresh copy |
15:54.01 | WilliamK | have no idea why SVN did that |
15:54.04 | Qwell | you could've just svn revert'ed those files |
15:54.17 | WilliamK | I just know it goofed the init script |
15:54.18 | WilliamK | :) |
15:54.25 | Qwell | it did it because you had modified one of those lines that got changed (deleted), so it didn't know what to do |
15:54.50 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
15:54.52 | WilliamK | just enough to pull the files |
15:54.53 | WilliamK | :) |
15:55.40 | WilliamK | I bet it probably happened the other night with the TE120P issue I had |
15:56.22 | jfitzgibbon | it's less SVN-specific than the 3-way merge algorithm. You'll get that any time you change the same part of a file that someone else does and your revision tool tries to do a 3 way merge and fails |
15:56.26 | thewiizle | anyway of making asterisk display which config files it is using |
15:56.38 | jfitzgibbon | thankfully SVN tries very hard to prevent you from checking in files that are conflicted |
15:56.49 | jfitzgibbon | CVS was ... less diligent... |
15:56.58 | shinao1 | hi, im kind of stuck with a TDM844B card... and i found out i need some sort of astribank/channelbank to go with it.. and im all out of money. Also, the asterisk server must sit in a server room. I wonder if there is any kind of channel bank that doesnt have a T1 connections? |
15:57.50 | Qwell | shinao1: what? Why would you need a channel bank for an analog card? |
15:58.22 | shinao1 | that uses only FXO/FXS or ethernet to connect to the PABX? |
15:58.54 | Qwell | shinao1: take a step back. start over. why do you think you need a channelbank? |
16:00.44 | defswork | shinao1: http://www.voip-info.org/tiki-index.php?page=Asterisk+Channel+Bank |
16:01.08 | Qwell | don't confuse him more...let's find out the reason first |
16:01.15 | outtolunc | if the asterisk server must be in the server room, the card must be in the asterisk server <G> so the 'lines' will need to be *stretched* |
16:01.30 | defswork | awww |
16:01.33 | Qwell | ... |
16:01.37 | defswork | you upset him with hard questions :) |
16:01.40 | outtolunc | my bad <G> |
16:01.52 | defswork | Why is usually my first question |
16:02.09 | Qwell | clearly somebody gave him invalid information |
16:02.32 | defswork | well with a channel bank he wouldnt need the TDM844B at all would he ? |
16:02.42 | defswork | but he said he'd ran out of money |
16:02.46 | Qwell | no, he would need something to plug it into though, heh |
16:02.56 | outtolunc | which he is 'out of money' |
16:08.00 | thewiizle | sip_registrations.conf |
16:09.34 | n3glv | hi |
16:10.16 | n3glv | if I have an old sip device, could it have problems with currant sip version? (I see sip/2.0 on messages) |
16:11.35 | *** part/#asterisk `paul (n=vina@124.107.13.212) |
16:11.51 | [TK]D-Fender | Qwell: Backwards is thinking very his hmmmMMMM?!?!? |
16:12.11 | n3glv | yofa? |
16:12.18 | n3glv | yoda even |
16:13.25 | [TK]D-Fender | n3glv: indeed |
16:13.35 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
16:13.58 | n3glv | anyway to overcome this? |
16:14.17 | n3glv | it's an old (v.1) zyxel p2000w wifi phone |
16:14.47 | Nugget | those are the shittiest phones ever made. |
16:14.50 | Nugget | worse than grandstream |
16:15.05 | [TK]D-Fender | n3glv: Whats your problem exactly? |
16:15.35 | n3glv | http://www.pastebin.ca/696841 |
16:15.40 | n3glv | it was free btw |
16:15.40 | thewiizle | Should a registration appear when i type 'sip show channels' regardless of whether it is registered or not? |
16:16.08 | n3glv | thewiizle, if it's sending it will say reg sent |
16:16.32 | thewiizle | hmm |
16:16.45 | thewiizle | ok |
16:16.49 | thewiizle | can i include files from sip.conf |
16:17.05 | thewiizle | using 'include sip_additional.conf' for example |
16:17.17 | n3glv | yes |
16:17.20 | n3glv | afaik |
16:17.29 | thewiizle | hmm doesnt seem to be working then |
16:18.31 | thewiizle | i have my registrations in a seperate file |
16:19.04 | n3glv | many systems do |
16:19.14 | n3glv | such as freepbx instsalls |
16:19.17 | thewiizle | :) |
16:19.19 | thewiizle | spot on guess |
16:19.22 | n3glv | installs even |
16:19.39 | thewiizle | the additional files however are not being read by asterisk, hense by channels and peers are not being configured |
16:19.51 | Zeeek | Russell et al, ladies, gents, and the rest of you: #asterisk-users-conference in 10 minutes. Thank you. http://voipUsersConference.org/join.php |
16:19.59 | n3glv | they take a leading # |
16:20.08 | n3glv | that's not a comment |
16:20.41 | *** join/#asterisk VoicePulse_ (n=contact@unaffiliated/voicepulse) |
16:21.47 | thewiizle | :) |
16:21.55 | thewiizle | i knew there was something missing |
16:22.09 | n3glv | so, freepbx system? |
16:22.28 | thewiizle | indeed |
16:22.35 | thewiizle | totally hand compuled so far |
16:23.15 | n3glv | ok, mostly I do distro's that use freepbx (that's a dirty word here) |
16:24.39 | thewiizle | yeh i can imagine |
16:24.45 | thewiizle | seems the best choice |
16:24.54 | thewiizle | freely configurable system with a decent gui |
16:25.09 | thewiizle | asterisknow used to be tops before they locked it down to a full release |
16:25.14 | *** part/#asterisk juliux (n=juliux@ubuntu/member/juliux) |
16:25.25 | n3glv | so, a device running old sip ver may have issues with sip 2.0? |
16:26.18 | [TK]D-Fender | n3glv: pastebin again, its expired |
16:26.37 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
16:28.03 | n3glv | oops |
16:29.40 | n3glv | http://www.pastebin.ca/697692 |
16:30.04 | *** join/#asterisk snk00sj (n=gnelisse@apollo.digitalbase.be) |
16:30.37 | snk00sj | hi, i am trying to compile asterisk from source (all the previous installations removed) |
16:31.03 | snk00sj | but on the last step make config i get the error : System startup links for /etc/init.d/asterisk already exist |
16:31.21 | snk00sj | if i remove this it still returns the same error (although it's gone) |
16:34.50 | [TK]D-Fender | n3glv : SIP/2.0 401 Unauthorized <--- how many times does it have to say you've got the wrong auth info before you go and FIX it? :) |
16:36.28 | n3glv | it has all correct info |
16:36.29 | n3glv | as far as I can tell |
16:37.04 | n3glv | it says lower about md5, the server is not running md5 afaik |
16:37.16 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
16:37.34 | n3glv | WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c73c930" |
16:38.03 | [TK]D-Fender | n3glv: SIp is valid, your auth criteria do not match |
16:38.44 | *** join/#asterisk sgarcia (n=sgarcia@78.Red-81-34-55.dynamicIP.rima-tde.net) |
16:38.55 | sgarcia | hi everyone |
16:39.26 | thewiizle | hmmm |
16:39.29 | thewiizle | 404 all day everyday |
16:40.35 | [TK]D-Fender | NOW its Miller Time (tm) :) |
16:40.50 | n3glv | Guiness |
16:40.53 | n3glv | real beer |
16:41.02 | thewiizle | lol guiness isnt beer |
16:41.13 | thewiizle | draught ale at its best :P |
16:41.14 | n3glv | miller is like making love in a canoe |
16:41.23 | n3glv | &*^%ing near watter |
16:41.29 | thewiizle | rather enjoyable with multiple possible outcomes? |
16:41.29 | n3glv | water even |
16:41.30 | n3glv | lol |
16:41.52 | n3glv | not an ale, it's a stout |
16:42.18 | [TK]D-Fender | load res_beernazi.so |
16:42.51 | thewiizle | load why_do_i_get_404.so |
16:43.05 | n3glv | grep 'where's my beer' fridge |
16:43.13 | thewiizle | no no no |
16:43.19 | thewiizle | grep "beer fridge" |
16:43.51 | thewiizle | ok ive added another extension |
16:44.03 | thewiizle | surely i should be able to dial an extension in the same context |
16:44.22 | thewiizle | but no |
16:44.25 | n3glv | this is being sent to the device |
16:44.26 | n3glv | From: <sip:9667@voipcoop.org;user=phone>;tag=15746980534AF4DDE6 |
16:44.40 | snk00sj | when i start asterisk using : asterisk -cvv it starts fine, although when i use /etc/init.d/asterisk start, it gives me this error : /usr/sbin/safe_asterisk: 161: Syntax error: Bad fd number => although everything compiled fine, where should i start looking ? |
16:44.43 | n3glv | is user=phone the * user? |
16:45.08 | [TK]D-Fender | 9667 <-- |
16:45.23 | [TK]D-Fender | thewiizle: Self-explanitory... |
16:45.29 | [TK]D-Fender | thewiizle: NOT FOUND <--- |
16:45.34 | thewiizle | yeh |
16:45.36 | thewiizle | i dont get why |
16:45.37 | [TK]D-Fender | thewiizle: PASTEBIN is your friend |
16:45.39 | [TK]D-Fender | ~pb |
16:45.39 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
16:45.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
16:45.52 | thewiizle | i have nothing to paste :( |
16:46.18 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
16:46.33 | [TK]D-Fender | thewiizle: Gee, fat load of good that does everyone don't you think? |
16:46.49 | thewiizle | Why, what do you want to see? |
16:47.45 | *** part/#asterisk sgarcia (n=sgarcia@78.Red-81-34-55.dynamicIP.rima-tde.net) |
16:48.01 | n3glv | the user is 9667, that is correct |
16:48.10 | n3glv | but what is user=phone in that line? |
16:48.29 | [TK]D-Fender | n3glv: forget that and start looking at your SIP setup for that device |
16:49.01 | [TK]D-Fender | thewiizle: Gee, I don't know, how about SIP DEBUG showing the failed call attempt? |
16:50.07 | thewiizle | Looking for *43 in from-internal (domain 195.26.235.104) |
16:50.08 | n3glv | that's all in the pasteing |
16:50.10 | n3glv | pastbin |
16:50.10 | thewiizle | thats about it |
16:50.28 | n3glv | www.pastebin.ca/697692 |
16:51.12 | [TK]D-Fender | n3glv: I wasn't asking about YOUR problem... |
16:51.40 | [TK]D-Fender | thewiizle: Well clearly that exten isn't in that context, what more do you want? |
16:51.50 | *** join/#asterisk M-I-A (n=yada@CPE00304827782b-CM0014f85e8abe.cpe.net.cable.rogers.com) |
16:51.58 | thewiizle | i dont think its a case of not in the context |
16:52.06 | thewiizle | it seems to be more like the context doesnt exist |
16:52.12 | [TK]D-Fender | n3glv: I that isn't your SIP SETUP, thats the CLI SIP DEBUG that shows that things don't match |
16:52.19 | n3glv | sri |
16:52.26 | [TK]D-Fender | thewiizle: And why would THAT be? |
16:52.38 | thewiizle | no idea |
16:52.54 | thewiizle | i cant find from-internal in extensions.conf or extensions_additional.conf |
16:52.54 | [TK]D-Fender | thewiizle: You tell a sip device to use a context and it doesn't even exist. Why are you even wondering why it can't FIND ANYTHING? |
16:53.03 | n3glv | going try and reg with softphone, but, it does the same on two servers |
16:53.08 | thewiizle | The context is a default one |
16:53.15 | [TK]D-Fender | thewiizle: FreePBX debugging I see...... you are in the WRONG PLACE.... |
16:53.42 | thewiizle | nah not so much debugging just picking this all back up |
16:54.07 | [TK]D-Fender | thewiizle: Well if the context doesn't even exist well I guess FreePBX is screwed |
16:54.19 | thewiizle | ai |
16:54.21 | thewiizle | it would seem that way |
16:54.28 | M-I-A | Am I allowed to advertise my Digium TDM2401e that I have for sale? |
16:54.44 | [TK]D-Fender | M-I-A: Sure. Ebay works, or the mailing list. |
16:54.46 | n3glv | thewiizle, if it's freepbx, use context=from-trunk or from-pstn |
16:55.16 | M-I-A | TK I have it on eBay but have not had a hit for close to 20 days :( |
16:55.39 | [TK]D-Fender | M-I-A: and when does the auction close? |
16:55.54 | M-I-A | 3 days from now... already had one auction close with no bids |
16:56.29 | [TK]D-Fender | M-I-A: either your rating suck, your auction info sucks, your location sucks, etc... |
16:56.37 | M-I-A | TK: not even one person watching the auction :( |
16:56.57 | [TK]D-Fender | M-I-A: and you can often go up to the last few minutes before seeing a shit-storm of bigs for people trying to grab it. |
16:57.17 | [TK]D-Fender | M-I-A: smart bidders don't announce they're watching... |
16:58.19 | M-I-A | TK what is a good asking price for that card? |
16:59.22 | [TK]D-Fender | M-I-A: Use some common sense. |
17:00.26 | n3glv | I just got a tdm400p with one fxo and one fxs for $100 |
17:00.51 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-d29f37af4fb0e6f1) |
17:00.51 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
17:01.53 | hmmhesays | hey [TK]D-Fender you want to hear that go awful noise I was talking about? |
17:02.13 | [TK]D-Fender | hmmhesays: No, s'ok I believe you :) |
17:02.33 | hmmhesays | haha ok, i'm going to post it on the support forums |
17:02.58 | *** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net) |
17:03.21 | M-I-A | TK: I've searched the web for pricing but those are brand new cards, I used mine for three months. So I was trying to price accordingly. Could you take a look at my auction and see if I'm totally whacked |
17:04.20 | M-I-A | n3glv: that seems like a really good price |
17:04.30 | UnixDog | russellb: when they going to fix the freebsd g729 regtool |
17:04.54 | j0 | is an x100p card considered stable enough to run a business line off? |
17:05.04 | [TK]D-Fender | M-I-A: You'll have to come to your own conclusion. Keep in mind though, its a big card so only larger installs will want it and those with cases big enough to support it. Then consider those who WOULD need it probably wouldn't want it USED. |
17:05.11 | [TK]D-Fender | j0: No. |
17:05.36 | j0 | [TK]D-Fender: thanks... is it worth trying to use an analog line at all? |
17:05.43 | j0 | I can't get a t1 here |
17:06.17 | [TK]D-Fender | j0: Sure, and if you HAVE the X100P may as well TRY it, just don't be surprised if you're disappointed if CID doesn't work, they audio is iffy, or have echo problems. |
17:06.31 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-d29f37af4fb0e6f1) |
17:06.32 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
17:06.51 | M-I-A | j0: I got a TDM2401e for sale :) |
17:06.53 | j0 | i have a knockoff one.. and unless it's just going to plain work, it's not worth the trouble |
17:07.23 | j0 | M-I-A: might be on the high end for me.. what's on it? |
17:08.02 | M-I-A | j0: 4 fxo and echo can. module |
17:08.40 | j0 | funny, i was just looking at that auction.. lol |
17:08.49 | M-I-A | LMAO! |
17:09.05 | M-I-A | thats really scary |
17:09.25 | j0 | these asterisk appliances look nice |
17:09.28 | [TK]D-Fender | M-I-A: And why are YOU selling it? |
17:09.51 | M-I-A | we went PRI |
17:10.05 | UnixDog | I saw one on ebay for 1600 |
17:10.07 | UnixDog | lol |
17:10.15 | *** part/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
17:10.39 | j0 | has anyone had experience with the remotely managed appliances from digium or trixbox? |
17:11.08 | UnixDog | its easy |
17:11.17 | UnixDog | just have to have ssh and www |
17:11.32 | j0 | its easy to have them do everything? |
17:11.55 | UnixDog | depends on what you mean by everthing |
17:12.06 | UnixDog | its a web appliance for the most part |
17:12.18 | j0 | i don't want to do anything more than do the initial setup... everything other than minor configuration they can handle.. |
17:12.19 | snk00sj | anyone know a mirror of asteriskdocs.org ? |
17:12.32 | M-I-A | make the morning coffee is what i've been trying to get my box to do for me |
17:14.46 | thewiizle | yo |
17:14.56 | thewiizle | whats the variable for number dialled when used in a context |
17:15.43 | thewiizle | eg, Dial _,1,Dial(SIP/$number) |
17:16.58 | hmmhesays | what the hell, you can't attach anything to an asterisk forum? |
17:18.22 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
17:20.06 | errr | when I setup my sangoma card it mad a zapata.conf and a zapata-auto.conf . If I want to change the context being used for incoming calls to a test context would I change it in zapata.conf or in zapata-auto.conf |
17:20.18 | errr | s,mad,made, |
17:22.03 | [TK]D-Fender | thewiizle: ${EXTEN} This is SUPER * 101... go lookup "asterisk variables" on the WIKI |
17:22.25 | [TK]D-Fender | errr: Depends on the CONTENTS of those files. |
17:23.15 | errr | [TK]D-Fender: would zapata.conf need to have an include => in it for it to be using the -auto.conf file? |
17:26.10 | *** join/#asterisk |omni| (n=rob@net82.allied-security.com) |
17:26.41 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
17:26.49 | errr | [TK]D-Fender: http://fluxbox.pastebin.ca/697749 (zapata.conf) http://fluxbox.pastebin.ca/697751 (zapata-auto.conf) |
17:29.49 | hmmhesays | http://myweb.cableone.net/mattman21/sample2.wav <-- if anyone wants to hear what I'm talking about |
17:30.11 | *** join/#asterisk Cybertoy (n=cybertoy@swillux.swill.org) |
17:30.34 | Kwakwa | errr, it will read zapata.conf |
17:30.41 | errr | Kwakwa: only? |
17:30.46 | Kwakwa | if you check the CLI> u can usually see what's loaded |
17:30.53 | Kwakwa | yeah, there's no include in that |
17:31.04 | errr | Kwakwa: ok. thanks |
17:31.07 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
17:32.26 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
17:32.29 | errr | to get zapata.conf to reload you must retsart asterisk?? |
17:33.36 | kaldemar | depends on the settings you changed. |
17:33.55 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
17:33.58 | kaldemar | a context change can be dealt with a reload. |
17:34.14 | errr | ah nice, that worked, thanks for the help :) |
17:34.21 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
17:38.12 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
17:39.51 | [TK]D-Fender | errr: zapata-auto.conf is not even USED anywhere. It is irrelevant |
17:40.40 | bkruse | errr: everything but signalling changes (besides channel declarations and their options) |
17:41.34 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
17:45.40 | *** join/#asterisk sniper[FOO] (i=Snip3r@217.27.214.111) |
17:45.43 | *** join/#asterisk naxeji (n=nax@M1097P021.adsl.highway.telekom.at) |
17:45.51 | sniper[FOO] | hi there |
17:48.24 | sniper[FOO] | had to revert to the 1.0 branch of * for a short time, does anyone remember how can I make a macro called from the Dial() app bail out and terminate the Dial() as if I set Set(MACRO_RESULT=CONTINUE)? |
17:49.12 | sniper[FOO] | now I use SetVar(MACRO_RESULT=CONTINUE) |
17:50.12 | sniper[FOO] | but nothing happens, the SetVar app gets executed and the bridge occurs back in the Dial app |
17:50.31 | sniper[FOO] | and that's something I really don't want |
17:51.24 | Kwakwa | What version of * u on sniper? |
17:51.33 | *** part/#asterisk Cybertoy (n=cybertoy@swillux.swill.org) |
17:51.58 | Kwakwa | ahh, I didn't read it all.. 1.0, ouch |
17:51.59 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
17:51.59 | [TK]D-Fender | sniper[FOO]: pastebin all related code. |
17:52.09 | sniper[FOO] | OK |
17:52.40 | [TK]D-Fender | ~pb |
17:52.41 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
17:53.04 | sniper[FOO] | I know what a pastebin is, thanks, though :) |
17:58.39 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
18:01.16 | naxeji | hi, can someone recommend me a good (stable) linux distribution for asterisk. Should I use a special distribution like AsteriskNOW, Astlinux or Trixbox, or is a normal debian enough? |
18:02.03 | chemikk | naxeji: debian stable |
18:02.16 | hmmhesays | whatever you are comfortable with |
18:02.25 | hmmhesays | stability has a lot to do with the administrator ;) |
18:02.45 | naxeji | ok, thx for the recommendation =) |
18:04.39 | *** join/#asterisk Godsey (n=jason@pdpc/supporter/sustaining/Godsey) |
18:06.21 | Godsey | is there a way to log the ip of registrations? |
18:06.59 | Godsey | about 3 days ago, my cdr-csv/Master.csv file started showing my device dialing out |
18:07.10 | Godsey | I'm trying to figure out how someone is registering as my device |
18:10.19 | sniper[FOO] | Kwakwa, [TK]D-Fender: http://pastebin.ca/697791 |
18:11.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:12.45 | shido6 | heh |
18:12.55 | shido6 | iax bandit or a sip bandit? |
18:12.57 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
18:14.56 | *** join/#asterisk bruder (n=sergio@201.21.180.98) |
18:15.43 | Godsey | sip bandit |
18:16.03 | Godsey | but I'm at a loss |
18:16.19 | Godsey | in asterisk logs, it looks just like my device is calling out |
18:16.47 | Godsey | it's a pap2-na.. I just restricted access to it and added a password (I'm stupid for not having one before) |
18:16.56 | Godsey | and changed the passwords for my device |
18:18.27 | Godsey | it looks like they dial and get my voicemail |
18:18.33 | Godsey | then my extension starts placing calls |
18:23.31 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
18:24.33 | [TK]D-Fender | sniper[FOO]: Not sure.... |
18:24.43 | [TK]D-Fender | sniper[FOO]: Maybe someone else will... |
18:34.55 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
18:37.39 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
18:37.56 | denon | totally OT I know, but I'm in kind of a bind, any of you guys feel like lending a hand on an openvpn config question? |
18:39.46 | jwh | sup? |
18:41.11 | bkruse | i can give it a shot |
18:41.51 | drako | ok another weird problem.... when i get the calls from the ISDN (BRI) interfaces and i put it on the queue it works perfect but if first i put a background and then waitexten before the queue, when the call reach the queue and is answer it comes with no SOUND from the Caller |
18:42.07 | drako | but if i get rid of the background and waitexten it works. |
18:43.05 | jwh | Godsey: you do have guest calling disabled right? |
18:43.15 | Godsey | jwh: I'm not sure now |
18:43.30 | Godsey | sip.conf [general[ first thing context=banned |
18:43.54 | Godsey | which is exten => i,1,Hangup and t,1,Hangup |
18:44.22 | *** join/#asterisk Peaceful (n=peaceful@70.98.162.62) |
18:44.26 | Godsey | I have this under my pap2-na device: insecure=port,invite |
18:44.39 | Godsey | and host=dynamic |
18:45.08 | Peaceful | So I upgraded from 1.2.13 to 1.2.24, and DISA() seems to not work anymore. Just gives a fast-busy after trying to dial. ??? |
18:45.32 | [TK]D-Fender | Peaceful: ...... |
18:46.07 | Peaceful | [TK]D-Fender, you lost me. |
18:46.24 | [TK]D-Fender | Peaceful: Didn't take much... |
18:46.33 | Peaceful | hehe |
18:46.46 | [TK]D-Fender | Peaceful: Ho about showing us something USEFUL? |
18:46.59 | Peaceful | [TK]D-Fender, like the error it doesn't give? |
18:47.03 | jwh | Godsey: hm |
18:47.05 | Peaceful | what, exactly? |
18:47.06 | *** join/#asterisk VijayG (n=vijay@58.68.47.120) |
18:47.17 | [TK]D-Fender | Peaceful: like your dialplan and its CLI output for what it DOES execute. |
18:48.05 | Godsey | jwh: when I set my ip I tend to get 1 way audio on incoming calls |
18:48.24 | Peaceful | [TK]D-Fender, Here's the dialplan (personal info changed): exten => 5606,1,DISA(333333333,internal,"MyCallerID" <80155555555>) |
18:48.27 | jwh | nat? |
18:48.34 | Godsey | yes, I have nat=always tho |
18:48.43 | [TK]D-Fender | Peaceful: pastebin EVERYTHING. |
18:48.55 | Peaceful | [TK]D-Fender, alrighty. Hold on. |
18:49.03 | Godsey | and i changed the port to 15061 and have that forwarded to pap2-na |
18:49.06 | [TK]D-Fender | Godsey: Where is your PAP2 relative to *? |
18:49.17 | Godsey | and pap2-na is configured to use 15061 and 15061 as ext-port |
18:49.25 | Godsey | asterisk server is not behind nat, pap2-na is |
18:49.38 | [TK]D-Fender | ~sipnat |
18:49.39 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
18:49.40 | [TK]D-Fender | ^^^^^ go read |
18:49.57 | Godsey | well, if I leave host=dynamic it works :) |
18:52.20 | Peaceful | [TK]D-Fender, ok, that's annoying. I try it again to reproduce the console output, but DISA works this time. Grrrr. |
18:59.19 | *** join/#asterisk shellprompt (n=shellpro@unaffiliated/shellprompt) |
19:00.36 | shellprompt | hello all - I have fallen at the first hurdle - before I have even installed! it seems that the documentation for asterisk "first timers" is not there - http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 can anyone point me in the right direction? |
19:00.48 | Qwell | ~tfot |
19:00.49 | jbot | it has been said that tfot is "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
19:00.52 | Qwell | erm |
19:00.54 | Qwell | ~book |
19:00.54 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
19:01.00 | [TK]D-Fender | shellprompt: Go read the readme's that COME with your source tarball |
19:01.02 | hmmhesays | so who wants to listen to my crazy sound problem |
19:01.07 | Qwell | shellprompt: there's a mirror |
19:01.20 | [TK]D-Fender | Qwell: Where? |
19:01.24 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
19:01.30 | Qwell | [TK]D-Fender: at your site? :P |
19:01.55 | [TK]D-Fender | Qwell: For the BOOK yeah, but thats for 1.2 I'd like to assume they're going to install 1.4 which it does NOT cover :) |
19:02.14 | [TK]D-Fender | Qwell: And Asteriskdocs did have guides for 1.4 IIRC |
19:02.23 | shellprompt | thanks for the pointer. it has still not downloaded so I have not had a chance to review the tarball. |
19:02.24 | [TK]D-Fender | Qwell: In articles seperate from TFOT |
19:02.52 | [TK]D-Fender | shellprompt: Always check the big neon sign labeled "README", or "INSTALL", etc :p |
19:03.57 | shellprompt | I apologise - no ignorance intended. |
19:07.46 | *** join/#asterisk TicoTuco (n=matheus@200.250.100.25) |
19:08.21 | jcanfield | Can i have an internal dialtone and an external dialtone? |
19:09.36 | jcanfield | after _9NXXXXXX switch to external dt. I found I can make it dt go away with ingnorepat. |
19:11.23 | rickross | anyone here experienced with SpectraLink cordless handsets? |
19:11.43 | *** part/#asterisk naxeji (n=nax@M1097P021.adsl.highway.telekom.at) |
19:11.49 | [TK]D-Fender | jcanfield: ?! |
19:12.01 | rickross | we have recently switched to using Polycom SOundPoint IP phones and are DYING for a reasonable way to get rid of the cords |
19:12.21 | [TK]D-Fender | rickross: INSANITY |
19:12.38 | rickross | TK - I don't get it |
19:12.57 | rickross | we're insane for using/not using something? |
19:13.04 | [TK]D-Fender | rickross: Soundpoit phones are the best... why are you looking to get rid of them? |
19:13.05 | jcanfield | [TK]D-Fender: after 9 is pressed, DT need to change tone....I guess going away will work. |
19:13.14 | rickross | oh, I am not! |
19:13.19 | rickross | I love the SOundPoint |
19:13.29 | rickross | but I need some wireless handsets, too |
19:13.50 | rickross | I am stuck at my desk now (cannot handle it :) |
19:13.58 | [TK]D-Fender | rickross: haven't tried those DECT ones.... not sure about base functionality. My Aastra 57i CT DECT is BLEH |
19:14.20 | rickross | I wondered about the Aastras :( |
19:14.36 | [TK]D-Fender | jcanfield: if you want dialtone top stop after 9 then stop using ignorepat. And that only applies to zaptel channels |
19:14.47 | rickross | I figured if Polycom liked SpecraLink enough to buy them, then maybe they're pretty good |
19:14.58 | [TK]D-Fender | rickross: They are tied to their base and I recommend AGAINST them. |
19:15.18 | [TK]D-Fender | rickross: SpectraLink may be better... depnds on how their bases work. |
19:15.38 | [TK]D-Fender | rickross: Just saying Aastra's = ass |
19:15.45 | jcanfield | [TK]D-Fender: Okay, that makes sense. |
19:15.47 | rickross | thx - appreciate the candor |
19:16.03 | *** join/#asterisk Bentley (n=rcourtna@S010600195bb1a3a2.cg.shawcable.net) |
19:16.19 | [TK]D-Fender | rickross: There are only certain bits of bile I spare this channel, the rest flies freely :) |
19:16.39 | rickross | where would you buy Polycom's? We bought some from Atacomm, but have had a long delivery problem |
19:17.10 | rickross | I'd love to know a good vendor who ships quickly with a good price |
19:18.15 | russellb | rickross: i can send you a phone very quickly for just $30 |
19:18.25 | russellb | i can't guarantee type, or functional state. |
19:18.31 | rickross | :) |
19:18.43 | rickross | thx, russellb ! |
19:18.46 | Bentley | hi all, does anyone here have a gpx2000? If so, does blind transfer with the TRNF button actually work? I've got 8 new phones where it doesnt |
19:18.50 | russellb | anytime! |
19:18.58 | [TK]D-Fender | rickross: www.telephonydepot.com |
19:19.45 | lirakis | Bentley: yes blind transfer works fine |
19:20.06 | Bentley | linagee, are you running Software Version: Program-- 1.1.4.18 Bootloader-- 1.1.4.6 ? |
19:20.08 | lirakis | Bentley: but i use the built in transfer ability of the phone.. i do not use "feature codes" |
19:20.15 | Bentley | same here |
19:20.18 | Bentley | hrm |
19:20.44 | lirakis | Bentley: yeah it works 100% as normal... ive used gxp-2000's a lot .. ive never had a transfer problem before |
19:20.55 | [TK]D-Fender | rickross: I and my clients have been very please with them |
19:21.12 | rickross | browsing their site now, TK |
19:21.14 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
19:22.15 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
19:22.37 | festr_ | hello, anyone using musiconhold random=yes? it does not work for me in recent 1.4 |
19:22.49 | festr_ | always start first file |
19:29.07 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
19:35.32 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
19:35.42 | duki | hello |
19:38.07 | duki | I just installed asterisk 1.4.5, and configured the minimum in sip.conf and extensions.conf. I run asterisk without errors but when trying to call a sip phone (registred one), the CLI tells me: |
19:38.11 | duki | No application 'Dial' for extension (sip, 2124, 1) |
19:38.34 | duki | I have this line in extensions.conf in the sip context: |
19:38.50 | duki | exten => 2124,1,Dial(SIP/mustapha,20,tr) |
19:39.26 | duki | It seems Dial is not recognized ! |
19:39.41 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
19:39.55 | duki | thanks for any help. |
19:41.06 | hmmhesays | anyone ever get "operation not permitted" when trying to modprobe ztdummy? |
19:42.00 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
19:42.02 | Meaty | Hi all, anyone know why asterisk may not send Authorization header in a REGISTER sip packet when asterisk is used as a sip client ? |
19:45.42 | Echinos | Is anyone else able to connect to FWD with IAX2 right now? |
19:46.11 | Echinos | I'm not sure if it is a config problem on my (and other users) end, or a server issue |
19:47.22 | [TK]D-Fender | duki: Go verify your modules.conf |
19:47.42 | Echinos | ie. IAX2 support on FWD is unstable, not just unstable in general |
19:51.56 | duki | [TK]D-Fender: Thank you, the dial module was not loaded. |
19:56.40 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579178.dsl.bell.ca) |
19:57.21 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
20:05.11 | *** join/#asterisk asdx (n=foo@adsl-152-79.click.com.py) |
20:05.15 | asdx | hi |
20:05.40 | asdx | my isp is blocking sip, is there a way to bypass that |
20:08.20 | Strom_M | use different ports? |
20:10.14 | asdx | they don't block ports, i think they do packet sniffing |
20:10.41 | Strom_M | well...perhaps use IAX instead then? |
20:12.54 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
20:15.38 | asdx | Strom_M: IAX works but a friend is using some special adapter that don't have IAX |
20:16.02 | Strom_M | tunnel SIP over something else then, maybe |
20:16.10 | asdx | yeh |
20:17.25 | Meaty | in ssh maybe |
20:17.33 | [TK]D-Fender | asdx: or setup * over there and have their device talk to * |
20:17.50 | [TK]D-Fender | Meaty: Yes, VoIP over TCP = fun! |
20:18.02 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
20:18.03 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
20:18.10 | *** join/#asterisk tamp4x (n=syntheti@vonmail.vonworldwide.com) |
20:18.12 | Meaty | only sip over tcp |
20:18.31 | asdx | is SIP over SSH possible? |
20:18.36 | tamp4x | anyone in ct area need a job? priv msg me. gui users do not. |
20:19.30 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.220.191) |
20:21.46 | shido6 | thats a damn good question |
20:21.50 | asdx | [TK]D-Fender: yeah |
20:21.55 | shido6 | i've never had the time to try it |
20:22.04 | shido6 | i'll try it tonight |
20:22.25 | Corydon76-dig | asdx: I'd suggest over openvpn |
20:22.35 | Corydon76-dig | Tunneling over TCP isn't the best of ideas |
20:22.49 | asdx | Corydon76-dig: ok |
20:22.55 | Corydon76-dig | especially for voip |
20:22.58 | asdx | thanks |
20:23.06 | shido6 | ssh has a faster setup time tho |
20:23.16 | shido6 | if you're in a hurry and need something pretty secure ssh might work |
20:23.27 | Corydon76-dig | shido6: you can tunnel over openvpn with no encryption |
20:23.27 | shido6 | bring a phone and a usb stick with putty |
20:23.45 | shido6 | how long does that take to setup? |
20:23.58 | Corydon76-dig | In fact, if the ISP is truly filtering, then tunneling with no encryptiion will work fine |
20:24.11 | Corydon76-dig | shido6: handshake only |
20:28.47 | *** join/#asterisk juxhi (n=juxhi@241-82.97-97.tampabay.res.rr.com) |
20:28.54 | juxhi | hello |
20:29.13 | juxhi | i am trying to compile zaptel but i keep getting an error |
20:30.20 | Strom_M | please share this error with us |
20:30.30 | juxhi | i am recreating it |
20:30.37 | Strom_M | lest we have to merely guess at it |
20:30.39 | juxhi | give me a sec |
20:30.40 | Strom_M | and you don't want that |
20:30.47 | Strom_M | because we'll think up things like |
20:31.00 | Strom_M | ERROR: YOU HAVE NOT ADDED ENOUGH SHARP CHEDDAR TO TEH RECIPE |
20:31.09 | juxhi | hmmm |
20:31.13 | juxhi | it might be that |
20:31.14 | Qwell | Strom_M: next error I wrote... |
20:31.16 | Strom_M | to which the solution is clearly "eat more sushi" |
20:31.24 | Qwell | I'm stealing that, and crediting you |
20:31.27 | Qwell | write* |
20:31.33 | Strom_M | hahahaahah, kickasss |
20:31.53 | Strom_M | make it show up at, like, verbosity 700 |
20:34.44 | Strom_M | VERY LOW SODIUM |
20:34.52 | juxhi | u sure u want it? |
20:34.52 | Strom_M | PLEASE RECYCLE |
20:34.55 | Strom_M | yes |
20:35.07 | juxhi | /usr/src/zaptel-1.4.5.1/wct4xxp/../oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_channel.c:9799: internal compiler error: Segmentation fault |
20:35.23 | Strom_M | ooh, that's a yummy oen |
20:35.27 | juxhi | seg fault |
20:35.29 | Strom_M | s/oen/one/ |
20:35.32 | juxhi | i love those |
20:35.46 | Strom_M | Qwell: uh, you code! you fix it! |
20:35.52 | drako | ...ok another weird problem.... when i get the calls from the ISDN (BRI) interfaces and i put it on the queue it works perfect but if first i put a background and then waitexten before the queue, when the call reach the queue and is answer it comes with no SOUND from the Caller |
20:35.57 | drako | but if i get rid of the background and waitexten it works. |
20:37.49 | juxhi | so what's up |
20:39.48 | juxhi | it does this even with the packages provided by the ubuntu repositories |
20:40.06 | Strom_M | juxhi: i dont know where to begin; what architecture are you compiling on? |
20:40.37 | juxhi | 386 |
20:40.48 | Strom_M | which version of ubuntu? |
20:41.17 | juxhi | alternative version 6.06 i think |
20:41.24 | juxhi | might be 610 |
20:41.26 | Strom_M | server? |
20:41.29 | juxhi | yes |
20:41.38 | juxhi | basically it's server |
20:42.14 | Qwell | juxhi: what version of gcc? |
20:42.45 | juxhi | 4.0.3 |
20:44.43 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
20:45.20 | linagee | Bentley: i'm running the latest. it seems very buggy. :( |
20:45.35 | linagee | Bentley: i would NOT NOT NOT TRIPLE QUADRUPLE NOT upgrade to the latest. |
20:45.55 | Strom_M | which "latest" are you going on about, linagee? |
20:46.03 | rene- | hey, when the manager documentation says variables passed to originate must be on its own line, does it means one Variable: var=val\r\n per each variable one wants to pass or maybe Variable" var1=val1\r\nvar2=val2\r\n\r\n |
20:46.04 | linagee | Bentley: don't be tempted if they say a new feature makes GOLD when you boot it. |
20:46.08 | chemikk | i love you people :) |
20:46.25 | Bentley | linagee, these phones came shipped with the latest |
20:46.30 | linagee | Bentley: yuck |
20:46.37 | linagee | Bentley: 1.1 or 1.4? |
20:46.45 | Bentley | i actually found out why blind xfer wasn't working |
20:47.12 | Bentley | 1.1.4.18 |
20:47.19 | *** join/#asterisk smace (n=chatzill@200.220.198.107) |
20:47.26 | linagee | Bentley: yes. just confirmed. that's the crappy one. |
20:47.33 | linagee | Bentley: does the phone boot every time you turn it on? |
20:47.39 | linagee | or like 25%? |
20:47.44 | Bentley | linagee, yes - it's working fine |
20:47.47 | linagee | weird |
20:47.55 | Bentley | all features work .. iw as just having probs with the xfer |
20:48.09 | Bentley | but turned out to be related to a freepbx config (i think) |
20:48.53 | linagee | Bentley: i think a lot of people are so pissed off as to start a class action lawsuit against them if they don't fix the "can't revert firmwares, latest one breaks it" thing. :-/ |
20:49.30 | linagee | Bentley: in which case i'm sure they would point and laugh and say, "the website SAID not to upgrade! lol" |
20:49.32 | asdx | is vpn something like samba/smb/cifs? |
20:49.34 | Bentley | heh - the phone's inexpensive .. you have to give them that |
20:49.46 | Strom_M | i'd start a class action lawsuit for foisting shitty phones on the market in the first place ;) |
20:49.55 | linagee | Strom_M: true. :( |
20:49.57 | Qwell | You guys must be talking about gs |
20:50.09 | linagee | Qwell: lol. i'm not going to say the name until i put a flame suit on |
20:50.14 | Qwell | ~gs |
20:50.15 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
20:50.16 | linagee | flame resistant |
20:50.17 | Qwell | You don't have to |
20:50.19 | Strom_M | Qwell: OMG HOW DID YOU KNOW !?!? |
20:50.47 | linagee | Strom_M: have you used a digium IAXy? i want an iax solution that's not going to have horrid NAT problems. |
20:50.55 | Strom_M | linagee: yes |
20:50.56 | linagee | (ata) |
20:51.01 | shido6 | they rock |
20:51.01 | Strom_M | i love love love my digium iaxy |
20:51.12 | linagee | Strom_M: is the IAXy good, or is there a good phone that has native IAX support? (not gs) |
20:51.19 | shido6 | get an iaxy |
20:51.22 | shido6 | unless |
20:51.22 | Strom_M | iaxy |
20:51.28 | shido6 | u live in a very very hot and sandy climate |
20:51.41 | linagee | Strom_M: i thought hard phones are always better than ATAs. more functions/features/fun |
20:51.50 | shido6 | hah |
20:52.03 | shido6 | that must have come from someone who doesnt putz with features.conf |
20:52.12 | linagee | Strom_M: i need something to distribute to family. *maybe* an iaxy would work.... :-/ |
20:52.34 | shido6 | pre provision your iaxy's BEFORE you ship them to family |
20:52.51 | linagee | shido6: LOL! voip-ing the family is not such an uncommon thing around here? lol |
20:52.52 | shido6 | and the PAP2-TNA u can get to and view a web interface to make changes if necessary |
20:53.04 | shido6 | PAP@t-NA, rather |
20:53.08 | shido6 | sunuva new keyboard |
20:53.28 | linagee | shido6: anything to screw ma bell out of another $0.50, right? :) |
20:53.43 | shido6 | well... |
20:53.48 | shido6 | thats how it starts |
20:53.56 | shido6 | then you look at your bill after the year is out |
20:54.03 | shido6 | and go......... wow..... i spent $2k |
20:54.07 | linagee | shido6: lol. yup. |
20:54.10 | Qwell | "Why do I even have a land line?" |
20:54.13 | shido6 | but I was on the phone for about 90% of my life |
20:54.13 | linagee | shido6: we could actually split the phone bill |
20:54.20 | shido6 | yeah |
20:54.24 | shido6 | why do I even have a land line |
20:54.27 | shido6 | oh yeah.. 911 |
20:54.30 | Qwell | I don't have one :P |
20:54.36 | linagee | shido6: my parents are paying cocks right now. they have horrid compression going on. |
20:54.37 | shido6 | in my old city |
20:54.45 | linagee | shido6: and caller id for cocks is not a standard feature! |
20:54.49 | linagee | s/cocks/cox/ |
20:54.56 | shido6 | you could walk to the hospitol sit in the emergency room for 3 hrs and see a doctor faster than the ambulance would show up |
20:55.12 | linagee | yikes |
20:55.12 | Qwell | shido6: I lived in L.A. You would often get busy signals calling 911. |
20:55.20 | J4k3 | 9 1 1 is slow in your town |
20:55.23 | J4k3 | ;) |
20:55.31 | linagee | shido6: best just to go to an emergency room beforehand or something and get their cell number and drive there yourself. lol |
20:55.33 | Qwell | and if you did get through, you'd have to go through an IVR |
20:55.35 | shido6 | gunshot wound to the stomach sitting in the triage getting your blood pressure checked is kind of hilarious |
20:55.42 | shido6 | waiting on the doctor |
20:55.44 | shido6 | im sorry |
20:55.48 | shido6 | waiting on a room to see a doctor |
20:56.04 | J4k3 | shit, I suggest carrying tampons |
20:56.12 | linagee | shido6: knowing triage first aid FTW. :) |
20:56.24 | J4k3 | (tampons work wonders on gunshot wounds... unless its a shotgun...) |
20:56.34 | shido6 | then u just need some salt |
20:56.38 | linagee | J4k3: people live through shot gun wounds? |
20:56.39 | UnixDog | looking at askozia vs the digium gui |
20:56.39 | shido6 | since you;ve already been peppered |
20:56.57 | UnixDog | yes |
20:57.14 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
20:57.17 | UnixDog | it depeneds on how close you where and what is in the load |
20:57.20 | J4k3 | linagee: yeah.. I know a guy who was randomly shot in the gut with a .410 (small) shotgun. he recently got a transplanted (!!!) colon so he could actually poop right. |
20:57.27 | J4k3 | he was using a colostomy bag for a couple years. |
20:57.33 | linagee | J4k3: yikes |
20:57.35 | putnopvut | Didn't that guy with Cheney get shot in the face with a shotgun? |
20:57.38 | J4k3 | yeah. |
20:57.43 | J4k3 | putnopvut: yeah, from a HUGE distance. |
20:57.46 | J4k3 | like 30 meters. |
20:57.47 | shido6 | thats the kind of day that must have been great |
20:57.49 | J4k3 | with an open choke shotgun |
20:57.54 | shido6 | talking to his 5 yr old son |
20:58.01 | shido6 | ... dad why are you so happy? |
20:58.21 | shido6 | (fill in the blank) |
20:58.24 | J4k3 | haha |
20:58.28 | J4k3 | ever seen Bad Santa? |
20:58.54 | shido6 | not yet |
21:00.09 | [TK]D-Fender | shido6, open choke is great.... if you're shooting slugs or flechettes :) |
21:01.20 | *** join/#asterisk ZackZ (n=zzumbaug@70.244.109.129) |
21:03.21 | ZackZ | If I have a PRI with 9 bchannels, how should I make my dialplan so it can rollover to the next available number when it dials out? |
21:04.03 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
21:04.08 | Qwell | ZackZ: look up zaptel groups |
21:04.09 | *** join/#asterisk tomcontr3 (n=tomcontr@92-175-28.dial.terra.cl) |
21:04.36 | tomcontr3 | Hi, Im still having problems with my TDM400P + 2FXO Card.... When I call I hear like static noise... |
21:04.48 | *** part/#asterisk ZackZ (n=zzumbaug@70.244.109.129) |
21:05.05 | [TK]D-Fender | tomcontr3, Have you lowered your gains to 0.0 ? |
21:05.09 | tomcontr3 | yep |
21:05.18 | tomcontr3 | I even set it to -5.0 |
21:05.18 | [TK]D-Fender | tomcontr3, no improvement? |
21:05.23 | tomcontr3 | but same story |
21:06.33 | tzafrir_home | tomcontr3, and you say that a simple analog phone performs well there, so it is not an issue with the line, right? |
21:06.33 | tomcontr3 | I thought it could be the computer where it was installed, because it was a little old (P3) so I change it to a Core Duo2 Server. But the noise was still there |
21:06.53 | tomcontr3 | correct |
21:07.01 | tzafrir_home | Do you have any other FXO interface? an FXS module, some ATA, whatever? |
21:07.08 | tomcontr3 | if I plug the phone directly to the phone line there is not noise |
21:07.19 | tzafrir_home | one or two channels is more than enough even for a P3 server |
21:07.25 | tomcontr3 | no, just the TDM400P with 2 FXO modules |
21:07.33 | hmmhesays | shit I ran 2 sip channels on a p233 |
21:08.53 | tzafrir_home | tomcontr3, can you try removing the echo canceller for a while, to eliminate it as a source for any mishaps? |
21:09.13 | tomcontr3 | this is my zapata.conf |
21:09.13 | tomcontr3 | http://pastebin.ca/697957 |
21:09.59 | ManxPower | Do you have the HPEC? |
21:10.09 | ManxPower | Also, if your gains are too high you can get ECFO |
21:10.12 | ManxPower | ~ecfo |
21:10.12 | jbot | Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly ... |
21:11.52 | tomcontr3 | ManxPower, what is HPEC? |
21:11.53 | tzafrir_home | yeah, only the gains are zero and below |
21:12.07 | tomcontr3 | you can check my zapata.conf... my gains are set to 1.0 |
21:12.48 | tzafrir_home | echocancel=64 ? |
21:12.49 | russellb | tomcontr3: a software echo canceller ... you'd know if you had it :) |
21:12.50 | tzafrir_home | That's low |
21:13.21 | tzafrir_home | anyway, what happens if you remove the echo canceller? echocancel=no |
21:13.33 | tomcontr3 | same thing.. |
21:13.38 | tomcontr3 | I try it |
21:14.11 | tomcontr3 | russellb, I have sip phones... I dont use softphones. |
21:14.15 | tzafrir_home | make sure that this is applied. You should see in 'zap show channel NNN' in the middle of a call that the echo canceller has "0 taps". |
21:15.38 | tomcontr3 | let me check |
21:15.42 | ManxPower | tomcontr3: professional quality Digium software echo canceler |
21:15.47 | [TK]D-Fender | he has STATIC, not ECHO |
21:16.05 | [TK]D-Fender | General craapy audio quality |
21:16.15 | ManxPower | [TK]D-Fender: I only mention that because of ECFO issues I had. |
21:16.25 | ManxPower | Users report it as "static" |
21:17.05 | [TK]D-Fender | tomcontr3, freebie test : disable EC. |
21:17.18 | tomcontr3 | I dont have the echo problem, I dont hear my self with delay,.... I hear a sound like if a were at the beach |
21:17.23 | *** join/#asterisk arguile (n=arguile@KTNRON06-1242488957.sdsl.bell.ca) |
21:17.38 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
21:18.31 | ManxPower | I give up. |
21:18.44 | waverly360 | ManxPower: give up on what? |
21:18.54 | ManxPower | tomcontr3: regardless of what you think, you will do better to follow [TK]D-Fender's advice. |
21:19.09 | waverly360 | Oooh..I must've missed something. |
21:19.12 | tzafrir_home | On our devices I heard such things from devices that were not properly grounded. But I'm not sure how this applies to a PCI card in the computer's case |
21:19.41 | tomcontr3 | you mean echocancel: no ? |
21:19.46 | tzafrir_home | right |
21:20.15 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-96-108.dsl.peoril.sbcglobal.net) |
21:20.26 | [TK]D-Fender | tomcontr3, If it improves then there's part or potentially all of your problem. |
21:20.57 | [TK]D-Fender | scientificmethod++ |
21:21.35 | *** part/#asterisk smace (n=chatzill@200.220.198.107) |
21:22.36 | MACscr | Anyone have any quick tips for using a single polycom 501 on an asterisk system using SIP? The asterisk system is remote and polycom is behind NAT. There is no stun server. I have forwarded ports on the polycom network, but i still cant even see any proof that the polycom is even hitting the asterisk server at all |
21:22.52 | tomcontr3 | no, now I have echo AND the static noice |
21:23.04 | MACscr | Heck, my cheap grandstream gxp2k was easier to setup that the polycom. Thats sad |
21:23.20 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
21:23.30 | rantsh | Hey people |
21:23.38 | rantsh | How is everything? |
21:24.01 | rantsh | I keep getting a problem with a queue since the other day |
21:24.14 | tomcontr3 | this is quite disapointing, because Digium products are no very cheap ... |
21:24.19 | MACscr | I hate to go through a bunch of bulk phone setup procedures if im only setting up one phone |
21:24.44 | rantsh | I got help from nicchap here and decided to upgrade my asterisk 1.2.3 to 1.2.24 (on a test environment of course) |
21:25.31 | tzafrir_home | tomcontr3, well, at least you know that the EC is effective :-( |
21:25.59 | [TK]D-Fender | MACscr, here : |
21:26.01 | [TK]D-Fender | ~sipnat |
21:26.02 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
21:26.10 | rantsh | I posted my configuration here http://pastebin.com/m20ed3434 |
21:26.43 | rantsh | and still, I keep getting asterisk accept callers in the queue when there are no agents logged in |
21:26.46 | tzafrir_home | MACscr, if just the phone is behine NAT, then just set nat=yes in the phone's entry in sip.conf . |
21:27.09 | rantsh | further more it rings one of the agents direct number even if he's not logged in |
21:27.15 | tzafrir_home | No need to forward posrts and such |
21:27.18 | rantsh | any ideas what could be going on? |
21:27.21 | MACscr | Tzafrir_home : yes, its just the phone. |
21:27.39 | tzafrir_home | Unless your packets travel through an aggressive NAT router |
21:27.55 | tzafrir_home | that mangles SIP headers on its own |
21:28.13 | MACscr | I still do not see why i cant see any indications in the asterisk cli that the phone is even attempting to contact the server |
21:28.39 | tzafrir_home | MACscr, have you enabled SIP debug? |
21:28.45 | MACscr | yes |
21:29.15 | tzafrir_home | next step: a sniffer (tcpdump / wireshark / whatever) |
21:29.32 | MACscr | Ah man, that type of stuff is a bit over my head. grr |
21:29.41 | rantsh | this is the ooutput in the * cli ... http://pastebin.com/d68392573 |
21:29.43 | [TK]D-Fender | MACscr, for your phone's sip.conf entry : "nat=yes", "canreinvite=no", and do NOT forward any ports to it. Describe your * side now... |
21:29.48 | tomcontr3 | tzafrir_home, something is something, but the this is that I have to make this work here ate the office,... and we bought that card, becuase we thought Digium was good |
21:30.11 | tomcontr3 | but anyway |
21:30.24 | tomcontr3 | I will have t wait and see that the support guys say |
21:30.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
21:33.34 | rob0 | IIRC, Digium includes support to get a device running. They did for us, even later when we went back begging for help. |
21:33.45 | rantsh | can anyone throw me a hand here? |
21:33.51 | rob0 | but they're probably closed for the weekend. |
21:34.42 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
21:36.08 | Qwell | rob0: still got another 2.5 hours, I believe |
21:36.35 | rob0 | oh, ha. |
21:36.52 | rob0 | Serves 'em right for sleeping in so long. |
21:37.59 | *** join/#asterisk jingles (n=foo@39.183.dowl.anc.borealisbroadband.net) |
21:38.20 | [TK]D-Fender | rantsh, pastebin "show agents" and "show queues" |
21:39.04 | rantsh | [TK]D-Fender: right away |
21:40.15 | rantsh | [TK]D-Fender: http://pastebin.com/d27c4e157 |
21:41.20 | rantsh | [TK]D-Fender: I don't understand why it keeps ringing 9923181 :s |
21:41.33 | [TK]D-Fender | rantsh, SIP/9923181 is available and because of that new callers don't get kicked |
21:41.55 | rantsh | [TK]D-Fender: but it's not declared as a member anymore |
21:42.03 | [TK]D-Fender | rantsh, *I* see it there... |
21:42.17 | [TK]D-Fender | SIP/9923181 (dynamic) (Not in use) has taken 2 calls (last was 1382 secs ago) |
21:42.23 | rantsh | I did change agents.conf to use 1001,1002,1003 |
21:42.44 | [TK]D-Fender | rantsh, doesnt' matter, that is a DYNAMICALLY added memeber based on that now disabled line from your dialplan. |
21:42.53 | rantsh | I know, I saw it too; I can't understand why he's a member |
21:42.57 | [TK]D-Fender | rantsh, You should use removequeuember it. |
21:43.20 | [TK]D-Fender | rantsh, ;exten => 56446,1,AddQueueMember(queue1|SIP/${CALLERIDNUM}) <-------- this was why |
21:43.38 | [TK]D-Fender | rantsh, You did this, then commented out the dialplan that added them in the first place |
21:43.51 | rantsh | :o |
21:44.18 | ManxPower | Well there's a few mins of [TK]D-Fender's life he will never get back |
21:44.21 | rantsh | [TK]D-Fender: so it keeps remembering him as a member even though I've reloaded and restart many (MANY) times? |
21:44.39 | [TK]D-Fender | rantsh, yup |
21:44.47 | [TK]D-Fender | ManxPower, Cry for time lost |
21:44.49 | rantsh | [TK]D-Fender: :s that sucks |
21:45.15 | [TK]D-Fender | rantsh, jsut think if you reloaded to take other changes and kept kicking out your agents! |
21:45.21 | [TK]D-Fender | rantsh, This is a FEATURE! |
21:45.52 | rantsh | so I should use removequeuemember in the dialplan to remove this agent, then scrape it out, right? |
21:45.58 | rantsh | [TK]D-Fender: I see your point |
21:46.02 | [TK]D-Fender | rantsh, yup |
21:46.26 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
21:46.43 | rantsh | [TK]D-Fender: I'll try that [TK]D-Fender, thanks for your help again |
21:49.54 | *** part/#asterisk InsomniaCity (n=insomnia@raptor.ukc.ac.uk) |
21:51.37 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
21:52.07 | rantsh | [TK]D-Fender THANKS MAN IT WORKED |
21:52.31 | rantsh | [TK]D-Fender: You're the best dude |
21:52.59 | [TK]D-Fender | rantsh, you're welcome |
21:53.18 | *** part/#asterisk TicoTuco (n=matheus@200.250.100.25) |
21:56.40 | _x86_ | HAHAHA! SCO files for bankruptcy! |
21:56.49 | _x86_ | "on the eve of the Novell trial" ;) |
21:56.53 | _x86_ | oh man this is awesome |
21:57.09 | Qwell | how is it awesome? |
21:57.17 | Qwell | It means Novell doesn't get anything for a while now |
21:58.03 | _x86_ | wasn't SCO the one chasing Novell for damages? |
21:58.18 | _x86_ | IBM was the one chasing SCO |
21:58.24 | _x86_ | iirc |
21:59.05 | Qwell | novell has counterclaims |
22:00.19 | _x86_ | ah |
22:03.05 | *** join/#asterisk jsidhu (n=atomik@66.206.163.185) |
22:06.27 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
22:06.37 | rantsh | I'm pretty sure anyone can help me with this n00b question |
22:06.52 | jsidhu | i need some help with setting up FXS. I have an analog DID line, which uses Wink Start, OutPulse 4 digits DTMF. How can I configure my TDM400's FXS module to work with this? I'd appreciate it if someone could point me in the right direction.. I'm sort of lost reading all kinds of different examples.. |
22:06.54 | [TK]D-Fender | NEXT!!@!@@!@ (c) BKW |
22:07.15 | rantsh | I want to play a message when an extension is busy/disconnected, how can I set it up? |
22:07.19 | rantsh | I tried: |
22:07.44 | rantsh | exten => 2050,105,Playback(tt-monkeys) ... but it didn't seem to work |
22:07.55 | tzafrir_home | jsidhu, FXS module? |
22:07.55 | [TK]D-Fender | jsidhu, if its raw analog, Answer the line and dump the call immediately into an IVR with no prompting and a XXXX pattern to trap the DID |
22:08.10 | [TK]D-Fender | jsidhu, and that should be FXO |
22:08.20 | jsidhu | FXO for an incoming DID line? |
22:08.24 | jsidhu | ok |
22:08.31 | rantsh | btw, 4 is a Queue cmd |
22:08.37 | jsidhu | good thing i ordered the card with a fxo and fxs |
22:08.41 | [TK]D-Fender | jsidhu, And as a fallback use "i" and "t" to default to whatever you'd consider your primary DID. |
22:08.46 | jsidhu | how do i deal with the wink? |
22:09.08 | jsidhu | i need to send a wink to get a dial tone on the line, its not a PSTN line |
22:09.22 | [TK]D-Fender | rantsh, it doesn't priority jump. just make it #5 |
22:09.37 | jsidhu | err.. i mean its not a regular phone line, its an analog incoming did line, that i need to send a wink on to get it to work |
22:09.43 | jsidhu | hopefuly im making some sense |
22:09.43 | rantsh | ok |
22:10.46 | [TK]D-Fender | jsidhu, then right after you answer I think you should call "Flash" to "wink" the line |
22:13.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
22:13.15 | jsidhu | TK: i need to wink the line before I can answer |
22:13.40 | jsidhu | if I dont wink, I wont get a dialtone |
22:13.54 | jsidhu | no dialtone, no calls incoming that can be answered |
22:13.56 | jsidhu | :/ |
22:16.13 | *** part/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
22:17.51 | *** join/#asterisk NirS (n=NirS@84.94.145.166.cable.012.net.il) |
22:17.58 | NirS | good evening everybody |
22:18.34 | NirS | anybody home ? |
22:19.11 | *** join/#asterisk jsidhu2 (n=atomik@66.206.163.184) |
22:19.20 | jsidhu2 | sorry, i timed out after my last reply |
22:19.27 | jsidhu2 | any ideas td |
22:20.09 | tzafrir_home | NirS, hi |
22:20.18 | NirS | hey tzafrir, wassup ? |
22:20.59 | NirS | I'm having some really funky IAX2 problems over here |
22:21.03 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
22:21.24 | NirS | keep getting INVALs for something that works in another location no problem |
22:23.16 | [TK]D-Fender | jsidhu : whe n it rings, answer, then issue the wink to accept the call, then dump into IVR as I mentioned |
22:23.37 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-7f8666313f0c626f) |
22:23.38 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
22:25.27 | NirS | anyone has a clue for this with DIDx ? |
22:25.27 | jsidhu2 | i think u misunderstand me, it will not ring unless I send a WINK on the line to tell whatever's on my telco's side that the line is hooked up to a PBX, when that wink is sent, I will get a dialtone and only then can calls come to that line. |
22:25.28 | NirS | Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL |
22:26.57 | [TK]D-Fender | jsidhuit sounds like you are mixing up your wording for an INCOMING call and that of an OUTGOING call. |
22:27.25 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
22:27.46 | jsidhu2 | i dont think so, why do you say that |
22:27.56 | jsidhu2 | this line that I speak of is not a regular analog phone line. |
22:28.17 | jsidhu2 | this is a DID analog line, which I must FIRST send a wink to, sending this wink activates the line. |
22:29.12 | [TK]D-Fender | jsidhu2, ok, WHEN do you send this wink, and what happens NEXT? |
22:29.38 | tzafrir_home | NirS, I really don't know IAX2 that well, but can you provide a more complete trace? |
22:29.47 | jsidhu2 | I have to send this wink to "turn the line on". when that wink is send, then the telco side will start sending me calls on this line. |
22:29.52 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
22:29.56 | *** join/#asterisk iPod-nano (n=iPod-nan@c-68-43-60-193.hsd1.mi.comcast.net) |
22:30.06 | NirS | that's exactly it, there is no more trace |
22:30.14 | iPod-nano | Where are the config files located on a Debian system? |
22:30.27 | NirS | I wish I could see that the request coming in from DidX is correct, but I can't |
22:30.39 | tzafrir_home | iPod-nano, /etc/<package-name>, and thus /etc/asterisk |
22:31.21 | tzafrir_home | which also happens to be the same sane choice made by Asterisk :-) |
22:31.53 | iPod-nano | Yeah, I'm trying to run my first Asterisk machine. |
22:32.53 | iPod-nano | I'll be surprised if the ancient computer I'm using will even succeed. |
22:33.44 | tzafrir_home | What computer is that? |
22:34.01 | iPod-nano | Um... old. |
22:34.04 | iPod-nano | Old Compaq. |
22:34.12 | tzafrir_home | CPU? memory? |
22:34.13 | iPod-nano | That surprisingly can handle Debian. |
22:34.21 | jsidhu2 | .. |
22:34.35 | iPod-nano | What does 60416 KB equate to? |
22:34.44 | jsidhu2 | 60mB |
22:34.46 | tzafrir_home | 64MB? |
22:35.02 | [TK]D-Fender | GTG, back in many hours :) |
22:35.06 | jsidhu2 | 65536=64mg |
22:35.13 | jsidhu2 | b |
22:35.15 | iPod-nano | And no clue how to make this thing tell me what speed its processor is. |
22:35.18 | tzafrir_home | minus some overhead |
22:35.29 | tzafrir_home | iPod-nano, cat /proc/cpuinfo |
22:36.27 | *** part/#asterisk Netgeeks-laptop (n=chris@204.11.231.198.static.etheric.net) |
22:36.32 | iPod-nano | I don't know enough about Linux. :-P |
22:37.17 | iPod-nano | Wow, 225.030 MHz |
22:37.24 | iPod-nano | This thing is fast! :-D |
22:37.29 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
22:37.31 | iPod-nano | I was expecting a 133. |
22:37.34 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
22:38.40 | jsidhu2 | crap. the TDM400p doesnt support Analog DID Lines?? |
22:39.38 | iPod-nano | Yeah, it;s old. |
22:39.50 | iPod-nano | That's why it's been reduced to a console-only server. |
22:40.30 | tzafrir_home | jsidhu2, technically it is not the TDM400P that doesn't support it. It is Zaptel (from what I understand) |
22:41.16 | jsidhu2 | i see, do you know of a way to use such Analog DID trunks with Asterisk? |
22:41.30 | tzafrir_home | iPod-nano, a really borderline computer. Don't expect ot get too much from it. May make a nice toy PBX |
22:42.02 | iPod-nano | It's not going to be controlling more than one or two clients. |
22:42.10 | jsidhu2 | it should be ok for 1 or 2 |
22:42.35 | iPod-nano | And for me it is just a toy, anyway. |
22:42.43 | iPod-nano | The whole point is to teach myself. |
22:43.10 | iPod-nano | And then I might upgrade to my screaming fast 500 MHz box. |
22:43.31 | tzafrir_home | You also need there more memory |
22:45.30 | UnixDog | why 500 mhz is more then enough |
22:45.43 | UnixDog | thats a kickass pbx box |
22:45.47 | iPod-nano | Yeah, but that's not the machine I'm using. |
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22:46.10 | UnixDog | 500 mhz/256 megram and a laptop drive |
22:46.20 | UnixDog | dood that be a rocking box |
22:46.21 | iPod-nano | I'm gonna start off simple with a computer that's useless for any other purpose. |
22:46.32 | UnixDog | a p2 266 |
22:46.41 | UnixDog | or a p1 233 |
22:46.55 | iPod-nano | Pentium I. |
22:47.07 | UnixDog | and what distro |
22:47.18 | iPod-nano | It was running Windows 98. I gutted it and installed Debian. |
22:47.27 | UnixDog | ok |
22:48.03 | UnixDog | what if I could give you a quick bsdinstall and pkgs that install everythign and have you up in less time then building on deb |
22:48.43 | UnixDog | + the digium gui |
22:48.54 | UnixDog | or my gui I work with |
22:49.14 | iPod-nano | GUI is practically out of the question with this machine. |
22:49.40 | UnixDog | why I run the asterisk-gui on a 486/100 |
22:49.46 | UnixDog | and its fine |
22:49.57 | UnixDog | it uses the asterisk builtin httpd server |
22:51.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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22:56.50 | watchy | anyone here do vlans with linux |
22:57.06 | drako | UnixDog, whats so good with astersk-gui? |
22:59.24 | UnixDog | I use it just to save time |
22:59.44 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
22:59.44 | UnixDog | ask digium whats so good about it |
23:00.01 | UnixDog | they have taken over a year to get ti tpo where is it now |
23:00.11 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
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23:00.25 | MACscr | This channel is way to anti gui, its completely rediculous and ignorant |
23:00.59 | iPod-nano | It's not anti-gui so much as it's that GUI isn't an option for some of us. |
23:01.12 | MACscr | That doesnt make it bad |
23:01.34 | MACscr | Ipod-nano : many people in here are Anti-gui, not matter how you spin it |
23:01.39 | outtolunc | or maybe it isn't anti-gui it it is ANTI support for guis that do not support themselves <G> |
23:02.31 | UnixDog | the big issue on guis is the fact they all use diff files |
23:02.38 | iPod-nano | I couldn't live if I had to live only in the console. |
23:02.48 | UnixDog | and only 2 I know of even have fully functional dialplans |
23:02.55 | iPod-nano | But, GUIs take more resources, it's that simple. |
23:03.20 | adorah | well, resources are dirt cheap these days.. |
23:03.26 | iPod-nano | Also, if your administering a remote machine over a network, graphics are a very bad idea. |
23:03.42 | iPod-nano | I'd rather ssh into my ancient machine. |
23:04.13 | iPod-nano | Me, I try to find uses for older computers. |
23:04.35 | UnixDog | but the big fact I still stand by this day. no one has put a fully functional dialplan out there for asterisk |
23:04.49 | adorah | when one get for ~200$ a machine that can handle E1 with 60 users, resources are not an issue.. |
23:04.55 | UnixDog | and there for using asterisk out of the box is not functional |
23:05.02 | [hC] | any of you guys using an aastra 57i with expansion modules & BLF? |
23:05.16 | UnixDog | BLF in asterisk sucks |
23:05.23 | UnixDog | its still in need of work |
23:05.35 | [hC] | yeah no kidding, that doesnt change the fact that people need to use it :) |
23:05.45 | [hC] | i wonder how much better it is in 1.4, or if it even got any attention |
23:05.45 | rob0 | Some of us are proud of being ridiculous and ignorant! |
23:05.48 | UnixDog | need want |
23:06.16 | [hC] | UnixDog: i dont really view BLF as a want. if customers want it, that means i need it. :) |
23:06.27 | rob0 | But the funny thing is when someone who needs help calls the people he wants to help ridiculous and ignorant. ;) |
23:06.35 | UnixDog | well its time to shove the stick up asterisk groups ass and get them to understand that there needs to be a fully functional Dialplan |
23:06.47 | adorah | customers demand BLF |
23:06.48 | rob0 | Hey all you R&I people, I need help! ;) |
23:06.55 | rob0 | Here |
23:07.11 | [hC] | It works, but on this aastra it likes to crash and lock up. not sure if its asterisk blf, or the phone, or what |
23:07.17 | rob0 | 's the thing: my * server at home is on lousy power and Internet service ... |
23:07.20 | [hC] | Looking for someone else doing it to hear their stories. |
23:07.43 | rob0 | ... and I'm not home much (I work ~500km away from home) |
23:08.09 | rob0 | ... I got it so it comes back on when the power is restored. |
23:08.42 | rob0 | ... But: when the ISP is down and it comes on, I don't get online, and * doesn't like that either. |
23:09.01 | rob0 | How can I make sure * will start even without an Internet connection? |
23:09.42 | rob0 | See, I have an FXO there so the POTS line should be picked up. |
23:10.02 | UnixDog | it should install a startupscript in /etc/rc,\.d |
23:10.08 | UnixDog | it should install a startupscript in /etc/rc.d |
23:10.30 | UnixDog | if not you can always start it at the bottom of rc.local |
23:10.39 | UnixDog | safe_asterisk |
23:11.18 | rob0 | I do this. It DID try to start at boot time. But it barfed because no Internet probably broke the SIP and IAX channels. |
23:11.31 | UnixDog | ? |
23:11.40 | rob0 | (I am guessing, I have no way of knowing what it said) |
23:11.41 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
23:11.55 | UnixDog | thats sounds funky since asterisk does not require a internetconnection to start |
23:12.19 | rob0 | some of my channels do |
23:12.24 | dan__t | Anyone use Teliax by chance? |
23:12.36 | dan__t | I have a pretty dumb question, but I can't find the answer heh. |
23:12.39 | rob0 | yeah, I was surprised, the machine was up and running when I got here, but * was not. |
23:12.50 | dan__t | I guess I'm wondering if outbound calls are supported through the plans |
23:13.04 | dan__t | To make sure that its just a case of me doing something dumb when configuring |
23:13.13 | rob0 | I think Teliax is an outbound provider ... |
23:13.29 | dan__t | Is it uncommon to have a provider that only does inbound? |
23:14.18 | UnixDog | they only do trunking |
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23:14.29 | UnixDog | they offer no functionality |
23:14.29 | dan__t | I suspect that's all I need. |
23:14.40 | UnixDog | there are other providers |
23:14.41 | dan__t | I've had the account for like two years and just finally started using it. |
23:15.01 | dan__t | I recall choosing it because they offered IAX2, and if I recall further, I believe that was not common back in the day? |
23:16.03 | UnixDog | nufone |
23:16.07 | UnixDog | they do iax |
23:16.20 | UnixDog | fwd has always offerd iax |
23:16.27 | rob0 | dan__t: I'm just a small-timer, but for inbound I use Stanaphone and ipkall -- both are free. |
23:16.49 | dan__t | Tryin' to rig a toll-free number out of the deal, too |
23:18.31 | dan__t | Welp, for paid IAX trunking, with the abiity to get a toll free number at decent rates, does anyone have any suggestions? |
23:18.40 | dan__t | I hate to poll like this, but I just want to see what's out there :) |
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23:28.34 | rob0 | dan__t: Asterlink maybe, I use them for outbound |
23:29.19 | rob0 | I never did get the inbound toll-free number working, but I don't really need it. |
23:29.23 | dan__t | ahh |
23:30.07 | *** join/#asterisk [ProB]CrazyMan (n=niethamm@pd907f938.dip0.t-ipconnect.de) |
23:30.31 | [ProB]CrazyMan | hello |
23:30.51 | dan__t | hi |
23:32.37 | giesen | how the hell do you do a catchall extension |
23:32.50 | giesen | it's driving me nuts |
23:32.59 | *** join/#asterisk Joneser (n=Joneser@pool-71-170-201-50.dllstx.fios.verizon.net) |
23:33.39 | [ProB]CrazyMan | I have to upgrade my asterisk, therefor I also need to update spandsp (app_rxfax.so and app_txfax). does anybody know which spandsp version is the one whch is stable for asterisk 1.2.23 ? |
23:37.22 | blitzrage | giesen: _. |
23:37.31 | file | [TK]D-Fender: You are here. |
23:37.38 | blitzrage | file: you are there |
23:37.47 | file | I am. |
23:37.51 | blitzrage | False. |
23:40.33 | *** join/#asterisk Poehali (n=actionma@74.93.5.186) |
23:40.39 | Poehali | hey people! I got it working! |
23:40.39 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
23:41.06 | Poehali | took me an entire week of trial and error |
23:41.17 | Poehali | and I tink I broke the box in the process |
23:41.54 | codefreeze | It's $180 for 1 OEM copy of Windows Vista Ultimate. (newegg). Am I a fool? What's so hot about ultimate vs. home vs. business? |
23:42.08 | file | features and licensing |
23:42.13 | file | although OEM... |
23:43.00 | TJNII | Anyone have a budgetone 100? |
23:46.15 | TJNII | Nm, I found the option that was horking it. |
23:47.37 | MACscr | Does anyone know if i can configure the voicemail button on a polycom 501 with its gui? |
23:49.43 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
23:50.03 | codefreeze | I'm using #asterisk for a filter! |
23:50.42 | codefreeze | I make a simple statement, like "I found a budgetone 100 for $579.99! Is that a great deal or what?" |
23:51.13 | codefreeze | And if nobody calls me an idiot, or names a cheaper price, or offers me a bridge, |
23:51.31 | codefreeze | it must be somewhere near a fair deal. |
23:52.37 | MACscr | Lol, your an idiot |
23:53.38 | codefreeze | Either that, or everyone is so stupefied, as to not know what to say! :) |
23:56.00 | codefreeze | MACscr: lol, that's a given! |
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23:59.22 | dan__t | orly? |
23:59.31 | dan__t | rob0, haven't seen you around in quite some time heheh |
23:59.44 | rob0 | I haven't been in here. |
23:59.52 | rob0 | (this channel) |