| 00:02.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:03.56 | riddlebox | is there a phone that I can have line appearance's on? A customer wants to use asterisk but they really like to see the individual lines and when they are in use, they dont care about seeing who is on the phone just if all lines are busy |
| 00:04.26 | defsdoor | a phone with blf ? |
| 00:04.27 | ManxPower | riddlebox: in theory yes, with 1.7 |
| 00:04.34 | ManxPower | ..er.. 1.6 |
| 00:04.47 | ManxPower | defsdoor: not BLF as in Asterisk |
| 00:04.56 | riddlebox | ManxPower, really |
| 00:05.09 | ManxPower | Asterisk's BLF relies on the phone notifying Asterisk, and since we are talking about phones, not lines. |
| 00:05.25 | ManxPower | UNLESS there is something in 1.4's SLA stuff that I don't know about (and I don't know much about 1.4's SLA) |
| 00:05.36 | defsdoor | aah - I thought he meant extensions not trunks |
| 00:05.36 | jblack | riddlebox: you should be able to, yes. asterisk -r w/ "show channels" will count open channels. |
| 00:05.52 | jblack | subtract from that the number of channels there are, and you know what's left over. |
| 00:06.31 | ManxPower | jblack: How would you get that into a phone's BLF? |
| 00:06.41 | jblack | I don't know what a BLF is. |
| 00:06.41 | riddlebox | defsdoor, not blf, they want to see the line, and press the button and call out from that line |
| 00:06.48 | riddlebox | jblack, busy lamp field |
| 00:06.59 | defsdoor | riddlebox: that's a little antiquated isn't it ? |
| 00:07.11 | riddlebox | defsdoor, thats what the customer wants |
| 00:07.17 | jblack | No idea. |
| 00:07.23 | Shaun2222 | ManxPower: hmm, looks like if i dont specify a return after set(GOSUB_RESULT=CONTINUE) it actually does what it should... |
| 00:07.24 | mosty | riddlebox, what kind of lines? |
| 00:07.39 | riddlebox | mosty, fxo ports |
| 00:08.10 | Shaun2222 | ehh maybe not. |
| 00:08.18 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 00:08.28 | Shaun2222 | is Auth fallthrough a good thing... |
| 00:08.38 | Shaun2222 | i see it checking the status of it alot. |
| 00:08.54 | mosty | riddlebox, can't you setup hints for zap channels? |
| 00:09.19 | SteveTotaro | ~trunk |
| 00:09.20 | jbot | [trunk] is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 00:09.26 | mosty | exten => 1,hint,ZAP/1 ; for example |
| 00:11.28 | jblack | jbot: I do not have a SIP Trunk, so there. |
| 00:11.28 | jbot | You do not have a SIP Trunk, so there.? |
| 00:11.35 | jblack | That's right. |
| 00:11.54 | riddlebox | mosty, I will try that |
| 00:12.21 | riddlebox | but then what would put for the button? |
| 00:13.00 | mosty | riddlebox, same as for a sip hint, but use the number you picked for the "extension" of the hint |
| 00:13.46 | SteveTotaro | ~ |
| 00:14.19 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-87-237.lns10.syd6.internode.on.net) |
| 00:15.05 | drmessano | jbot: I glue SIP together by hand |
| 00:15.06 | jbot | You glue SIP together by hand? |
| 00:15.11 | drmessano | jbot: yes |
| 00:15.12 | jbot | You don't say! |
| 00:15.18 | drmessano | jbot: Say I do! |
| 00:15.19 | jbot | I do! |
| 00:15.29 | drmessano | jbot: Say my name |
| 00:15.30 | jbot | my name |
| 00:15.40 | adeel | is there any way to get the colorized output from the * cli without having to set the -c option on startup? setting -c for some reason sends *'s cpu usage to 100% |
| 00:15.42 | drmessano | jbot: Say ban me |
| 00:15.42 | jbot | ban me |
| 00:15.42 | jblack | jbot: smack drmessano |
| 00:15.43 | jbot | ACTION smacks drmessano upside the head. |
| 00:16.11 | jblack | You deserved that for fondling the bot too much. =) |
| 00:16.14 | drmessano | jbot: kick jblack |
| 00:16.15 | jbot | ACTION kicks jblack |
| 00:16.20 | drmessano | na na |
| 00:16.47 | drmessano | jbot: drop database; |
| 00:16.55 | drmessano | oops |
| 00:17.15 | drmessano | jbot: drop table asterisk; |
| 00:17.46 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 00:18.26 | ManxPower | There's nothing wrong with playing with the bot, as long as you do it in private and wash your hands after. |
| 00:18.59 | drmessano | lol |
| 00:19.11 | drmessano | jbot: my son's name is robert'); DROP TABLE Students;-- |
| 00:19.12 | jbot | drmessano: okay |
| 00:19.42 | drmessano | jbot: we call him "little bobby tables" |
| 00:24.24 | [hC] | this is messed up. somehow a user has a message in their old folder (asterisk 1.4.14) that is 0 bytes, and when asterisk tries to play it, app_voicemail exits quietly and hangs up the call |
| 00:25.09 | [hC] | (voicemail, that is) |
| 00:25.17 | jblack | [hC]: hmm. |
| 00:27.11 | [hC] | and of course you cant just delete msg0000.* |
| 00:27.19 | [hC] | you need to renumber everyone after it one less. |
| 00:27.22 | [hC] | what a pita. |
| 00:28.05 | mosty | i always found it annoying that when voicemail was emailed, it's always called msg0000.wav |
| 00:28.55 | defsdoor | [hC]: simple script |
| 00:29.48 | jblack | untested, but probably close: |
| 00:29.50 | jblack | for x in `seq -w 1 1000`; do mv msg$x.wav msg$[$x-1].wav; done |
| 00:30.42 | jblack | That's not right. But it is close. |
| 00:30.42 | SteveTotaro | jblack, there is a bridge app that can bridge arbitrary channels |
| 00:31.18 | jblack | SteveTotaro: Yeah, that's right. |
| 00:31.22 | SteveTotaro | i was following it when it was a patch sometime ago, not sure if it made it into any versions yet |
| 00:31.58 | lmadsen | SteveTotaro: it's in trunk |
| 00:32.01 | lmadsen | and works great |
| 00:32.04 | lmadsen | I've done lots of testing iwth it |
| 00:32.11 | lmadsen | I've even got a backport for 1.4 (not supported) |
| 00:32.36 | SteveTotaro | that is awesome, maybe i need to upgrade, that opens alot of possibilities |
| 00:33.06 | SteveTotaro | or at the very least setup a box to use when i want that bridging capability |
| 00:33.14 | jblack | I can imagine a Dial(target,B(target2))... or a complementary g option to go with G |
| 00:33.26 | SteveTotaro | how can i get the backport to 1.4? |
| 00:33.34 | jblack | actually, G is pretty darn close. |
| 00:33.41 | defsdoor | jblack: that's aweful |
| 00:34.52 | jblack | You have no idea how awful I'm thinking. |
| 00:35.17 | defsdoor | jblack: typeset -Z4 n=0 ; for FN in msg* ; do mv $FN msg${n}.wav ; ((n=n+1)) ; done |
| 00:35.35 | jblack | oh, the seq. |
| 00:35.38 | defsdoor | :p |
| 00:35.42 | jblack | Yeah. What you're suggesting does look better. |
| 00:36.10 | defsdoor | only works in ksh - bash doesn't support typeset -Z |
| 00:36.22 | defsdoor | bed& |
| 00:36.47 | SteveTotaro | ./msg lmadsen would you share your backport with me for bridging? |
| 00:37.06 | jblack | SteveTotaro: It's actually pretty easy to do with G, as I think about it. |
| 00:38.30 | SteveTotaro | cool, i am not up to speed on the G flag |
| 00:40.16 | lmadsen | jblack: ya, but when you want to explicitly link two existing channels together based on dialplan code, that won't work :) |
| 00:41.06 | jblack | Oh, I was thinking "local to X, at X+101, Meetme()" |
| 00:41.34 | lmadsen | X+101... ewww.. old style thinking :) |
| 00:41.37 | lmadsen | SteveTotaro: www.leifmadsen.com/patches/bridge-trunk-rev-49865.patch |
| 00:41.39 | riddlebox | jblack, so far I can get the light to show the zap channel is offhook, now I just gotta figure out how to press the button and be able to dial out of that line |
| 00:42.41 | jblack | I'm thinking something like: http://pastebin.com/m54dffa7a |
| 00:42.47 | lmadsen | SteveTotaro: nothing wrong with keeping the conversation in here -- it's no secret :) |
| 00:42.58 | mosty | riddlebox, you might need one sip account on the phone per zap line |
| 00:43.00 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 00:43.01 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 00:43.19 | mosty | riddlebox, or you could just let asterisk decide which channel to use, which is probably much smarter |
| 00:43.45 | SteveTotaro | i am new to irc, i thought everyone could see it but it was just tagged as addressed to you, still learning |
| 00:43.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 00:43.53 | riddlebox | mosty, for some reason the guys wants to be able to select lines, I told him about call parking and everything |
| 00:43.55 | mosty | riddlebox, you're trying to emulate a previous setup, when there are much better and simpler ways to do it these days :) |
| 00:44.02 | jblack | Slightly simpler: http://pastebin.com/m3eabbe6 |
| 00:44.21 | jblack | bleh, but 101 shoudl be Meetme(600) |
| 00:44.24 | ManxPower | jblack: +101 support was removed in 1.4 |
| 00:44.45 | jblack | http://pastebin.com/m69b73262 |
| 00:44.45 | riddlebox | mosty, lets just say if I can make him happy, there will be alot of money coming my way, from different sites |
| 00:44.52 | jblack | ManxPower: Pardon? |
| 00:44.54 | SteveTotaro | i thought there was an option to turn on +101 |
| 00:45.01 | SteveTotaro | or a patch |
| 00:45.09 | mosty | riddlebox, then i think you will need one sip account per zap line per phone |
| 00:45.14 | jblack | It's in the current show application Dial() instructions |
| 00:45.31 | jblack | For 1.4.17. Do you suppose they're wrong? |
| 00:45.38 | mosty | riddlebox, and that sip account would only be able to dial over the corresponding zap channel |
| 00:45.54 | riddlebox | mosty, I am looking into it |
| 00:46.14 | SteveTotaro | i think you define priorityjumping=yes in the general section of your extensions.conf |
| 00:46.43 | SteveTotaro | it is off by default, i may be wrong though, i mostly use 1.2 |
| 00:47.01 | jblack | I misread G anyways. It's n and n+1, not n+101 |
| 00:47.17 | ManxPower | jblack: Generally a feature is deprecated in one release and removed in the next release. If that is the case with +101, then it is not in 1.4. Perhaps they made an exception for it. |
| 00:47.31 | SteveTotaro | jblack what is G, all i see on the wiki is a lower case g |
| 00:47.57 | ManxPower | SteveTotaro: NEVER EVER use the wiki for application docs. "show application X" is where you should go. |
| 00:48.17 | jblack | I don't see a deprecation warning in the help |
| 00:48.27 | SteveTotaro | i do both but i don't have a 1.4 or 1.6 box up right now |
| 00:48.43 | jblack | I think this would probably work; http://pastebin.com/m6b8bcd6 |
| 00:49.08 | ManxPower | SteveTotaro: the wiki is almost always out of date. |
| 00:49.20 | SteveTotaro | besides, the wiki had a note about queueprio crashing asterisk |
| 00:49.32 | SteveTotaro | so it best to look both places and the mailing list |
| 00:49.48 | jblack | we're not talking queues here, but dial. ;) |
| 00:50.01 | SteveTotaro | i am talking about asterisk |
| 00:50.14 | jblack | I certainly hope so. ;) |
| 00:50.20 | SteveTotaro | so what is the G option? |
| 00:50.34 | jblack | I'll get you a page number for your book |
| 00:50.58 | SteveTotaro | i assume it bridges obviously |
| 00:51.00 | jblack | You can find it on page 384. It's present as late as 1.4.17 |
| 00:56.42 | SteveTotaro | cool thanks, i need to get up to speed on this stuff so when 1.4/6 becomes stable enough for me i can utilize the new features |
| 00:56.42 | jblack | whoah. take a step back tex... |
| 00:56.43 | jblack | You were trying to tell me what exists and doesn't exist, what works and doesn't work, based on 1.2??? |
| 00:56.43 | SteveTotaro | i was not trying to tell you anything |
| 00:56.43 | ManxPower | jblack: I suspect more calls are handled by 1.2 in the world than 1.4 |
| 00:56.43 | SteveTotaro | just what i was following in a patch |
| 00:56.43 | SteveTotaro | that had great implications |
| 00:56.43 | jblack | Well, thanks for caring, then. We're not talking about the agent stuff (which even in latter 1.4 is still seeing a lot of bugfixes) |
| 00:56.43 | hmmhesays | good lord I cannot get this thing to work for the life of me |
| 00:57.10 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:57.44 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 01:00.23 | SteveTotaro | hmmhesays, whats your issue? manx and i can help you, maybe even jblack |
| 01:01.34 | jblack | What's up hmmhesays ? |
| 01:03.27 | *** join/#asterisk RoyKa (n=roy@91.149.35.137) |
| 01:03.50 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 01:10.13 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-67884289dd7813cb) |
| 01:15.10 | *** join/#asterisk andresmujica (n=andresmu@201.244.24.243) |
| 01:19.59 | *** join/#asterisk goodmove (n=yves@209.88.71.201) |
| 01:24.27 | jblack | There's a pile of foul-mouthed brits on 123-4567 with bad lines, though. =) |
| 01:26.20 | *** join/#asterisk r3c4ll (n=r3c4ll@190.142.54.54) |
| 01:27.05 | riddlebox | is there a way to play a dialtone sound so that the person thinks they are hearing the dialtone while they are dialing? |
| 01:27.19 | jblack | There is the r option to dial,but it's pretty evil. |
| 01:27.57 | jblack | ATA adaptors and soft phones typically provide their own dialtone, though |
| 01:28.50 | riddlebox | this is through a sip phone, I have kind of figured out how to turn a button on the phone to a trunk appearance, and you can press it and dial out from it |
| 01:30.01 | jblack | ATA, soft, or what? |
| 01:30.15 | riddlebox | jblack, on a sip phone |
| 01:30.33 | jblack | Let me ask differently. Are you holding a phone in your hands? |
| 01:30.54 | riddlebox | jblack, sorry yeah it is a hard phone, which uses sip |
| 01:31.02 | jblack | Ok, so it's an IP phone. |
| 01:31.11 | jblack | What model? |
| 01:31.20 | riddlebox | Grandstream GXP-2000 |
| 01:31.43 | riddlebox | yeah yeah I know |
| 01:32.00 | riddlebox | everyone says big pile of crap, but they work great in my house |
| 01:32.12 | jblack | i don't know they're crap. Just know the reputation. |
| 01:32.24 | jblack | Gimme a moment to look it up |
| 01:32.46 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 01:33.58 | jblack | http://www.voipinfo.org/wiki/view/GXP-2000 isn't exactly comforting |
| 01:34.25 | jblack | The impression I'm getting is the GXP-2k does provide it's own dialtone, but that it's come and gone with various firmware updates. |
| 01:35.07 | riddlebox | really |
| 01:36.10 | jblack | I'd check the configuration menu for it, to see if you can turn it on explicitely. |
| 01:37.49 | riddlebox | jblack,http://pastebin.ca/869287 I am not sure if that is the best way to do it, but it does work |
| 01:38.38 | jblack | that doesn't make any sense to me. |
| 01:39.34 | jblack | I think you're trying to say 'dial 801, then dial a did as defined by your outgoing extension. |
| 01:40.08 | riddlebox | sorry I didnt show the rest |
| 01:40.17 | jblack | However, I don't see a waitexten here to get the input |
| 01:41.01 | jblack | Why don't you show me your dialplan, and tell me what context your phone is in (which you can get from sip.conf) |
| 01:41.03 | riddlebox | thats what it is I couldnt remember how to get input |
| 01:41.12 | jblack | Ok. I see where you're at. |
| 01:41.19 | riddlebox | http://pastebin.ca/869289 |
| 01:42.36 | jblack | Ok, so now, it looks like you're... trying to connect a pots line to your phone. |
| 01:42.39 | jblack | Correct? |
| 01:42.42 | riddlebox | yup |
| 01:42.46 | jblack | Ok. |
| 01:43.00 | jblack | Go pull up your sip.conf. Tell me the context you're setting for the phone, please |
| 01:43.24 | jblack | And paste your whole dialplan, so that I can completely understand what you've done so far, and where you need to go |
| 01:44.16 | riddlebox | jblack, not to new at it |
| 01:44.43 | jblack | Ok, because the stuff you pastebinned looks like ... well, pardon the expression, babble to me. |
| 01:45.03 | riddlebox | the context is internal, which has an include to my outgoing |
| 01:45.29 | jblack | Ok. Please pastebin your entire extensions.conf, please. |
| 01:46.05 | jblack | Doublecheck that you don't expose any passwords (watch for variables you set in general, and any sip/iax passwords in dial lines) |
| 01:46.32 | riddlebox | hold on |
| 01:46.40 | riddlebox | got two dogs fighting |
| 01:46.48 | jblack | Heh. I'll wait |
| 01:49.37 | riddlebox | the winner is the husky lab mix |
| 01:50.37 | riddlebox | actually jblack, it all works, I just need to find out if I can have it play a dialtone sound, I could always require the users to dial a 9 |
| 01:51.00 | jblack | Ok. Fair enough. |
| 01:51.05 | jblack | Fix your grandstream. ;) |
| 01:51.28 | riddlebox | yeah, hehe or get better phones |
| 01:51.37 | riddlebox | it works for now |
| 01:51.44 | SteveTotaro | american pit bull terrier (red nose from the colby line) is what i have |
| 01:52.37 | jblack | if you're on a seriously limited budget, you can consider linksys ATAs, which let you get a pile of walmart analog phones. |
| 01:53.03 | SteveTotaro | or the digium 24 port analog card |
| 01:53.19 | jblack | Well, digiums tend to be expensive. |
| 01:53.30 | SteveTotaro | grandstream atas are actually not bad |
| 01:53.43 | jblack | going with ATAs can run about $35 + phone per line. |
| 01:54.41 | jblack | SteveTotaro: Glad to hear that _somebody_ likes them. Most of the people with experience with them seem to despise them. |
| 01:55.19 | SteveTotaro | the ATAs are good, only experience with the phones was not so good but that was the bt101 days |
| 01:55.32 | tzanger | SteveTotaro: got your email |
| 01:55.35 | SteveTotaro | they had some nice looking phones on display at Astricon in AZ |
| 01:55.37 | tzanger | (akohlsmith) |
| 01:55.43 | *** join/#asterisk fiXXXerMet (n=meep@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 01:56.22 | SteveTotaro | i think i may be one of the only people that doesn't have a cool alias |
| 01:56.43 | SteveTotaro | tzanger, what do you think? |
| 01:57.48 | jblack | riddlebox: But yeah, you can get really nasty if you want to use a 9 extension hack, doing a waitexten with the m option, with a moh class that has a single dialtone music file. But.. <shudder> |
| 01:57.56 | riddlebox | SteveTotaro, it is illegal to own a pit bull or a mix of a pit bull here |
| 01:57.59 | jblack | [TK]D-Fender: Ping |
| 01:58.14 | ZX81 | ~book |
| 01:58.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:58.27 | SteveTotaro | they try to pass that legislation here every year but it never gets through |
| 01:58.35 | SteveTotaro | I am in MD |
| 01:58.39 | tzanger | it'd be fairly straightforward to tie in to any kind of CMS; one of the things on the board was integration with their CRM so that when they picked up the phone it jumped to a journal, in addition to the usual click-to-dial |
| 01:58.56 | [TK]D-Fender | jblack, pong |
| 01:59.07 | SteveTotaro | some counties they are illegal, like prince george's county where my brother is a cop |
| 01:59.12 | jblack | [TK]D-Fender: Today just got a little better for you. :) |
| 01:59.16 | fiXXXerMet | SteveTotaro: I'm in MD too ;) |
| 02:00.54 | fiXXXerMet | How does the asterisk dialplan support calling external programs? I'd like to write a script where a user can enter their order number and it'll call a script that checks our database for the status of their order, and reads it back to them. |
| 02:01.05 | SteveTotaro | oh yeah what part fiXXerMet? |
| 02:01.20 | jblack | fiXXXerMet: There's two ways. You probably want ATI. |
| 02:01.27 | jblack | Pardon, you probably want AGI. |
| 02:02.47 | SteveTotaro | tzanger, i have done that with wildfire and an activex jabber control, the CRM was web based all on thin clients, it worked really really well |
| 02:03.17 | SteveTotaro | i love when fairly far out ideas "just work" |
| 02:03.32 | jblack | You set up an extension in your dialplan that pushes the call into agi, which calls a script/program you define. A pile of useful environment variables are set, and * and your script talk with stdin/stdout. There's also eagi (where you can access the channel directly, such as for recording) and deadagi, which you use if you want to work after hangup |
| 02:03.59 | tzanger | yep, it worked out well for them. recording box sat between the PRI and the PBX |
| 02:04.05 | fiXXXerMet | AGI, eh? I saw that mentioned in the book. I'll read that chapter now. |
| 02:04.15 | jblack | You'll love it. =) |
| 02:04.17 | SteveTotaro | as a bridge? |
| 02:04.21 | fiXXXerMet | Sweet! :) |
| 02:05.05 | SteveTotaro | i prefer port mirroring so recording is more parallel than in the middle |
| 02:05.33 | tzanger | SteveTotaro: it did no routing or PBX functions. IAX between PBX and it, and TE407 I think to PSTN |
| 02:05.56 | tzanger | load balancing two of them was the idea both for call volume and failover |
| 02:06.12 | tzanger | writing was to a NAS over gige |
| 02:07.06 | SteveTotaro | writing for my deployment was over a second gig nic directly to a NAS as well |
| 02:08.00 | SteveTotaro | entire call was recorded and then trimmed based on time in queue and answer event so all the music on hold garbage was stripped |
| 02:08.23 | fiXXXerMet | ~book |
| 02:08.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 02:08.47 | tzanger | nice we didn't get ther,e they actually wanted NO modification whatsoever for sbarnes-oxley or however you spell it |
| 02:09.22 | SteveTotaro | it was funny listening to people bitch when call volumes were high |
| 02:09.42 | SteveTotaro | and kids screaming in the background, conversations with others |
| 02:09.56 | tzanger | heh |
| 02:10.20 | SteveTotaro | how did you associate the call with the customer? |
| 02:11.47 | tzanger | src/dest and callerid for incoming |
| 02:11.47 | *** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
| 02:12.01 | tzanger | then there was a backend task that grouped calls from known common numbers |
| 02:12.06 | tzanger | and faxes, etc. |
| 02:12.08 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 02:12.27 | tzanger | actually faxes were also done with tone detection but could have been done on src or dest too |
| 02:13.04 | *** join/#asterisk kamanashisroy (n=root@202.56.7.142) |
| 02:13.10 | SteveTotaro | we tracked sipcallid as well in case there was no record of their ANI and had to be pulled up by customer number or whatever |
| 02:14.03 | St1ckm4n | I'm looking to talk with someone that has experience installing/configuring asterisk witout using trixbox or other prepackaged version |
| 02:14.24 | SteveTotaro | st1ckm4n, you are in the right place then |
| 02:15.28 | tzanger | we never did ANI becaues they did not use tollfree. I haven't seen any guaranteed ANI info with regular DIDs |
| 02:15.42 | St1ckm4n | we had a company configure a switch for us over a year ago and have been having all kinds of issues with it since it is running in a medium sized call center. One thing I have heard is that Trixbox/Asterisk@home can cause issues in a heavier use enviroment |
| 02:16.06 | SteveTotaro | i was dealing with a full t3 inbound call center |
| 02:16.21 | St1ckm4n | something about it using an agi script everytime a call is initiated or ended |
| 02:16.57 | SteveTotaro | every call had to be recorded for QA, and legal. |
| 02:17.07 | tzanger | SteveTotaro: holy shit, how did you handle write i/o? |
| 02:17.21 | SteveTotaro | i did not use asterisk at all |
| 02:17.32 | tzanger | SteveTotaro: trixbox can barely handle a small office, fuck anything larger |
| 02:17.38 | tzanger | it gets out of hand in a big hurry |
| 02:18.06 | tzanger | and then of course consultants like me get called in to fix it, and it's hopeless. I gave up trying to support trixbox quite a while ago |
| 02:18.08 | SteveTotaro | i had seven hp boxes with quad port sangoma t1 cards |
| 02:18.36 | SteveTotaro | stripped down asterisk, they just took the tdm and converted to ulaw sip |
| 02:18.41 | tzanger | I'm wearing a sangoma fleece thing right now |
| 02:18.53 | tzanger | yeah that's what I'm doing with 1200 channel Tadex chassis right now |
| 02:18.57 | SteveTotaro | really, how did you get that? |
| 02:19.18 | tzanger | SteveTotaro: sangoma occassionally throws a job offer my way :-) I live about 90 minutes from their head office |
| 02:20.14 | St1ckm4n | is asteriskNow a prepackaged version like Trixbox or is it much better? |
| 02:20.20 | SteveTotaro | ah a canuck |
| 02:20.25 | SteveTotaro | lol |
| 02:20.34 | SteveTotaro | do you know stephen bosch? |
| 02:20.35 | [TK]D-Fender | St1ckm4n, same shit different smell |
| 02:21.15 | St1ckm4n | so it sounds like I'm better off staying away from all of those and just doing the long manual process of building it all from scratch |
| 02:21.19 | tzanger | SteveTotaro: no, not by name anyway |
| 02:21.33 | kyron | St1ckm4n, you sound like my echo |
| 02:21.38 | tzanger | St1ckm4n: it's not long nor tedious... download, ./configure, make menuselect, make && make install && make samples if you need |
| 02:21.46 | SteveTotaro | next time you talk to David tell him that "Steve Totaro wants a fleece and will wear it proudly" |
| 02:21.48 | fujin | Isn't asteriskNow an official asterisk project? |
| 02:21.52 | tzanger | SteveTotaro: I will |
| 02:21.54 | tzanger | for sure |
| 02:22.13 | kyron | IMHO, building * should be treated like building the kernel.. |
| 02:22.16 | Qwell | tzanger: get one for me too. I'll throw it in the pile of clothes I'll never wear |
| 02:22.18 | *** join/#asterisk windawg (n=win@209.129.161.251) |
| 02:22.20 | kyron | `Just do it` |
| 02:22.21 | tzanger | haha |
| 02:22.32 | SteveTotaro | asterisknow is an official digium project |
| 02:22.37 | [TK]D-Fender | 'Its easier to get forgiveness than permission' |
| 02:22.41 | SteveTotaro | but little incentive to develop it |
| 02:22.42 | kyron | everyone can make mistakes |
| 02:22.44 | St1ckm4n | I'm downloading the Centos 5.1 dvd right now, getting ready to wipe this shit and start clean |
| 02:22.58 | kyron | St1ckm4n, yerk...CentOS! |
| 02:22.58 | SteveTotaro | Hi Qwell |
| 02:23.14 | St1ckm4n | yeah, going off a consultants recommendations |
| 02:23.19 | kyron | St1ckm4n, at the very least, build your kernel |
| 02:23.20 | SteveTotaro | fedora core 7 |
| 02:23.29 | kyron | St1ckm4n, he wants your money ;) |
| 02:23.51 | tzanger | St1ckm4n: well look at it this way. go install trixbox/freepbx/whatever, and get stuck in their hell, or spend maybe 5 more hours at it and be far less at the mercy of us consultants. |
| 02:24.05 | kyron | mweheheh |
| 02:24.10 | tzanger | and actually no, I don't consult for general asterisk these days |
| 02:24.17 | SteveTotaro | ~trixbox |
| 02:24.22 | jbot | it has been said that trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 02:24.22 | tzanger | I design drivers and do specialized, high density installs |
| 02:24.26 | kyron | well, seems people like to be at the mercy of RH and RPMs anyways ;) |
| 02:24.31 | file | tzanger in person = scary |
| 02:24.41 | tzanger | file: <shock> |
| 02:24.43 | tzanger | that hurts |
| 02:24.53 | file | scary in a good way ... |
| 02:24.54 | file | maybe ... |
| 02:25.03 | SteveTotaro | what is scary about him? |
| 02:25.07 | SteveTotaro | big, intense? |
| 02:25.16 | kyron | fluffy? |
| 02:25.25 | tzanger | file: everyone I know who has met you has said something similar... the way you can carry on a conversation, watch a presentation and actively code at the same time... yikes |
| 02:25.50 | Qwell | a conversation? ha |
| 02:25.55 | file | I don't converse. |
| 02:25.57 | *** join/#asterisk shido6 (n=shido6@74-130-120-227.dhcp.insightbb.com) |
| 02:26.08 | SteveTotaro | converse was so 80s |
| 02:26.08 | tzanger | file: you spoke with kiwi at i360 |
| 02:26.15 | Qwell | I've seen him carry on a verbal conversation, while coding, and talking to 8 people on IRC |
| 02:26.16 | St1ckm4n | is Centos going to give me more headaches? |
| 02:26.40 | file | o.o |
| 02:26.44 | tzanger | St1ckm4n: I've heard a lot of people here, including one of hte authors of TFOT, who really like Centos |
| 02:27.21 | JunK-Y | yeah, file is multi-thread. |
| 02:27.25 | SteveTotaro | redhat enterprise = centos |
| 02:27.38 | tzanger | I'm a slacker myself, but I like debian for servers like this |
| 02:28.06 | tzanger | RPM = teh pain |
| 02:28.10 | tzanger | I stay away at all costs |
| 02:29.16 | JunK-Y | tzanger: will u be at it360 this year? |
| 02:29.21 | tzanger | JunK-Y: no, not this time |
| 02:29.30 | JunK-Y | too bad. |
| 02:29.31 | tzanger | JunK-Y: I was only there because I was invited, but I got next ot nothing out of it |
| 02:29.33 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:29.42 | St1ckm4n | I've also decided to bite the bullet and just use asterisk 1.4 and deal with the dialplan alternative of agentcallbacklogin |
| 02:30.26 | Corydon76-dig | Good m4n |
| 02:30.30 | file | tzanger: come now, you got to meet people |
| 02:30.37 | tzanger | yes that was the good part |
| 02:30.42 | tzanger | hanging out with the asterisk people |
| 02:30.44 | *** join/#asterisk troublehelix (n=win@209.129.161.251) |
| 02:31.54 | SteveTotaro | that was the cool thing about going to astricon for my first time in october |
| 02:32.30 | Corydon76-dig | I might have to go to Astricon sometime |
| 02:32.32 | SteveTotaro | got to put faces with names and sit down with and get to know some really cool people |
| 02:32.52 | tzanger | yeah that's the fun part |
| 02:33.01 | tzanger | ending up on an impromptu expert panel was interesting |
| 02:33.05 | St1ckm4n | We've been creating a seperate queue in the past for every new customer and used queue stats for reporting but this screws with the call distribution, If I can lump our customers into 2 or 3 queues aside from building our own reports is there anything out there that will give us similar reports based off the did/clid? |
| 02:33.19 | SteveTotaro | i was a little nervous speaking in the call center panel though it was not bad once it got underway, larger turnout than i expected |
| 02:33.33 | tzanger | St1ckm4n: I'ms ure you can glean that kind of info out of a db-stored CDR |
| 02:33.46 | jblack | St1ckm4n: did you consider setting account codes for cdr? |
| 02:34.17 | St1ckm4n | jblack:I haven't I'll need to look into that |
| 02:34.40 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 02:35.45 | SteveTotaro | anyone do worldcommunitygrid? |
| 02:36.28 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 02:37.21 | jblack | Is that as similiar to .*@home as it sounds? |
| 02:37.41 | SteveTotaro | yes |
| 02:37.41 | jblack | (which is called boinc these days) |
| 02:38.03 | SteveTotaro | boinc is the client but can be connected to many different projects |
| 02:38.17 | SteveTotaro | i do worldcommunitygrid |
| 02:38.24 | SteveTotaro | via the boinc client |
| 02:39.26 | kyron | St1ckm4n, ever considered debian or ubuntu...might want to ;) |
| 02:39.35 | SteveTotaro | Points Generated (Rank) 4,479,338 (#702) |
| 02:40.20 | St1ckm4n | kyron: I haven't I was leaning towards CentOS just because of what a recent consultant had mentioned |
| 02:40.26 | SteveTotaro | http://document.write(window.location.host)www.worldcommunitygrid.org/team/viewTeamInfo.do?teamId=GCHH2PTJQ1 to join the Asterisk team if you are into that kind of thing |
| 02:40.44 | SteveTotaro | http://www.worldcommunitygrid.org/team/viewTeamInfo.do?teamId=GCHH2PTJQ1 |
| 02:41.15 | kyron | St1ckm4n, bbiab |
| 02:46.16 | tzanger | wtf is boinc |
| 02:46.57 | SteveTotaro | a client for grid computing |
| 02:47.28 | SteveTotaro | i like to boinc ;) |
| 02:48.03 | SteveTotaro | ranked 719 out of ~350,000 |
| 02:48.39 | St1ckm4n | jblack: if I specify an accountcode in my dial plan will asterisk queue statistics allow me to do reporting for those individual codes? |
| 02:48.45 | SteveTotaro | so i can boinc good |
| 02:49.06 | SteveTotaro | cdr will |
| 02:50.52 | St1ckm4n | good to know, that was my biggest worry of consolidating the queues was losing the individual reporting |
| 02:52.24 | *** join/#asterisk PepOSX (n=angeldav@190.78.221.19) |
| 02:54.14 | kyron | St1ckm4n, RH/CentOS Vs Gentoo is, imho, analogous to Trixbox Vs * |
| 02:54.52 | [TK]D-Fender | kyron, No, not an appropriate comparison. |
| 02:55.19 | St1ckm4n | that comparison scares me |
| 02:55.43 | kyron | [TK]D-Fender, oh, you don't use RH's config interface and edit all files manually? |
| 02:55.54 | [TK]D-Fender | kyron, RH/CentOS & Gentoo are general purpose distros. Trixbox is a full distr that comes with *, FreePBX and a pile of other junk and tells you "how it is" and if you don't like it, tough. |
| 02:56.13 | [TK]D-Fender | kyron, and yes, I DO edit all my confs manually. |
| 02:56.24 | [TK]D-Fender | kyron, I'm a former Slacker :) |
| 02:56.35 | kyron | gone sour |
| 02:56.55 | [TK]D-Fender | kyron, No.... Always been sour, just add salt & Tequila! |
| 02:57.01 | kyron | ahhh |
| 02:57.06 | kyron | yes |
| 02:58.01 | kyron | point is, RH has a bad habbit of wrapping things in a way that the original project's code has been so convoluted most of em won't support you or even want to answer your questions. |
| 02:59.46 | [TK]D-Fender | kyron, Not sure I can attest to that for RH, but that fits Trixbox to a tee :) |
| 03:00.01 | JunK-Y | any feedback on http://www.amazon.ca/Programming-POSIX-Threads-David-Butenhof/dp/0201633922/ref=pd_sim_b?ie=UTF8&qid=1201057096&sr=8-4 ? |
| 03:00.20 | kyron | [TK]D-Fender, I therefore stand by my analogy |
| 03:00.36 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 03:00.55 | kyron | JunK-Y, a good reference |
| 03:02.02 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 03:03.07 | JunK-Y | good is not enough, i need excellent! |
| 03:03.08 | JunK-Y | :P |
| 03:04.20 | kyron | that all depends on what you make of it.. |
| 03:04.52 | kyron | Microsoft is extremely well documented...doesn't mean it's good ;) |
| 03:07.06 | *** part/#asterisk fiXXXerMet (n=meep@cmu-24-35-53-185.mivlmd.cablespeed.com) |
| 03:07.54 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 03:09.27 | jblack | The world would be a better place if someone would device a nice cordiality linter. |
| 03:09.41 | jblack | s/device/devise |
| 03:10.19 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-123-159.lns10.syd6.internode.on.net) |
| 03:11.40 | kyron | St1ckm4n, anyways, to each there own. All RH derivative, imho, prove to be a real PITA as soon as you don't "walk the path". As soon as you need something a little off of their packaging scheme, you get to see what RH (Real Hell) RPMs can be ;) |
| 03:12.15 | skopii | ^^^that is the damn truth lol |
| 03:12.36 | kyron | St1ckm4n, but hey, my experience with RH-ish distros is archaïc, so don't take my word for it, I've converted too long ago to be considered an unbiased preacher ;) |
| 03:13.18 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 03:13.19 | St1ckm4n | I appreciate your input, this is something I should research more since it is going to the base of our new phone switch |
| 03:13.33 | kyron | skopii, yeah well, not to self and all, this might actually be true to all "purely" binary distros out there ...and SRPMs is a bad patch |
| 03:14.38 | SteveTotaro | anyone see this yet? http://voipcalls4free.blogspot.com/ |
| 03:15.13 | drmessano | Hmm |
| 03:15.29 | SteveTotaro | i will have to give mark fedor a call and see if i can get some gizmos |
| 03:15.30 | drmessano | So I cant install from source on a RH derivitive? |
| 03:16.16 | UnixDog | ok other then just spouting off . need to know honest issues you guys have with grandstream phones ..... the gxp series |
| 03:16.33 | UnixDog | no just saying they suck I need to know issues |
| 03:16.51 | UnixDog | as I have a meeting on friday with 3 of the devs |
| 03:16.54 | drmessano | Didnt the GXP2000 have issues with echo and crappy audio? |
| 03:17.10 | UnixDog | not in the latest firmware |
| 03:17.19 | Qwell | drmessano: among others |
| 03:17.26 | SteveTotaro | and bridged calls |
| 03:17.45 | drmessano | I heard it was better now, but I remember a long string of problems |
| 03:20.12 | jblack | I could always use another did. Anyone have a spare grandcentral invite |
| 03:20.29 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 03:20.42 | javar | hi |
| 03:20.59 | drmessano | jblack, gimme your info |
| 03:21.03 | jblack | drmessano: Someone was having trouble with no dialtone, earlier. |
| 03:21.07 | jblack | drmessano: You rock. :) |
| 03:21.09 | javar | somebody know how can do NOT load extensions.ael? |
| 03:21.30 | jblack | James Blackwell, grandcentral@linuxguru.net |
| 03:21.48 | [TK]D-Fender | UnixDog, Other complaints : feel cheap, poor use of LCD, flakey firmware (stability issues), etc |
| 03:21.55 | jblack | javar: You could probably disable the ael module in modules.conf |
| 03:22.03 | [TK]D-Fender | javar, "noload => pbx_ael.so" |
| 03:22.11 | jblack | noload => pbx_ael.so |
| 03:22.12 | javar | jblack: ah sure.. |
| 03:22.14 | jblack | bleh. got beat |
| 03:22.34 | UnixDog | ok br fone |
| 03:22.49 | jblack | I also disable cdr_odbc, func-odbc, res_jabber, cdr_pgsql, res_odbc, res_smdi, cdr_custom, chan_zap and chan_skinny |
| 03:23.08 | jblack | If you're looking for modules to turn down |
| 03:23.20 | [TK]D-Fender | jblack, I think you need chan_zap.so for ztdummy to be around for timing.... |
| 03:23.40 | jblack | Believe it or not, meetme works with chan_zap noloaded. :) |
| 03:23.58 | [TK]D-Fender | jblack, Really? Wonder if it links to it directly or something.... |
| 03:24.08 | jameswf-home | as a matter of curiosity why doesnt chan_zap build by default. |
| 03:24.18 | Qwell | jameswf-home: because you didn't install zaptel |
| 03:24.19 | [TK]D-Fender | jameswf-home, depends on Zaptel |
| 03:24.20 | *** part/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133) |
| 03:24.22 | JunK-Y | jblack: i know, but you must have a zap kernel mod loaded. |
| 03:24.53 | SteveTotaro | modprobe ztdummy |
| 03:24.56 | jameswf-home | if you dont have zaptel it obviously wont load but is there a reason it cant sit as a little bloat |
| 03:25.07 | Qwell | jameswf-home: it can't compile without zaptel |
| 03:25.20 | UnixDog | well the newer units are heaver |
| 03:25.23 | JunK-Y | SteveTotaro: have you tried like 150 meetme with ztdummy? |
| 03:25.33 | Corydon76-dig | noloading func_odbc? Sacrilege! You heretic! |
| 03:25.40 | UnixDog | and the new firmware is better and the lcd I will mention |
| 03:25.48 | SteveTotaro | no, i always put some sort of card in my boxen |
| 03:25.51 | javar | jblack: i did that you told me, but still load extensions.ael |
| 03:25.53 | JunK-Y | preload => odbc_cory! |
| 03:26.10 | SteveTotaro | even if is a an emty tdm400p |
| 03:26.17 | UnixDog | bbiab tech call |
| 03:26.19 | Corydon76-dig | Heresy in thought, heresy in deed, and heresy in, uh, FOUR counts |
| 03:26.26 | jblack | javar: Hmm. it works for me. |
| 03:26.39 | javar | :( |
| 03:26.39 | Qwell | SteveTotaro: does that work for timing? |
| 03:26.43 | SteveTotaro | i wonder how fc 7 would do with ztdummy |
| 03:27.07 | Qwell | SteveTotaro: could probably use HRES timers |
| 03:27.18 | SteveTotaro | yes, you have to modify the zaptel source code first but it works with no modules |
| 03:27.19 | jblack | Perhaps you forgot to save modules.conf before you restarted *. |
| 03:27.28 | javar | jblack: i put noload => pbx_ael.so at the end of the modules.conf file, then at the cli> module reload |
| 03:27.44 | JunK-Y | javar: you need to restart, not reload |
| 03:27.52 | JunK-Y | to reparse the modules.conf |
| 03:28.02 | javar | JunK-Y: ah ok, let me try |
| 03:28.10 | jblack | Um, put it under the [modules] section, not in the [global] section |
| 03:28.44 | javar | JunK-Y: it works now |
| 03:28.51 | JunK-Y | javar: aleluia! |
| 03:28.56 | javar | :) |
| 03:28.58 | jblack | And I'd do a full restart of *, |
| 03:29.03 | jblack | Damnit. Got beat again |
| 03:29.23 | javar | jblack: thanks, i put in the [modules] |
| 03:30.14 | jblack | res_speech.so Generic Speech Recognition API 0 |
| 03:30.19 | jblack | huh. |
| 03:30.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:30.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:31.36 | SteveTotaro | Quell, to do timing with an empty tdm400p you can do "modprobe wctdm timingonly=1" |
| 03:32.43 | JunK-Y | SteveTotaro: what's the timingonly used for exactly, since an empty card already gives timing, no? |
| 03:32.45 | SteveTotaro | or you can modify wctdm.c and set static int timingony = 1; |
| 03:33.28 | SteveTotaro | it wont give timing unless you do the above |
| 03:33.47 | UnixDog | ok the newer version of the phones are heaver |
| 03:34.00 | UnixDog | and the new firmware fixes alot |
| 03:34.07 | UnixDog | so what else |
| 03:34.28 | UnixDog | what wron with the lcd |
| 03:34.45 | UnixDog | I see no issues with it it bright and clear |
| 03:34.54 | kyron | drmessano, it's just harder work... out of the box, RH doesn't come with gcc installed...as an example. bad one but still one ;) |
| 03:34.59 | SteveTotaro | easily scratched |
| 03:35.34 | [TK]D-Fender | kyron, My CentOS installs always did.... |
| 03:35.46 | drmessano | Hmmm |
| 03:35.48 | UnixDog | well if you choose to rm the clear plastic cover on any screen you have that issue |
| 03:36.10 | drmessano | a base, base install |
| 03:36.30 | drmessano | No groups of packages |
| 03:36.37 | drmessano | Will not have GCC |
| 03:36.39 | jblack | drmessano: Thnks |
| 03:36.52 | drmessano | but a normal install, nothing unchecked, will |
| 03:36.56 | UnixDog | anything else with the grandstreams ? |
| 03:37.10 | drmessano | Thats not a full install, but the few packages checked by default |
| 03:37.11 | SteveTotaro | they should change their name |
| 03:37.13 | UnixDog | as this meating is to redirect them to fixing design |
| 03:37.25 | drmessano | So no, CentOS does come with GCC |
| 03:37.25 | UnixDog | why |
| 03:37.25 | drmessano | jblack: no probs |
| 03:37.38 | SteveTotaro | it has a bad rep since the BT100 |
| 03:38.11 | drmessano | kyron, if you do nothing but install RPMs and rely on your ability to find RPMs for every single app out there, youre limiting yourself |
| 03:38.18 | SteveTotaro | it is like valuejet after they crashed over the everglades |
| 03:38.28 | SteveTotaro | they changed their name to airtran |
| 03:38.48 | drmessano | If you use it for the commong packages, you can actually not be a friggin n00b and install something from source now and then and be fine with RH |
| 03:39.00 | drmessano | s/commong/common |
| 03:39.24 | drmessano | It doesnt kill you to wget a tarball now and then :) |
| 03:39.35 | SteveTotaro | that is all i do |
| 03:39.45 | SteveTotaro | beside a few yum installs |
| 03:40.13 | drmessano | Yep.. saying that you need RPMs to do anything on a RH based distro is .. .. "n00bish" |
| 03:40.33 | drmessano | "ZONO, NO RPM... SAD FACE.. I NO INSTALL" |
| 03:40.43 | drmessano | Eh, yeah |
| 03:40.59 | UnixDog | well you have to let the bt100/101/102 |
| 03:40.59 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 03:41.26 | SteveTotaro | i have always liked grandstream ATAs though |
| 03:41.41 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 03:41.43 | drmessano | [TK]D-Fender uses them to hold doors open |
| 03:42.34 | SteveTotaro | i was an authorized grandstream dealer for a while during the bt101,102 days |
| 03:42.47 | SteveTotaro | sold in my webstore, lots of RMAs |
| 03:43.16 | SteveTotaro | no profit and a big headache, so i am a little biased perhaps |
| 03:43.17 | drmessano | I know one local company thats using GXP2000s with this little toaster PBX device for "VoIP Systems" and they seem to like em |
| 03:43.30 | drmessano | But then again.... |
| 03:44.04 | drmessano | He's a little umm.. |
| 03:44.15 | SteveTotaro | well to me, the only thing the user sees and physically interacts with is the phone |
| 03:44.24 | drmessano | "I got my A+ in 79, im cool right?" |
| 03:44.44 | SteveTotaro | so i only do polycom or cisco if the customer insists on it |
| 03:45.22 | JunK-Y | i was born in 79 :) |
| 03:45.28 | drmessano | lol |
| 03:45.30 | SteveTotaro | "users" will decide if the "phone system" is any good based on the phone sitting on their desk |
| 03:45.57 | drmessano | To a point |
| 03:46.15 | UnixDog | well I ope to get them back on track and we should see alot of changes in the next few months |
| 03:46.18 | drmessano | You could put a $6 wal mart phone on their desk.. if they have dialtone, they dont care |
| 03:46.33 | UnixDog | so back to making the list and writiing the first email; |
| 03:46.43 | SteveTotaro | not my customers |
| 03:47.11 | *** join/#asterisk Q3Man (n=Steve@nv-65-173-88-122.sta.embarqhsd.net) |
| 03:47.30 | SteveTotaro | i don't do customers that want to deal with $6 phones |
| 03:47.48 | drmessano | What I meant was |
| 03:48.11 | UnixDog | well the fact is this it the need for low cost phones in the market for startup companies. and low cost devices for endusers who dont have alot of money to get in the door |
| 03:48.12 | SteveTotaro | they are used to lucent/definity setups |
| 03:48.28 | UnixDog | I agree the bt100/101/102 where crap |
| 03:48.32 | drmessano | People want to make phone calls.. I dont care if you put a $400 phone on their desk, or a $6 phone on their desk.. their calls sound bad, or they cant dial, and youre dead |
| 03:48.46 | UnixDog | the bt 200 seems to work alot better |
| 03:49.05 | SteveTotaro | poly 301s at a minimum |
| 03:49.13 | UnixDog | and the gxp earlier units had issues but they have been working to make the units heavier |
| 03:49.21 | drmessano | Screw pretty lamps, nice displays, and dozens of buttons.. they want dialtone and it BETTER work all the time.. the rest is fluff |
| 03:49.41 | SteveTotaro | yeah, the cheap customers want that |
| 03:49.46 | UnixDog | I would not put polys in a callcenter |
| 03:50.00 | UnixDog | they would get to expensive to replace all the time |
| 03:50.11 | UnixDog | so you have to have lower cost units |
| 03:50.31 | SteveTotaro | no, i use quintum tenor AX units and regular plantronics sets |
| 03:51.03 | CrashSys | Hmmm... Anyone familiar with AsteriskGuru QueueStats? All my calls in a queue are showing up as SIP and not SIP/101 |
| 03:51.28 | SteveTotaro | i would not put any good electronics in a call center |
| 03:51.41 | [TK]D-Fender | Polycom IP 301 = dead end. No point in buying. |
| 03:51.47 | SteveTotaro | they get a thin client and an analog headset |
| 03:51.59 | SteveTotaro | poly 301 is a good phone and cheap |
| 03:52.11 | UnixDog | 301 has no speaker phon |
| 03:52.12 | UnixDog | ee |
| 03:52.25 | UnixDog | <PROTECTED> |
| 03:52.25 | [TK]D-Fender | SteveTotaro, The IP 320/330 is better and cheaper. The 301 is on the way out the door |
| 03:52.37 | SteveTotaro | better how? |
| 03:52.47 | UnixDog | 301 is good for a lobby phone |
| 03:52.51 | UnixDog | send it here |
| 03:52.53 | SteveTotaro | oh yeah, you are right |
| 03:52.59 | UnixDog | I will take the 301 |
| 03:53.27 | [TK]D-Fender | SteveTotaro, lets see, native PoE, pixel based display, lit line keys, MicroBrowser support, full duplex speakerphone, 2.5mm stand headset jack, and $CHEAPER$ |
| 03:53.32 | SteveTotaro | i have 501s in my office |
| 03:53.33 | [TK]D-Fender | SteveTotaro, That about cover it for you? |
| 03:53.36 | UnixDog | 300/301 make great lobby and front door phones |
| 03:53.55 | [TK]D-Fender | Anybody buying a 301 at anything near retail is a chump. |
| 03:54.04 | SteveTotaro | i guess depending on where you buy them would dictate price |
| 03:54.06 | UnixDog | lol |
| 03:54.14 | SteveTotaro | anyone buying anything at retail is a chump |
| 03:54.17 | UnixDog | 301 are worth about 20 bucks |
| 03:54.22 | [TK]D-Fender | SteveTotaro, Maybe in Bizarro World you will find validation :) |
| 03:54.23 | SteveTotaro | sign up for var pricing |
| 03:54.41 | SteveTotaro | i said i agreed with you, scroll up |
| 03:54.51 | [TK]D-Fender | SteveTotaro, the thing is that the IP 320/330 retails quite a bit less than the 301. The 301 is DEAD. |
| 03:55.08 | SteveTotaro | 301 works just fine |
| 03:55.16 | UnixDog | 301/300/500 eol |
| 03:55.21 | [TK]D-Fender | VAR would just mean that it cost less that for other people, but relative cost remains the same. |
| 03:55.46 | [TK]D-Fender | SteveTotaro, Oh yes, the 301 DOES work just fine, just that if it came to a new purchase you'd have to be on crack to chose it :) |
| 03:55.54 | CrashSys | Anyone know why asterisk 1.4.17 would go from queue logging with SIP/101-0938409 to just SIP/101 ? |
| 03:55.56 | [TK]D-Fender | SteveTotaro, I do have 2 myself... |
| 03:56.18 | SteveTotaro | i have 501s |
| 03:56.28 | SteveTotaro | i have installed many 301s |
| 03:56.34 | [TK]D-Fender | SteveTotaro, Homestly I'd have preferred an IP 301 to the Aastra 57i CT I had at the office for a while. |
| 03:56.49 | [TK]D-Fender | SteveTotaro, IP 501 is a very hard phone to suggest these days as well... |
| 03:57.00 | SteveTotaro | it works |
| 03:57.03 | SteveTotaro | great |
| 03:57.03 | [TK]D-Fender | SteveTotaro, the IP 320/330 were indeed category killers. |
| 03:57.20 | SteveTotaro | thats fine, i already agreed with you, stop beating a dead dog |
| 03:57.28 | [TK]D-Fender | SteveTotaro, Again I agree it is a good phone, but the price point vs feature comparison and typical needs makes it a very hard fit. |
| 03:57.39 | SteveTotaro | fine |
| 03:57.45 | [TK]D-Fender | SteveTotaro, and I agree with you as well, and we're moving up to horses :) |
| 03:58.07 | CrashSys | fender for president |
| 03:58.09 | SteveTotaro | nah, horses are too big to beat |
| 03:58.19 | [TK]D-Fender | SteveTotaro, for reference I have a 301 & 501 in my bedroom here :) |
| 03:58.34 | SteveTotaro | thats wonderful |
| 03:58.46 | [TK]D-Fender | SteveTotaro, I like my "filly"-mignon tender! Get to it boy! |
| 03:59.11 | [TK]D-Fender | SteveTotaro, Had an IP 430, sold that off. I run IP 600/601/430's at the office primarily. |
| 03:59.33 | SteveTotaro | i am really impressed |
| 03:59.35 | [TK]D-Fender | SteveTotaro, IP 6XX is where the real comfort sets in... |
| 03:59.43 | CrashSys | I'm still waiting for a customer to splurge for the side-car's... i'd like to configure those once... just once... to say I did... |
| 03:59.56 | [TK]D-Fender | CrashSys, Funny thing... NOTHING to configure :) |
| 04:00.11 | CrashSys | the SLA crap :) |
| 04:00.14 | [TK]D-Fender | CrashSys, your buddies just roll off onto it naturally. No magic. |
| 04:00.38 | [TK]D-Fender | CrashSys, and * SLA = 2-bit craptastic hack. |
| 04:00.47 | [TK]D-Fender | russellb, we still love you though! |
| 04:00.51 | CrashSys | which is why I only want to configure it once :) |
| 04:01.10 | [TK]D-Fender | CrashSys, You can do that on something like a 501 and up just to say you did it. |
| 04:01.25 | [TK]D-Fender | CrashSys, But I live even my enemies better than that :) |
| 04:01.36 | CrashSys | Yeah, but I want to mess with the sidecars :( |
| 04:01.47 | [TK]D-Fender | CrashSys, well then by all means! |
| 04:01.55 | CrashSys | see if I cant make a pong game with sipsak or something on them |
| 04:02.20 | [TK]D-Fender | CrashSys, But thats one thing Aastra KILLS Polycom on : Presence & speed-dial/soft-keys |
| 04:02.23 | jblack | Oh, speaking of things that some people like... |
| 04:02.35 | jblack | [TK]D-Fender: You don't have a whole lot of love for the grandstreams, right? |
| 04:02.36 | [TK]D-Fender | CrashSys, make them sweep like KITT lights! |
| 04:02.47 | [TK]D-Fender | jblack, other side of the coin :) |
| 04:02.49 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 04:02.50 | CrashSys | Now we're talking |
| 04:03.07 | CrashSys | sample the swoosh sound effect and p ut it into the phone |
| 04:03.09 | jblack | Is it personal, or are they really _that_ bad? |
| 04:03.10 | [TK]D-Fender | CrashSys, or play "Speed-dial-whack-a-mole"! |
| 04:03.12 | CrashSys | have it be the reboot sequence |
| 04:03.52 | CrashSys | too bad queuemetrics is dual-licensed :( |
| 04:04.15 | SteveTotaro | i have an unlimited license for QM |
| 04:04.21 | [TK]D-Fender | jblack, they feel like flimsy plastic shit. They make poor use of their screen, their firmwares have worse odds than a magic 8-ball. I don't buy "crap". I don't care if some people feel "satisfied" with them, its worth a bit more for something you KNOW will be good. |
| 04:04.39 | CrashSys | Steve: Shoot me a single-server copy then |
| 04:05.13 | jblack | Ahhh... SteveTotaro tried to change my mind about them earlier. |
| 04:05.26 | SteveTotaro | no, i said their ATAs were not too bad |
| 04:05.30 | SteveTotaro | not the phones |
| 04:05.50 | [TK]D-Fender | jblack, yeah, there are a number of people who love to pimp them, but frankly cost is the only thing going their way. And for that : |
| 04:05.54 | [TK]D-Fender | ~ygwypf |
| 04:05.55 | jbot | extra, extra, read all about it, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 04:05.57 | CrashSys | I have actually heard that grandstream ATA's dont suck any worse then the linksys/sipura ones... |
| 04:06.22 | CrashSys | Multitech/etc makes better tho |
| 04:06.26 | [TK]D-Fender | I just have more faith in the Linksys stuff. |
| 04:06.36 | [TK]D-Fender | CrashSys, Mediatrix ATAs are SICK... |
| 04:06.50 | Corydon76-dig | ATAs just suck in general |
| 04:06.52 | CrashSys | Never messed with a Mediatrix... dont do much ATA stuff really... |
| 04:07.00 | drmessano | I like the PAP2s, except for the lack of power for dual G729 |
| 04:07.00 | [TK]D-Fender | CrashSys, Transparent WAN proxying, etc... cryptic bitches to configure, but VERY powerful |
| 04:07.19 | [TK]D-Fender | drmessano, another reason I don't recommend them. |
| 04:07.23 | SteveTotaro | for any port density i do an atran 600 with FXS ports or a quintum tenor ax |
| 04:07.25 | Corydon76-dig | The only reason to use them is if you have some nontraditional thing that needs a phone interface, like a paging system |
| 04:07.38 | [TK]D-Fender | SteveTotaro, at what starting point? |
| 04:07.39 | jblack | Sorry for that. I misunderstood you when you said "20:53 < SteveTotaro> grandstream atas are actually not bad" |
| 04:07.42 | Corydon76-dig | or an audio board |
| 04:07.43 | CrashSys | I use Zhone Z-Plex 10's! |
| 04:08.01 | [TK]D-Fender | SteveTotaro, (in terms of port density) |
| 04:08.06 | Corydon76-dig | CrashSys: that's a channel bank, not an ATA |
| 04:08.18 | SteveTotaro | it acts as an ATA |
| 04:08.25 | CrashSys | Yeah, but it's in the same class as an Adtran 600 |
| 04:08.26 | SteveTotaro | analog terminal adaptor |
| 04:08.30 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:08.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:08.41 | Corydon76-dig | Hah, not even close to an Adtran 600 |
| 04:08.57 | Corydon76-dig | The Zhone Zplex's are about the worst channel bank you can buy |
| 04:09.14 | SteveTotaro | why not, it is by definition and analog terminal adapter, no? |
| 04:09.14 | CrashSys | Work fine for what I use 'em for, which is just deliver 24-pairs of dialtone... |
| 04:09.26 | CrashSys | I dont use any of the IAD crap |
| 04:09.33 | Corydon76-dig | The Zplex does not support callerid |
| 04:09.34 | CrashSys | Caller-ID/faxing works |
| 04:10.08 | Corydon76-dig | or is that only on their FXO ports? |
| 04:10.18 | CrashSys | Prolly FXO... I only use the FXS |
| 04:10.24 | CrashSys | the FXO I have heard is a gold-plated turd |
| 04:10.28 | SteveTotaro | i only use FXS too |
| 04:10.33 | Corydon76-dig | I've only dealt with the 16/8 variety |
| 04:11.04 | CrashSys | The FXS works ok as just a dumb channel bank... never tried to use the IAD features... |
| 04:11.06 | Corydon76-dig | Every other channel bank with FXO ports correctly interprets callerid |
| 04:11.26 | Corydon76-dig | At least, the ones I've used |
| 04:11.46 | CrashSys | I buy 'em on e-bay for $160 buy-it-now or $120 if you wait... |
| 04:11.50 | SteveTotaro | wouldn't know, i don't see a use for an FXO channel bank |
| 04:12.38 | CrashSys | I've used a CAC Adit 600 before with FXO, but everything worked on that... flawlessly... |
| 04:12.57 | SteveTotaro | yes, ebay has some great deals channel banks |
| 04:13.08 | SteveTotaro | populated with FXS cards |
| 04:13.48 | Corydon76-dig | plus if you've ever tried reconfiguring the Zplex's, you have to type very slowly... |
| 04:13.54 | SteveTotaro | cheapest way to go provided you have an extra T1 port |
| 04:14.10 | SteveTotaro | for 24 ports of analog dialtone |
| 04:14.44 | CrashSys | Corydon: I'm not advocating them for use as anything other then a cheap way to get 24-ports of dialtone... |
| 04:15.14 | SteveTotaro | rock solid dialtone |
| 04:15.31 | Q3Man | can asterisk handle incoming faxes? |
| 04:15.34 | CrashSys | Corydon: I just configure the IP port then tftp over the config and it programs itself... and yes, to configure it without already having a file is 1100 or so line of typing :) |
| 04:16.42 | Corydon76-dig | CrashSys: and if you send the config file, you have to make sure that you pause after every line, or it locks solid |
| 04:17.17 | Corydon76-dig | The one I had could only be configured via serial |
| 04:17.23 | CrashSys | network restore makes it download a txt file witht he commands, and it cycles through them itself 1 at a time... like what's up on voip-info.org |
| 04:17.44 | *** join/#asterisk j__1 (n=j__1@c-98-195-149-98.hsd1.tx.comcast.net) |
| 04:17.59 | x86 | SteveTotaro: the Adit 600 can do MGCP/IP trunking to the phone system too, so you don't even need a T1/E1 port ;) |
| 04:18.19 | SteveTotaro | but the tenor AX is awesome to ship to a to a remote call center or office |
| 04:18.21 | CrashSys | Yeah, the Adit 600 was nice... things just tended to work :) |
| 04:18.35 | SteveTotaro | i thought MGCP was not compete in Asterisk |
| 04:18.47 | SteveTotaro | *complete* |
| 04:18.51 | SteveTotaro | getting tired |
| 04:18.54 | x86 | works decent enough |
| 04:19.09 | CrashSys | works for point-A to point-B stuff |
| 04:19.10 | x86 | unfortunately, they don't make SIP or IAX cards for it.... |
| 04:19.45 | SteveTotaro | how much can an adit 600 be had for? |
| 04:19.55 | CrashSys | Does 1.6 have any threading/trunking issues with IAX like 1.4 had? |
| 04:20.17 | CrashSys | Steve: They can run from $500 to $1500 depending on options... look on ebay |
| 04:20.39 | SteveTotaro | the MGCP works though? |
| 04:21.01 | SteveTotaro | i have a few Adtran 600s but I would like to play with the MGCP |
| 04:21.24 | x86 | you need the MGCP card for it |
| 04:21.57 | SteveTotaro | anyone connect asterisk with a NEC IPK with a MGCP card? |
| 04:22.03 | x86 | CMG-02 |
| 04:22.38 | SteveTotaro | well if the price gets up too high, i would just opt for the quintum tenor ax |
| 04:22.43 | SteveTotaro | and do sip |
| 04:24.43 | x86 | Adit 600 with (24) FXS ports set me back $1400 new |
| 04:24.56 | x86 | not sure how much the CMG-02 is |
| 04:24.58 | jameswf-home | looked on ebay some guy is selling our channel banks for 2-500 over list... people really should do research and frankly get it new |
| 04:26.30 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 04:26.33 | SteveTotaro | http://cgi.ebay.com/QUINTUM-TENOR-AX-SERIES-AXM-2400-GATEWAY_W0QQitemZ260203975434QQihZ016QQcategoryZ61839QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 04:27.12 | x86 | yeah i dont buy any production VoIP hardware from eBay |
| 04:27.25 | x86 | I only used a trusted reseller |
| 04:28.08 | SteveTotaro | $2.5k new |
| 04:28.38 | jblack | I liked the way voipsupply.com treaded me when I bought the spa-8k |
| 04:28.47 | [TK]D-Fender | AudioCodes MP-124 can be had for less than the ADIT and offers redundancy. |
| 04:28.47 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 04:29.07 | CrashSys | I miss atacomm |
| 04:29.14 | [TK]D-Fender | jblack, was it an ATR tread, or just 4-season? |
| 04:29.23 | jblack | all weather. |
| 04:29.37 | [TK]D-Fender | jblack, master of none it is :) |
| 04:29.57 | jameswf-home | anyone going to the IT expo I heard on the ride home florida is under water :) |
| 04:30.20 | CrashSys | Which expo? |
| 04:30.28 | jameswf-home | Internet telephony |
| 04:30.45 | CrashSys | Going to the florida linux expo in february... |
| 04:31.56 | mocker | [TK]D-Fender: A server I'm looking at is getting a weird error, any chance you can take a look at http://pastebin.ca/869411 ? |
| 04:32.05 | jameswf-home | The \y need to do this stuff in vegas... My boss might go for that |
| 04:32.17 | mocker | It's a failed transfer |
| 04:33.09 | *** join/#asterisk nDuff (n=ccd@user-387ocuv.cable.mindspring.com) |
| 04:33.26 | mocker | What I don't get is the 'Failed to authenticate user "user2"' after user2 answered. |
| 04:33.36 | [TK]D-Fender | mocker, I don't see anything I can help with, sorry |
| 04:33.49 | [TK]D-Fender | mocker, Some sort of redirect perhaps? |
| 04:34.10 | CrashSys | ahhh, star2star is sponsoring... kristian will probably be there... red-bulls in hand... |
| 04:34.13 | mocker | [TK]D-Fender: The phones are Polycoms, I haven't seen that type of error before. |
| 04:34.41 | nDuff | How long of a number can be passed as a DID w/ pri_cpe signalling? I'm looking at a situation where I might need to get caller ID info sandwitched into the DID field on incoming calls (so, <cid>#<did>), but neither I or the guy on the other end knows how many digits we have to work with. |
| 04:35.55 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:36.22 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:40.37 | jameswf-home | typicaly DID's are 3-5 digits |
| 04:41.56 | drmessano | Shorter the better |
| 04:44.14 | jameswf-home | Thats what she said :) |
| 04:44.18 | drmessano | ROFL |
| 04:44.33 | jameswf-home | !did |
| 04:44.38 | jameswf-home | ~did |
| 04:44.39 | jbot | hmm... did is Direct Inward Dialing, or just a phone number |
| 04:44.55 | jameswf-home | bot dyslexia |
| 04:45.04 | drmessano | jbot: don't PM me without asking me pls |
| 04:45.09 | drmessano | lol |
| 04:47.12 | *** join/#asterisk fedya (n=fedya@c-76-26-182-13.hsd1.fl.comcast.net) |
| 04:47.29 | fedya | anyone going to the itexpo tomorrow? |
| 04:49.30 | jameswf-home | my boss will be there |
| 04:52.56 | fedya | yea my boss changed his mind |
| 04:53.00 | fedya | i have a $2k pass |
| 04:53.05 | fedya | so now i'm not going... |
| 04:53.13 | fedya | was gonna see if i could salvage it for some $ |
| 04:55.23 | fedya | boss not taking you? |
| 04:56.12 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 05:01.36 | jameswf-home | Nah I am kept in the dundeon woth the other ninja coding monkeys |
| 05:01.45 | jameswf-home | *dungeon |
| 05:03.31 | drmessano | lol |
| 05:06.30 | jameswf-home | this never gets old http://www.youtube.com/watch?v=8zEQhhaJsU4 |
| 05:07.14 | jameswf-home | I dont like windows but I love uncle steve |
| 05:09.30 | [TK]D-Fender | Ok, time to visit the land of Oz.. later all... |
| 05:11.24 | fedya | jameswf-home, what do you do? |
| 05:12.55 | mocker | Anyone know what the most recent SIP firmware is? |
| 05:13.08 | jameswf-home | I am a slave to the man.... I like long walks on the beach...... |
| 05:13.09 | mocker | Er, for Polycom |
| 05:13.23 | [TK]D-Fender | mocker, SIP 3.0.0 |
| 05:13.23 | mocker | PolycomSoundPointIP-SPIP_430-UA/1.6.7.0133 |
| 05:13.28 | mocker | That seems old to me. :) |
| 05:13.34 | [TK]D-Fender | mocker, VERY |
| 05:13.49 | mocker | [TK]D-Fender: I think that's my problem. ;) |
| 05:14.17 | [TK]D-Fender | mocker, *a* problem perhaps... |
| 05:14.21 | [TK]D-Fender | *yawn* |
| 05:14.25 | [TK]D-Fender | Ok, I'm off... |
| 05:14.58 | drmessano | I know what TKs problem is now |
| 05:15.17 | drmessano | Hes afraid to be loved.. which is why he quits so fast.. and doesnt give anyone a chance to say goodnight |
| 05:15.23 | drmessano | He just needs a REALLY big hug |
| 05:16.06 | drmessano | jbot: hug [TK]D-Fender |
| 05:16.07 | jbot | ACTION hugs [TK]D-Fender tightly until [TK]D-Fender turns slightly blue |
| 05:16.47 | jameswf-home | jbot: eat poop |
| 05:16.47 | jbot | ACTION slurps up all the poop available |
| 05:17.19 | jameswf-home | jbot is like that one kid who will do anything for a dollar |
| 05:17.46 | jameswf-home | jbot kill the kitten |
| 05:17.47 | jbot | ACTION shoots a magneto-ionized quark gun at the kitten |
| 05:18.01 | jblack | jbot: screw me |
| 05:18.02 | jbot | ACTION asks jblack to bend over |
| 05:18.07 | jameswf-home | lmao |
| 05:18.18 | jameswf-home | jbot screw you |
| 05:18.19 | jbot | ACTION asks you to bend over |
| 05:18.22 | drmessano | jbot: get trunked |
| 05:18.23 | jbot | ACTION fetches trunked |
| 05:18.26 | drmessano | LOL |
| 05:18.47 | jameswf-home | jbot fix my car |
| 05:18.47 | jbot | ACTION opens the hood of my car , looks inside and says "that'll be 500 dollars." |
| 05:18.53 | drmessano | ROFL |
| 05:19.02 | drmessano | jbot you're fired |
| 05:19.14 | jameswf-home | jbot want to fight |
| 05:19.16 | drmessano | Just like a salesperson |
| 05:19.40 | jameswf-home | jbot commit suicide |
| 05:19.40 | jbot | Hehe my survival instinct prevents me from killing myself |
| 05:19.48 | jameswf-home | lol |
| 05:19.51 | jblack | jbot: Hurt jameswf-home |
| 05:19.52 | jbot | ACTION takes a large hammer and plays whack-the-mole on jameswf-home's hands, laughing all the while |
| 05:20.04 | jblack | Ok. Someone needs to teach jbot the three laws. |
| 05:20.12 | jameswf-home | jbot whois jameswf-home |
| 05:20.25 | jblack | A bot that will protect itself, then blithely whack people with hammers, is a dangerous bot. |
| 05:20.45 | jameswf-home | jbot 1984 |
| 05:20.46 | jbot | hmm... 1984 is a novel about 2007, written by George Orwell in 1948. |
| 05:20.57 | drmessano | jbot jblack |
| 05:21.05 | drmessano | Crap |
| 05:21.17 | jameswf-home | ~ron paul |
| 05:21.18 | jbot | i guess ron paul is my choice for president in 2008 |
| 05:21.52 | drmessano | jbot: jblack has a linksys callcenter in his 200 sq ft apartment |
| 05:21.58 | drmessano | ~jblack |
| 05:22.02 | drmessano | bastard |
| 05:23.05 | jameswf-home | jbot you bastard |
| 05:23.14 | jameswf-home | jbot bastard |
| 05:23.33 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:23.39 | jameswf-home | jbot get me a beer |
| 05:23.39 | jbot | ACTION fetches me a beer |
| 05:24.01 | jameswf-home | jbot get drmessano a hooker and a donkey |
| 05:24.02 | jbot | ACTION fetches drmessano a hooker and a donkey |
| 05:26.42 | drmessano | lol |
| 05:27.00 | drmessano | Did you catch what I said about you in the "other channel"? |
| 05:27.07 | drmessano | Yesterday I think it was |
| 05:27.20 | jameswf-home | te drunk chair thing |
| 05:27.24 | drmessano | ROFL.. yah |
| 05:27.30 | drmessano | Figured you would like that |
| 05:33.12 | *** join/#asterisk nighty^ (n=nighty@p8172-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 05:39.27 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 05:40.16 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 06:09.53 | *** join/#asterisk thethimble (n=Advait@adsl-75-49-105-70.dsl.irvnca.sbcglobal.net) |
| 06:10.22 | thethimble | is it possible to spoff outgoing caller id using asterisk? |
| 06:11.03 | thethimble | spoof* |
| 06:14.14 | thethimble | or can you even make outgoing calls with it? |
| 06:14.43 | drmessano | LOL |
| 06:14.53 | thethimble | yeah, thanks |
| 06:14.59 | drmessano | Are you trying to prank people? |
| 06:15.07 | thethimble | well, kinda |
| 06:15.13 | drmessano | Hmm |
| 06:15.14 | thethimble | i'm trying to setup an allcall system for our school |
| 06:15.30 | drmessano | ~book |
| 06:15.31 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 06:15.32 | thethimble | i want the outgoing caller id to be the school's number |
| 06:15.33 | drmessano | Read up |
| 06:15.39 | drmessano | yes you can dial OUT with Asterisk |
| 06:15.45 | drmessano | Can you send mail with Sendmail? |
| 06:16.19 | thethimble | huh? |
| 06:16.31 | drmessano | Read up on Asterisk |
| 06:16.36 | drmessano | yes you can dial out with asterisk |
| 06:16.41 | drmessano | You can actually dial in too |
| 06:16.44 | drmessano | and talk |
| 06:16.49 | thethimble | wow, cool |
| 06:16.53 | thethimble | thanks for the help |
| 06:16.57 | drmessano | no probs |
| 06:18.39 | jblack | thethimble: sorry. The callerid is locked |
| 06:18.51 | drmessano | jblack: In 1.4, you can make calls with Asterisk, right? |
| 06:19.00 | drmessano | I saw that patch commited, wasnt sure |
| 06:19.23 | jblack | No, all that patch gave me was smoke signals. :(((( |
| 06:19.40 | drmessano | SAD FACE :( |
| 06:20.16 | drmessano | Good thing I can recieve calls.. now I just need friends |
| 06:20.35 | jblack | Wait.. were you actually asking me/ |
| 06:20.52 | jblack | If not, I thought we were goofing. |
| 06:20.58 | jblack | I can make calls. |
| 06:21.20 | drmessano | lol |
| 06:21.26 | drmessano | We are man, we are |
| 06:21.28 | drmessano | lol |
| 06:21.44 | jblack | And receive them, and forward them, and.... put them on hold and send them to voice mail and.... record them and inject sound into them and... |
| 06:22.01 | drmessano | Heres one for ya, jblack |
| 06:22.17 | jblack | The only one * hasn't solved for me is my turret's syndrome problem. |
| 06:22.22 | drmessano | Ive messed with system() a little |
| 06:22.29 | jblack | Ok |
| 06:22.32 | *** join/#asterisk nvala1 (n=me@c-67-188-103-195.hsd1.ca.comcast.net) |
| 06:22.57 | drmessano | I can update my XMPP status by hitting an HTTP address |
| 06:23.06 | nvala1 | hi. Anybody used sip_ping.pl from the VoIP Hacks book? |
| 06:23.15 | drmessano | If I wget the address it pulls an XML file |
| 06:23.15 | jblack | "Hi, dominos, I'd like a large pizza YOU UGLY SOB. Can you please EAT POO send it to 333..." |
| 06:23.38 | jblack | ok |
| 06:23.49 | drmessano | Can do something ugly like wget http://blah && rm blah.xml? |
| 06:24.15 | jblack | Yeah, you can system("uglylittlehack")... But you'll go blind if you do it too much. |
| 06:24.23 | drmessano | lol |
| 06:24.51 | drmessano | I man need to man wget.. I think I can get it to hit the address to check it, without downloading |
| 06:24.56 | drmessano | That MAY be enough |
| 06:25.02 | drmessano | err |
| 06:25.06 | jblack | Until I had the phone systme ran here, my closet server did a system("mpg123 rinringring.mp3") |
| 06:25.09 | drmessano | s/man/may |
| 06:25.23 | jblack | Just to tell me to go downstairs and answer the phone. |
| 06:25.29 | drmessano | heh |
| 06:25.41 | jblack | drmessano: Personally, I'd just do an agi(xmpccupdate.sh).... |
| 06:26.01 | jblack | or .pl, and use perl's http:: module to do the update. |
| 06:26.10 | drmessano | ok |
| 06:26.44 | nvala1 | I have a * server, and defined multiple extensions within my LAN. I can call between those extensions no problem .. |
| 06:26.56 | drmessano | That may be easier actually.. I can feed the JID as a command line var to the sh for each line |
| 06:27.02 | nvala1 | but, if I have a peer outside my LAN, then I don't get any voice .. |
| 06:27.11 | drmessano | ~sipnat |
| 06:27.12 | 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 |
| 06:27.13 | jblack | though system() will do what you need to do today, you might someday want to do something slightly more... like change statuses by dialing an extension, or feeding back xmpc status back through *, and you'll appreciate the env variables and stdin/stdout. |
| 06:27.25 | nvala1 | thnx dr |
| 06:27.42 | jblack | and a agi() can be just as braindead as system() if you want it to be |
| 06:28.19 | drmessano | I need to read up a bit.. Id like to be able to dial an ext for each user, and DTMF a status |
| 06:28.30 | drmessano | Like 1 thru 9 |
| 06:28.45 | jblack | can you elaborate? |
| 06:28.47 | drmessano | Eventually anyway |
| 06:28.48 | drmessano | Well |
| 06:28.52 | jblack | What is the ext pattern? |
| 06:29.27 | *** join/#asterisk ZX81_ (n=ZX81@202.20.97.211) |
| 06:29.42 | jblack | exten => 1, Set(gtalkuser="Bob") |
| 06:29.47 | jblack | exten => 2, Set(gtalkuser="Frank") |
| 06:29.49 | drmessano | What I initially planned was ugly.. like adding ext 1101 thru 1109 to send the wget blah 1 drmessano or wget blah 7 drmessano |
| 06:29.49 | jblack | And so on? |
| 06:30.11 | jblack | hmm. |
| 06:30.26 | *** join/#asterisk xachen (n=justin@pdpc/supporter/student/xachen) |
| 06:30.34 | jblack | I'd make a context that fed certain extensions (say 1101 - 1109) straight to an agi, pick the dialed extension out.... |
| 06:30.38 | drmessano | I can send a string like wget blah.php?jid=drmessano;status=1 |
| 06:30.50 | jblack | And then have the script do what it wants based upon that. |
| 06:31.12 | jblack | This is for people to manually indicate to gtalk whether or not they're "here" ? |
| 06:31.35 | *** join/#asterisk Jenna (i=JJ@gateway/tor/x-202a7f926b7e3b9f) |
| 06:31.55 | Jenna | Hey all, |
| 06:32.00 | drmessano | We're talking out using XMPP for an in/out presence board..and it would be cool to be able to manipulate that ove the phone |
| 06:32.09 | jblack | So, yes. |
| 06:32.12 | drmessano | yeah |
| 06:32.16 | jblack | Here's what I'd do. |
| 06:32.39 | jblack | I'd add somewhere into the IVR "Press X say you are in. Press Y to say you are out" |
| 06:32.40 | Jenna | Im on design my own ATA things. Does anyone know of the any available design spec documents ? |
| 06:33.11 | drmessano | jenna: Check the RFCs for SIP |
| 06:33.19 | jblack | Have X and Y bounce them to anther context. Then, based on whether it's X or Y, feed their voicemailbox to the script. |
| 06:33.39 | jblack | thats assuming users have individual sip accounts... ? |
| 06:33.48 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 06:33.49 | drmessano | Yeah, they would |
| 06:33.57 | jblack | Yeah, do that... OR.... :) |
| 06:34.09 | jblack | You can set variables in sip.conf, under sip contexts. |
| 06:34.10 | Jenna | drmessano, hmm. IC. thanx |
| 06:34.36 | jblack | Would you like to automate the inbox/outbox? |
| 06:35.05 | jblack | (you can, if users are logged in with sip only when they're there) |
| 06:35.37 | drmessano | Mainly what I want to do is change the extended away statues to some canned responses.. for example |
| 06:36.25 | drmessano | I am at my desk, so I can control most of my status.. The XMPP server is tied to Asterisk, so it will handle "on phone".. If I walk away, it will go to "Away" for me automaticall.. if I dont set an away already.. |
| 06:36.35 | drmessano | But If say I go to an appt with a client |
| 06:37.00 | drmessano | I can set "With client" on my status, by having it canned in a script, over the phone |
| 06:37.23 | drmessano | and back at the office, ed can look at his XMPP client or the in/out board on the web and know not to call me |
| 06:38.33 | jblack | Ok. |
| 06:38.47 | jblack | Then you want people to dial an agi, rather than dial your phone. |
| 06:38.53 | drmessano | Yes |
| 06:39.13 | jblack | The agi would check with xmpc to see your status, and based on that either sned them straight to your voice mail, or Dial() you. |
| 06:39.28 | jblack | That's a perfectly valid usage of agi. |
| 06:39.31 | drmessano | No, im not worried about thta part. |
| 06:39.39 | drmessano | That would be nice, no doubt |
| 06:39.41 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 06:39.50 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 06:39.51 | drmessano | Moreso the in office status my coworkers would see |
| 06:40.00 | drmessano | Being able to update it form the field |
| 06:40.17 | jblack | so... |
| 06:40.48 | drmessano | I go out with at a client location.. I call into the status line and dial 1047 "Thank you, you are now set to "with a client"" |
| 06:41.04 | drmessano | Everyones XMPP status for me changes to "With a Client" |
| 06:41.06 | jblack | Tell me a story about what happens on February 13, 2008. |
| 06:41.14 | drmessano | I leave the client |
| 06:41.23 | drmessano | I dial back in with 1048 |
| 06:41.32 | drmessano | "You are now "away"" |
| 06:41.39 | jblack | Ahh. You want to manipulate your xmpp status by dialing in/ |
| 06:41.41 | drmessano | So everyone sees me as away, but not busy |
| 06:41.44 | drmessano | yes |
| 06:41.56 | drmessano | I could do it pretty ugly |
| 06:42.10 | drmessano | Like assign 5 ext numbers x 60 users |
| 06:42.13 | drmessano | lol |
| 06:42.18 | jblack | yeah, you could do an ugly hack. |
| 06:42.26 | drmessano | Give 104 1041 thru 1045 to dial |
| 06:42.28 | jblack | changestatus,1,VMPasswd() |
| 06:42.28 | drmessano | or something |
| 06:42.36 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 06:42.47 | jblack | changestatus,n,GetDTMF() |
| 06:43.12 | jblack | and based on the dtmf, set any of a handful of statuses with a list of systems().. |
| 06:43.45 | drmessano | Yeah |
| 06:43.56 | jblack | If I implemented it, I'd prefer dropping straight into agi, getting vmpasswd for the exten, give a prompt for statuses you can set, and then modifying xmpp status with a perl module. |
| 06:44.33 | drmessano | At the very least you need 1 ext per user to dial |
| 06:44.44 | drmessano | Because you need the XMPP status coded |
| 06:44.45 | drmessano | No |
| 06:44.50 | drmessano | Because you need the XMPP NAME coded |
| 06:44.57 | jblack | A pile of systems() would work, but it start ugly and quickly become disastrous as you decide "Bob would like this too" "and so would frank" |
| 06:46.13 | drmessano | you could do say exten ==> 1041,1,System(wget blah.php?JID=bob;status=1) |
| 06:46.20 | jblack | Yeah, I get that. |
| 06:46.21 | drmessano | exten ==> 1042,1,System(wget blah.php?JID=bob;status=2) |
| 06:46.25 | drmessano | But ugly |
| 06:46.30 | drmessano | thats hundreds of lines |
| 06:46.47 | jblack | I'd rather keep the list in ldap and grab it while in perl or php. |
| 06:46.58 | jblack | Some simple 1:1 mapping somewhere. |
| 06:47.17 | jblack | extension bleh = jid blah |
| 06:47.23 | drmessano | yep |
| 06:47.52 | jblack | That would let you keep a real simple dialplan, and do the lifting where the lifting is easy. |
| 06:48.38 | drmessano | Something where you call an IVR |
| 06:48.41 | drmessano | Prompts for ext |
| 06:48.45 | drmessano | It checks the mapping |
| 06:48.51 | drmessano | Prompts for status |
| 06:49.05 | drmessano | Then executes the string for your JID and status # |
| 06:49.08 | jblack | Yeah. All the dialplan knows is that extension 1234 is for updating status. |
| 06:49.22 | jblack | The status part of the ivr could be in the agi. |
| 06:49.24 | drmessano | That would be very cool |
| 06:50.24 | jblack | Yeah, that wouldn't be too hard to write. |
| 06:50.27 | drmessano | You could set up mappings in a conf file to keep it simple |
| 06:50.38 | drmessano | 109=joe |
| 06:50.43 | jblack | I could even be persuaded to write it with suitable encouragement. |
| 06:51.21 | jblack | sure, a conf file, or a ldap search, or any given way. |
| 06:51.33 | drmessano | I need to play some more with the https strings on Openfire |
| 06:51.44 | drmessano | Ive done it, but I need to nail that part down |
| 06:51.55 | jblack | It could even be in a variable for each sip context in sip.conf |
| 06:52.55 | drmessano | Wait, thats different |
| 06:53.07 | jblack | Sorry... user defining data, not user defined data. |
| 06:53.34 | jblack | I.E. I like to keep "The user's name is James Blackwell" and "James Blackwell's phone number is 206-203-5855" in the same place. |
| 06:54.04 | jblack | I don't like it spread about in 20 different files. |
| 06:54.21 | drmessano | I got ya.. I was just being silly |
| 06:55.02 | jblack | for all the talking we've done tonight... a lot of your arrows point at getting an ldap server going. |
| 06:55.16 | jblack | donadie: Please go away. |
| 06:55.42 | jblack | donadie: iow, please go "away". |
| 06:56.05 | drmessano | I like being able to set "away" |
| 06:56.17 | jblack | see? one per customer |
| 06:56.36 | drmessano | yep "away" for joe |
| 06:56.56 | drmessano | "away" for steve |
| 06:57.09 | drmessano | "away" for shirley |
| 06:57.13 | drmessano | and so on |
| 06:58.27 | drmessano | away away |
| 06:58.33 | drmessano | away away away, donadie |
| 06:58.35 | drmessano | lol |
| 06:58.59 | drmessano | Got a script that makes you sway, set that bugger to away |
| 06:59.09 | jblack | Gah! |
| 06:59.19 | jblack | Here, or there! Please stop making me dizzy! |
| 06:59.29 | drmessano | LOL |
| 06:59.31 | drmessano | Sorry |
| 06:59.37 | drmessano | I was hitting him in here too |
| 06:59.53 | drmessano | While I was talking to you |
| 06:59.57 | jblack | Is that guy still here? |
| 07:00.06 | drmessano | yeah |
| 07:00.12 | drmessano | That message was a few mins ago |
| 07:00.29 | jblack | Not bot guy. The.... "all-knowing" one |
| 07:00.35 | drmessano | oh |
| 07:00.38 | *** join/#asterisk bkruse_home (n=kruz@76.73.154.120) |
| 07:00.38 | *** mode/#asterisk [+o bkruse_home] by ChanServ |
| 07:00.50 | drmessano | Dont see him |
| 07:01.05 | jblack | Thanks for small favors. |
| 07:01.18 | drmessano | Youre still wrong |
| 07:01.42 | jblack | I can respect arrogance, but man... one should have all of one's ducks in a row before playing that role... imho |
| 07:01.45 | thethimble | hey, drmessano, i've been reading forever and i still haven't found out how to dial out |
| 07:01.47 | jblack | Me? |
| 07:02.11 | jblack | I'm wrong about things all the time. What did I screw up this time. :) |
| 07:02.16 | drmessano | jblack: didnt want you to feel more than inferior, so I told you that you were wrong |
| 07:02.22 | drmessano | Nothing specific ;) |
| 07:02.34 | jblack | Ok. jblack is teh sux0rs. |
| 07:02.36 | jblack | I couldn't agree more |
| 07:02.37 | bkruse_home | anyone here have experience with iaxclient? |
| 07:02.39 | drmessano | lol |
| 07:03.02 | jblack | thethimble: I can try and help you too, if you expland (expand? explain? dunno) a little bit more. |
| 07:03.09 | drmessano | thethimble, Dialing OUT is a basic function of Asterisk.. thats like telling me you cant figure out how to send mail with Postfix |
| 07:03.41 | thethimble | lol |
| 07:03.43 | jblack | bkruse_home: I haven't. Sorry. |
| 07:04.12 | thethimble | ok..... lemme explain...... i want to make a system that calls ~400 numbers sequentially and plays a message |
| 07:04.17 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 07:04.28 | jblack | Oh, really. |
| 07:04.39 | drmessano | Mass Pizza orderer, I see |
| 07:04.42 | drmessano | Damn spammer |
| 07:04.44 | drmessano | j/k |
| 07:04.45 | thethimble | lol |
| 07:04.46 | jblack | Gee. sorry. I'm brand new at this asterisk thing. |
| 07:04.53 | thethimble | Yeah....same |
| 07:04.55 | jblack | asterisk can dial numbers? |
| 07:04.58 | Olobola | heeeelllllloooooooo... all my contexts are on one line now!? How is this possible? |
| 07:05.03 | *** join/#asterisk J4zen (n=Jvan4zen@a82-95-153-17.adsl.xs4all.nl) |
| 07:05.04 | drmessano | Since 1.4.11 |
| 07:05.20 | jblack | Olobola: Please paste your extensions.conf to a pastebin (~pb to get a list) |
| 07:05.20 | drmessano | Olobola: You failed at life |
| 07:05.31 | drmessano | I mean |
| 07:05.33 | drmessano | Paste pls |
| 07:05.37 | thethimble | would it be possible to make a system like that? |
| 07:05.39 | jblack | drmessano: You're in a bad mood again? |
| 07:05.41 | drmessano | ~pb |
| 07:05.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 07:05.44 | drmessano | No |
| 07:06.11 | jblack | thethimble: No. Asterisk is very limited. It can only make 1 phone call at a time. |
| 07:06.19 | thethimble | .... |
| 07:06.21 | Olobola | it's not over yet: [general] static=yes writeprotect=no |
| 07:06.22 | drmessano | jblack: If I told you my sip.conf had the word poo in in, how did it get there? |
| 07:06.28 | drmessano | Just being MR Obvious |
| 07:06.36 | jblack | drmessano: Heh |
| 07:06.46 | thethimble | jblack: i know that.....i want it to call a list of people one at a time |
| 07:06.53 | Olobola | everything is on one line.. |
| 07:07.07 | drmessano | Olobola, how did you edit it? |
| 07:07.10 | drmessano | Windows? |
| 07:07.18 | Olobola | nano |
| 07:07.26 | jblack | thethimble: Oh, I'm glad. It's ok with you that the bug only allows 5 second calls? |
| 07:07.28 | drmessano | HMMMMMMMmmmmmmmMmmmmmmmMmmm |
| 07:07.31 | Olobola | this happened over night |
| 07:07.41 | drmessano | By itself? |
| 07:07.46 | jblack | Olobola: How's that paste coming along? |
| 07:07.56 | jblack | drmessano: Dude, why don't you go get a beer or something. |
| 07:08.05 | drmessano | lol |
| 07:08.10 | Olobola | I can't paste my passwords etc |
| 07:08.28 | jblack | on second though, why don't you grab some popcorn instead. |
| 07:08.34 | drmessano | haha |
| 07:08.43 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:08.47 | jblack | Olobola: I don't want your passwords. Feel free to replace passwords with XXXX |
| 07:09.15 | drmessano | thethimble, you can generate a slew of .call files to do that.. start from there |
| 07:09.24 | jblack | I'm vastly limited in that regard |
| 07:09.45 | drmessano | ESP is your friend |
| 07:09.54 | thethimble | drmessano, on a scale from 1 to 10, how complacated would that be |
| 07:09.57 | jblack | Wouldn't it be ERP in this case? |
| 07:10.06 | drmessano | thethimble, are you a Linux user? |
| 07:10.07 | jblack | thethimble: Oh, at least a 10. Possibly a 23. |
| 07:10.31 | drmessano | thethimble: do you have any linux experience? |
| 07:10.45 | thethimble | my linux skills are equivalent to those of a toddler |
| 07:10.57 | jblack | I'm sure he has plenty of experience with ./spamusenet |
| 07:11.01 | drmessano | On a scale from 1 to 10, its probably a 7306 for you |
| 07:11.05 | thethimble | lol |
| 07:11.07 | thethimble | wonderful |
| 07:11.41 | thethimble | i have managed to figure out sudo and apt-get, however |
| 07:11.48 | thethimble | but besides that, i'm clueless |
| 07:12.00 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:12.07 | jblack | why exactly do you want to call 300 people? |
| 07:12.16 | thethimble | senior prank |
| 07:12.17 | drmessano | you need to first undertstand Linux, then Asterisk, then call files.. that could take 11, maybe 12 years |
| 07:12.21 | jblack | Ahhh. |
| 07:12.22 | thethimble | i want to cancel school for a day |
| 07:12.33 | jblack | Ok. Helping you would be illegal. |
| 07:12.44 | thethimble | yeah....i guess |
| 07:12.57 | drmessano | Grab that book.. come back when youre 30 |
| 07:13.10 | jblack | Seriously. these days, it would be considered a terrorist act, and americans that help you would be at risk for a very dark jail cell for a very long tie. |
| 07:13.14 | jblack | s/tie/time. |
| 07:13.33 | drmessano | I love asterisk, but wont have my cornhole buttered over it |
| 07:14.08 | thethimble | alright, thanks for the help jblack, drmessano |
| 07:14.13 | thethimble | i'm going to go cry |
| 07:14.16 | Olobola | http://pastebin.com/d133a044e |
| 07:14.18 | jblack | Perhaps you can find a less legally dangerous prank (No, I'm not getting in trouble for contributing to the deliquency of a minor by suggesting you put up 300 pictures of your principal's head photoshopped onto a playboy bunny) |
| 07:14.46 | jblack | uhhh. geeze. |
| 07:14.49 | drmessano | ewwww |
| 07:15.05 | jblack | I think drmessano is right. I think you edited this with something that stripped line endings.. |
| 07:15.15 | drmessano | lol |
| 07:15.17 | jblack | i.e. a unix editor on a windows filesystem, or a windows editor on a unix filesystem. |
| 07:15.18 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:15.34 | drmessano | or you ignored how nano wraps text and tried to fix it |
| 07:15.39 | jblack | Or... you used vi for the first time, and hit J a WHOLE lot. |
| 07:15.47 | drmessano | He said nano |
| 07:15.55 | drmessano | I think he got confused |
| 07:15.57 | Olobola | putty and nano is all I use |
| 07:16.11 | drmessano | You didnt edit these in windows |
| 07:16.12 | drmessano | ? |
| 07:16.16 | jblack | Does nano have a join command, perhaps ctrl-j ? |
| 07:16.24 | jblack | Or a reformatting command? |
| 07:16.29 | *** join/#asterisk eharris (n=eharris@cpe-66-25-189-206.austin.res.rr.com) |
| 07:16.34 | Olobola | nano and putty, it has never been a problem. |
| 07:16.36 | *** join/#asterisk r0d3nt (i=nobody@pinky.ratman.org) |
| 07:16.53 | Olobola | maybe I did hit some funky command accidentally |
| 07:16.55 | drmessano | No, but if you scroll and put your line feeds in the wrong spots, the lines will break funky |
| 07:16.59 | drmessano | Especially LONG ones |
| 07:17.11 | jblack | AHA. |
| 07:17.39 | drmessano | Nano is bad about that |
| 07:17.39 | jblack | Make a few "paragraphs" separated by an extxra black line. Go to line 1. then hold down ctrl-j |
| 07:17.47 | jblack | It causes exactly what we see there. |
| 07:17.56 | drmessano | Hmm |
| 07:18.20 | jblack | Olobola: You'll have to fix this by putting the returns back in by hand. |
| 07:18.28 | jblack | In the future, avoid ctrl-J unless you mean it. |
| 07:18.35 | Olobola | thank you |
| 07:19.26 | jblack | if you don't mind extra advice..... also make sure you know where your penis is when you zip your pants up. Failing to watch out for that can have very painful results. |
| 07:19.53 | drmessano | Unless you're thimble sized |
| 07:20.15 | thethimble | :'( |
| 07:20.21 | jblack | What do you think the odds are that thimble is going to get himself expelled during the last 2 months of school? |
| 07:20.22 | drmessano | Then its like a speeding bus coming by with your standing on the curb.. close, but nah |
| 07:20.42 | drmessano | Hmm |
| 07:21.00 | thethimble | yeah, the consequences would be pretty bad |
| 07:21.11 | drmessano | What do you think the odds are he'll end up in Gitmo? |
| 07:21.14 | jblack | only if you dislike the idea of working in a gas station for the rest of your life. |
| 07:21.14 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 07:21.29 | jblack | drmessano: I'd give that one a... 1 in 6 chance at this point. |
| 07:21.40 | jblack | The schools are hyper-sensitive these days, after all |
| 07:21.40 | drmessano | Dont knock Gas station jobs... FREE ICEEs |
| 07:21.59 | drmessano | Blue Wackadoo ICEEs are worth it |
| 07:22.32 | jblack | YOu want to screw with the school in a way that will get you suspended at worst? |
| 07:22.48 | jblack | Wear a T-Shirt that says "Principal ____'s illegitimate child". |
| 07:23.31 | jblack | At least then you have a shot at the 1st amendment. |
| 07:23.37 | thethimble | That would be wholly self-demeaning |
| 07:23.55 | thethimble | our principle is an 80year old crypt keeper |
| 07:23.59 | thethimble | pal* |
| 07:24.00 | drmessano | Wear a t-shirt that says "I slept with Mrs ___" |
| 07:24.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:25.46 | thethimble | so, now that cancelling school and whoreish tshirts are out of the question |
| 07:25.49 | thethimble | any other ideas? |
| 07:26.16 | drmessano | Toilet Paper is a terrorist act, so thats out |
| 07:26.35 | thethimble | a bit cliche too |
| 07:26.39 | drmessano | yes |
| 07:27.45 | drmessano | Anything computer related will get you expelled for hacking |
| 07:28.26 | jblack | And anything that scares the staff is likely to get you in the sort of trouble you'll never get rid of. |
| 07:28.38 | jblack | You don't live in the same world that we lived in growing up. |
| 07:29.01 | thethimble | *sigh* |
| 07:29.03 | jblack | And trust me. Adults, especially adults that work at schools, are scaredy cats. Anything weird creeps them out. |
| 07:29.10 | jblack | I know dude. It sucks to be you. |
| 07:29.34 | drmessano | When I was in 7th grade, I put a thumbtack on a teachers chair |
| 07:29.40 | drmessano | I got a saturday for it |
| 07:29.47 | jblack | Those were the days. :) |
| 07:29.49 | drmessano | In 2008, I would charged with assault |
| 07:29.56 | drmessano | In 2008, I would BE charged with assault |
| 07:30.06 | drmessano | Expelled |
| 07:30.13 | drmessano | Have a record |
| 07:30.33 | jblack | these days, you can't even bring aspirin to school, for fear of suspension/expulsion for "drugs" |
| 07:30.45 | drmessano | So really the options are |
| 07:31.04 | drmessano | Do nothing.. or go rob a bank.. If youre going to go jail, make it worth it |
| 07:31.11 | drmessano | lol |
| 07:31.44 | drmessano | Its all or nothing anyway.. unless youre Britney |
| 07:32.22 | Olobola | what is wrong with this line? >>>exten => _8665551212,1,AGI(pprd.php) |
| 07:32.26 | thethimble | if it was possible to spoof caller ID, and batch call using someone else's phone line, what are the chances i would get caught? |
| 07:32.40 | jblack | About 100%. |
| 07:33.19 | jblack | There's two types of callerid. One is the stuff that your phone prints out. The other, called ANI, is still there. |
| 07:33.48 | Olobola | I forgot to reload, works now. |
| 07:34.03 | thethimble | what if i ran it off the school's own phone line after hours? |
| 07:34.15 | jblack | If you do naughty things with callerid that annoy people enough to call the phone company, they _will_ have the proper ANI, and you will almost certainly be facing federal charges. |
| 07:34.28 | jblack | thethimble: Dude. Stop, and listen to me. |
| 07:35.00 | jblack | You're fucking with 20 years of jail. Seriously. No kidding. |
| 07:35.00 | thethimble | ok, sorry |
| 07:35.40 | thethimble | oh well......thanks for all the help anyway |
| 07:35.48 | drmessano | thethimble |
| 07:35.57 | drmessano | If you did it with Asterisk... you would be a "hacker" |
| 07:35.57 | jblack | If you thought 4 years of high school was a never-ending-dull-drag... well, you're really going to hate 20 years in a federal penetentiary. |
| 07:36.05 | drmessano | and that word.. gets you years |
| 07:36.13 | thethimble | hahaa |
| 07:36.16 | jblack | And thats' where you'll go, because it'll be a federal offense. |
| 07:36.17 | drmessano | People are stupid |
| 07:36.32 | thethimble | kk, advice taken |
| 07:36.34 | thethimble | i won't |
| 07:36.58 | jblack | i hope so. Because it's more fun to lay a girl called Bobbie, then get laid by a man called Bob. |
| 07:37.18 | drmessano | Bobbie is lot hotter than Bobby |
| 07:37.50 | thethimble | lol |
| 07:37.54 | jblack | Wanna really screw with people in a safe way? |
| 07:37.55 | drmessano | Bobbie give you her number.. Bobby burns his name into your arm with a paperclip |
| 07:38.06 | jblack | Realy piss off the teachers, in a way that you don't screw yourself over? |
| 07:38.37 | jblack | Wear a shirt that says "Me teechers taut me to reed & right good!" |
| 07:38.55 | jblack | And on the back, put a list of every teacher you had. |
| 07:39.01 | jblack | They'll positively hate you for it. |
| 07:39.01 | drmessano | LOL |
| 07:39.03 | drmessano | I like that |
| 07:39.37 | *** join/#asterisk Stefan1979 (n=stan@cpe.atm2-0-11038.0x50c42746.boanxx15.customer.tele.dk) |
| 07:40.37 | jblack | It's obnoxious, disrespectful, tactless, rude all wrapped up in one package. Teen angst in a T |
| 07:41.14 | jblack | If you do that, please put, on the back "O RLY? K THX BYE!!!!" |
| 07:42.23 | jblack | Or... You could get (from google) a picture of every teacher that's gone to jail over the last 20 years, and on the front say "Hey teach, what's your crime?" |
| 07:42.24 | *** join/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 07:42.57 | thethimble | haha, not bad |
| 07:43.12 | jblack | Bring an extra shirt. |
| 07:43.19 | jblack | ;) |
| 07:43.35 | `Sauron | Is the purpose of this to get in trouble, or just piss people off? |
| 07:43.55 | `Sauron | Because I think you're being a bit too un-subtle for the "piss people off" part |
| 07:43.56 | *** part/#asterisk Maxous (n=Maxous@76.97.3.24) |
| 07:44.05 | `Sauron | the goal being pissing people off without getting caught |
| 07:44.11 | `Sauron | presumably |
| 07:44.18 | jblack | `Sauron: The purpose here is to give thethimble an idea to replace the one that would land him in a federal-pounding-in-the-ass-institution |
| 07:44.33 | thethimble | "Lake Highlands High School: A friend of a student delivered marijuana-spiked muffins to the teachers lounge. "He brought bran muffins to suburban Lake Highlands High School on May 16, saying they were part of an Eagle Scout project. When school employees ate the muffins, they began complaining of nausea, lightheadedness and headaches, and were briefly hospitalized. "They were just thinking it would be fun to get these teac |
| 07:44.54 | thethimble | and they never went to jail |
| 07:44.56 | thethimble | how unfair |
| 07:45.10 | jblack | I'm amazed they didn't. |
| 07:47.08 | `Sauron | you don't want to be in PMITA prison? |
| 07:47.09 | `Sauron | Aww. |
| 07:47.40 | jblack | He did. That's why we're offering less drastic alternatives. |
| 07:47.48 | `Sauron | Why not let him go? |
| 07:47.58 | `Sauron | We could sell the documentary rights to WE or something. |
| 07:48.15 | jblack | Because then he'll learn all sorts of antisocial stuff in the pen like carjacking, hostage taking for fun and profit, etc.... |
| 07:48.27 | jblack | And I'll have to be that more careful about locking my house and car. |
| 07:48.51 | `Sauron | does he know where you life? |
| 07:48.53 | `Sauron | *livve |
| 07:49.03 | `Sauron | erk |
| 07:49.05 | `Sauron | s/vv/v |
| 07:49.07 | jblack | Metaphorically, sauron. Not literally. |
| 07:49.08 | `Sauron | :) |
| 07:49.16 | drmessano | I dont want Asterisk to be known as "The PBX that lands you in prison" |
| 07:49.20 | drmessano | People would talk |
| 07:49.21 | lowlevel | oooh federal pmita prison |
| 07:49.28 | lowlevel | not good |
| 07:49.29 | lowlevel | ;) |
| 07:50.24 | drmessano | app_pmita |
| 07:51.07 | jblack | Once upon a time, threatening teens with a life at mcdonald's was enough. These days, you have to threaten ass-raping. |
| 07:51.35 | drmessano | McDonalds is cool.. Im making $7 an hour now, AND I handle all their IT |
| 07:51.43 | jblack | Wow! |
| 07:51.51 | creativx | greasy IT |
| 07:51.52 | jblack | Do you get to play tennis often? |
| 07:51.58 | drmessano | lo |
| 07:52.02 | drmessano | lol |
| 07:52.18 | jblack | I haven't heard from him in at least a week. |
| 07:52.23 | jblack | He must be POTUS by now. |
| 07:52.26 | drmessano | Its cool man, one minute I am putting salt on fries, next minute, VLANs |
| 07:52.53 | lowlevel | mcdonalds isn't really a bad gig these days |
| 07:52.59 | jblack | Just make sure you stick the right chips in the frier. |
| 07:53.08 | jblack | lowlevel: Uh-huh. sure. |
| 07:53.09 | drmessano | lowlevel: if youre 17 |
| 07:53.09 | lowlevel | I applied there several times, but never did get an interview |
| 07:53.10 | lowlevel | ;) |
| 07:53.52 | jblack | Why make $30 an hour for sitting on your ass, when you can instead make $7.50 an hour for running around like a headless chicken, turning into a grease-smelling slave-of-the-man. |
| 07:53.54 | lowlevel | well, you can make about $26/hour managing a fast food place.... |
| 07:53.56 | jblack | Right? |
| 07:54.09 | drmessano | Toys R Us is a tough one.. all that screening out of sex offenders now |
| 07:54.10 | jblack | Sure. After you salt the fries for about 18 years. |
| 07:54.15 | lowlevel | I know university educated people working for less |
| 07:54.32 | jblack | And I know high school dropouts making more. |
| 07:54.33 | `Sauron | Actually, should I lose my job for some odd reason (Like not giving a crap about it anymore, hehn), I'll become a night stock person at the local grocery |
| 07:54.36 | `Sauron | $10/hour starting |
| 07:54.37 | `Sauron | HAH |
| 07:54.49 | `Sauron | And I'd be out of there as quick as... mofo. |
| 07:54.53 | lowlevel | sauron: factor in cost of living, and you know what.. that might not even be the end of the world |
| 07:54.53 | lowlevel | ;) |
| 07:55.07 | drmessano | Besides, most of the managers at McDonalds have never kissed a girl |
| 07:55.09 | jblack | But I know a lot more high school dropouts that make next to nothing than I do collegiates.. And I know a lot more collegiates making $50/hr+ than I do high school dropouts |
| 07:55.11 | drmessano | :( |
| 07:55.23 | `Sauron | lowlevel: well, I know what I make now (a lot more than that) - but I figure if it comes down to paying the bills... shrug, why not. |
| 07:55.31 | lowlevel | I make 70k a year and I'm deeper in debt everyday here.. better off working at mcD's in some no-where town |
| 07:55.37 | `Sauron | I'll whore myself out to the highest paying to put food on the table |
| 07:56.07 | lowlevel | sauron: $5 and a can of beef-a-roni for ya |
| 07:56.44 | thethimble | alright gentlemen (and ladies, perhaps), thank you for all your suggestions |
| 07:56.52 | thethimble | i'm out |
| 07:56.59 | lowlevel | later |
| 07:56.59 | `Sauron | any ladies in here would refuse to id themselves as such |
| 07:57.06 | *** part/#asterisk thethimble (n=Advait@adsl-75-49-105-70.dsl.irvnca.sbcglobal.net) |
| 07:58.14 | lowlevel | a friend of mine used to work at mcdonalds.... |
| 07:58.29 | lowlevel | and.. his father got him a factory job later.. which he didn't want to leave McD's for. |
| 07:58.40 | lowlevel | (lots of action at the McD's apparently) |
| 07:58.42 | lowlevel | heh |
| 07:58.46 | drmessano | lol |
| 07:59.00 | lowlevel | 'uh, I got a job' hahaha |
| 07:59.10 | lowlevel | he was like 17 or something |
| 07:59.18 | lowlevel | I dunno, I found it funny :) |
| 07:59.21 | lowlevel | *shrug* |
| 07:59.38 | drmessano | lol |
| 08:06.39 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 08:07.21 | *** join/#asterisk WorgiL (n=usta@85.106.183.141) |
| 08:07.32 | jblack | I hate newbs. |
| 08:08.04 | jblack | Get a great defensive line setup, can take almost anything the ai throws at you... look over to your parter, and he's got like... 1 gun tower. |
| 08:08.31 | WorgiL | hi everyone, how can i set asterisk auido codecs to g729a ? thanks |
| 08:08.50 | jblack | WorgiL: You have alread purchased the g729 codecs for each * server? |
| 08:09.05 | WorgiL | i don't know jblack |
| 08:09.13 | jblack | Then you haven't. |
| 08:09.14 | WorgiL | new installed |
| 08:09.23 | mosty | WorgiL, go to www.digium.com and buy some g729 licenses |
| 08:09.29 | jblack | Dont use g729. It requires you to purchase licenses and get certificates. Use ulaw instead. |
| 08:09.50 | jblack | how old are the g729 patents? |
| 08:10.09 | WorgiL | jblack i don't know anything |
| 08:10.18 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 08:10.19 | WorgiL | just installed and testing on ubuntu server |
| 08:10.37 | jblack | WorgiL: Ok. Then forget g729. Just disable it. |
| 08:10.38 | WorgiL | but i saw i need g729a coecs |
| 08:10.46 | jblack | No, you do not. |
| 08:11.04 | mosty | WorgiL, you don't need g729, use gsm for a low bandwidth codec, or ulaw or alaw for high quality codec |
| 08:11.14 | *** join/#asterisk ixx (i=foobar@70.112.149.89) |
| 08:11.20 | WorgiL | mosty how can i set it |
| 08:11.36 | mosty | start by reading the book |
| 08:11.37 | mosty | ~book |
| 08:11.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 08:11.45 | WorgiL | because when i talking with anyone my noise losting |
| 08:12.32 | jblack | windows must be getting good for the disabled, because I swear 3/4 of the new people that join this channel can't read. |
| 08:12.33 | mosty | WorgiL, i have no idea what you're trying to say |
| 08:13.25 | WorgiL | mosty you know it using high bandwidth i mean this |
| 08:13.41 | *** part/#asterisk nvala1 (n=me@c-67-188-103-195.hsd1.ca.comcast.net) |
| 08:14.09 | mosty | WorgiL, look at the section on setting up SIP (and IAX if you use that) in the book, and then use gsm for your codec |
| 08:14.41 | WorgiL | mosty thanks |
| 08:14.55 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 08:17.22 | *** join/#asterisk rrrobert (n=rrrobert@58.65.160.140) |
| 08:17.39 | Olobola | are most of you on the east coast? |
| 08:17.46 | Olobola | or east |
| 08:18.20 | *** join/#asterisk steliosk (n=Stelios@85.72.92.83) |
| 08:18.43 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 08:19.39 | *** join/#asterisk reber (n=reber@193.253.213.73) |
| 08:20.00 | drmessano | east here |
| 08:20.31 | *** join/#asterisk Azam (n=azamzia@58-65-160-140.nayatel.pk) |
| 08:20.41 | Olobola | I'm in the santa cruz mountains |
| 08:21.03 | rrrobert | hello, Azam |
| 08:21.11 | Azam | hi |
| 08:21.19 | *** join/#asterisk LARefugee (n=chatzill@c-67-168-190-26.hsd1.wa.comcast.net) |
| 08:21.39 | rrrobert | nice to see you, how is your asterisk development going along |
| 08:21.44 | *** join/#asterisk kpanic (n=kpanic@pdpc/supporter/sustaining/kpanic) |
| 08:21.52 | *** part/#asterisk kpanic (n=kpanic@pdpc/supporter/sustaining/kpanic) |
| 08:25.00 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 08:26.01 | *** join/#asterisk mcinerney (n=john@peter.mcinerney.id.au) |
| 08:26.51 | *** join/#asterisk r0d3nt (i=nobody@pinky.ratman.org) |
| 08:33.27 | *** join/#asterisk sergey (n=sergey@91.189.233.71) |
| 08:38.46 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 08:43.04 | sergee | jblack: that's an interesting question, what is TTL for patents in the US? |
| 08:46.12 | sergee | jblack: you can check pat numbers here http://www.itu.int/ipr/IPRSearch.aspx?iprtype=PS you can view patents themselves here http://www.google.com/patents |
| 08:46.17 | WorgiL | how can i install codec_g729a.so ? |
| 08:47.19 | mosty | WorgiL, we told you already, you don't need to |
| 08:47.50 | WorgiL | mosty i found on debian server, downloaded but i don't know how can i install |
| 08:48.13 | mosty | WorgiL, you need to buy licenses from digium, have you done that yet? |
| 08:48.21 | WorgiL | no |
| 08:48.28 | WorgiL | i installed asterisk addon |
| 08:48.41 | Dr-Linux | anybody tried vicidial? |
| 08:49.27 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 08:49.33 | FuriousGeorge | hi all |
| 08:49.35 | mosty | WorgiL, g729 doesn't work in asterisk unless you buy licenses from digium |
| 08:49.43 | WorgiL | mosty i run menuselect but looking only mp3 format |
| 08:50.00 | WorgiL | how can i use gsm format on server |
| 08:50.16 | sergee | WorgiL: read it: |
| 08:50.19 | sergee | ~book |
| 08:50.19 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 08:50.22 | mosty | WorgiL, configure your sip.conf to use gsm. read the book |
| 08:50.30 | FuriousGeorge | i was just about to upgrade asterisk when i caught it... all of my landlines are locked... no one noticed because calls are routed through the Internet at telco when all the lines are tied up |
| 08:50.58 | FuriousGeorge | i would like to file a bug report, but i would like something more useful than just saying "all my lines are tied up" |
| 08:51.24 | WorgiL | mosty, thanks, |
| 08:51.46 | *** join/#asterisk af_ (n=getsmart@88-149-230-244.dynamic.ngi.it) |
| 08:53.43 | FuriousGeorge | anyone in here got any suggestions as to that? |
| 08:54.40 | mosty | zap show status ? |
| 08:55.40 | FuriousGeorge | mosty: sure but that wont look any different than normal usage |
| 08:56.02 | mosty | zap show channel 1 |
| 09:02.00 | mosty | are the zap channels listed in "show channels" ? |
| 09:02.06 | sergee | FuriousGeorge: what kind of lines do you have? |
| 09:02.30 | FuriousGeorge | verizon pstn lines in the USA |
| 09:02.33 | FuriousGeorge | serge |
| 09:02.55 | FuriousGeorge | mosty: where does zap show channel 1 say 'failed to detect hangup' ;) |
| 09:04.02 | mosty | FuriousGeorge, what version of asterisk is this? |
| 09:04.09 | FuriousGeorge | 1.4.6.2 |
| 09:04.15 | FuriousGeorge | i was just about to upgrade |
| 09:04.30 | FuriousGeorge | to 1.4.7 |
| 09:05.08 | sergee | FuriousGeorge: PRI? |
| 09:05.20 | FuriousGeorge | sangoma 201 |
| 09:05.24 | FuriousGeorge | chan_zap |
| 09:05.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:09.26 | FuriousGeorge | err sangoma a200 |
| 09:11.06 | mosty | FuriousGeorge, don't bother reporting it, and upgrade to 1.4.17 |
| 09:11.35 | mosty | and zaptel 1.4.8 and wanpipe 3.2.2 |
| 09:11.46 | FuriousGeorge | got hte latest wanpipe, i think |
| 09:11.57 | FuriousGeorge | maybe i need to play with busydetect again, if that does work |
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| 10:20.54 | _Azaz3lle_ | hola guys |
| 10:21.21 | _Azaz3lle_ | have someone expirienced such bug: When calling the echo test (600) and then putting it on hold taking the call off-hold afterwards gives the music on hold to the listener instead the echotest sounds. |
| 10:28.32 | jblack | You put echotest on hold? |
| 10:28.52 | jblack | So, basically, you got the wrong channel back after unholding, right? |
| 10:29.03 | _Azaz3lle_ | yep |
| 10:29.20 | jblack | weird. |
| 10:29.25 | _Azaz3lle_ | dunno, whether this is an asterisk bug, or dialplan one... |
| 10:29.43 | _Azaz3lle_ | btw, version is 1.4.10 |
| 10:30.04 | jblack | 1.4.17 is out. You could try that |
| 10:30.38 | _Azaz3lle_ | okay, i know, will try it with default configuration, and let you know. |
| 10:30.47 | _Azaz3lle_ | thx in advance. |
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| 10:33.11 | FlatFoot | morning all |
| 10:33.29 | FlatFoot | anyone got a real world example of zaptel.conf for the uk ? |
| 10:33.40 | FlatFoot | fro a TE210P |
| 10:33.43 | FlatFoot | *for |
| 10:38.15 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 10:42.35 | atop | inbound CLIP shows in a debug trace without the leading zero for the phone number, so the network is sending it like that, and they tell me that's just the way it is. How can I make inbound calls from the PRI show with the leading zero on the call logs and users phones? |
| 10:42.50 | Dr-Linux | huh |
| 10:42.55 | Dr-Linux | please see here: http://phpfi.com/291634 |
| 10:43.16 | Dr-Linux | vicidial.conf file exists but why it can't read it? |
| 10:43.16 | atop | ouch |
| 10:43.44 | atop | obvious thought... permissions? |
| 10:45.04 | Dr-Linux | atop: this file has enough permissions |
| 10:45.23 | Dr-Linux | -rw-rw-r-- 1 root root 15071 Jan 23 15:34 vicidial.conf |
| 10:45.51 | atop | asterisk is trying to read the file tho right? is * running as root? |
| 10:45.58 | atop | or am I way off base here? |
| 10:46.14 | Dr-Linux | yes as root |
| 10:48.10 | atop | the error looks odd, I've not seen it before or used vicidial, but I'd expect the error to show the name of just the file it was after, so 'vicidial.conf' rather than '=> vicidial.conf' It's likely an issue with the code that generates the error, but is it possible theres an error in the syntax of the file that calls the include? |
| 10:48.45 | atop | it's almost like its looking for a file with the name of "=> vicidial.conf" which would explain it not finding your file |
| 10:49.43 | jblack | Heh. voip-info.org's account registration is bugged |
| 10:51.19 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:51.28 | Dr-Linux | fixed :) |
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| 10:56.40 | J4zen | Has anyone ever used Outcall before? |
| 10:59.11 | atop | any ideas on fixing the inbound CLI from the provider to add a zero to the beginning? |
| 10:59.25 | DarKnesS_WolF | i have a very wirdo case ... if i crete a call from asterisk to a provider the qulati is okay but not great |
| 10:59.28 | defswork | atop: I do that |
| 10:59.35 | DarKnesS_WolF | if i accept a call from the provider or from a DID the qulati is amazing ! |
| 10:59.42 | DarKnesS_WolF | how can i debug and fix this ? |
| 10:59.57 | atop | defswork, you may be may saviour! Can I ask what you have done? |
| 11:00.07 | defswork | exten => _X.,1,Set(CALLERID(number)=0${CALLERID(number)}) |
| 11:00.27 | atop | I was looking in that area, yes |
| 11:00.45 | atop | so you make that change in the context for the trunk that needs modding, right? |
| 11:00.54 | defswork | yup |
| 11:02.11 | atop | I'll have a go and run some tests, thanks. |
| 11:02.22 | J4zen | Does anyone have some kind of Office integration for Outcall? |
| 11:03.11 | J4k3 | DarKnesS_WolF: sounds like you're using one codec in one direction, and a different codec in the other direction |
| 11:03.44 | J4k3 | the best quality will be obtained by forcing ulaw or alaw (allow=ulaw;disallow=all) |
| 11:04.04 | J4k3 | at a cost of 80kbit (each way) per call |
| 11:04.27 | DarKnesS_WolF | J4k3: all are g729 for testing i'm forcing g729 |
| 11:04.34 | J4k3 | funky |
| 11:04.37 | DarKnesS_WolF | PBX ---<> Zap ---* ----> g729 provder in both |
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| 11:09.11 | Mavvie | something is seriously wrong with AGI on 1.4.17 |
| 11:09.28 | Mavvie | I use two AGI->get_data strings after each other, with 10000 ms timeout.... |
| 11:09.49 | Mavvie | and I hear the second and the data get sstored in the return value of the first. |
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| 11:11.01 | nebojsajsimic | hi all |
| 11:11.33 | nebojsajsimic | is there some docs for configuring Asterisk with mysql ??? |
| 11:11.38 | nebojsajsimic | ??? |
| 11:13.13 | Mavvie | hmm... seems to be because I didn't use AGI->ReadParse() |
| 11:13.14 | jblack | probably somewhere on google. certainly in the book, at about pg 265 |
| 11:13.15 | Mavvie | very strange. |
| 11:14.34 | defswork | and just dial the extension if not |
| 11:15.47 | nebojsajsimic | in book is only PostgreSQL example |
| 11:15.49 | Azam | nebojsajsimic try this link http://www.ucl.ac.uk/is/mysql/c/ |
| 11:16.00 | nebojsajsimic | thx |
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| 11:26.03 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 11:30.01 | maagic | hi! i have quad e1 pri card. i have one e1 connected to pstn and second e1 to another pbx. is it possible to "redirect" all calls from pstn to another pbx? or should i ask how can this be done. |
| 11:30.22 | Mavvie | maagic: easiest piece of work. |
| 11:30.30 | Mavvie | create two contexts, one for each PRI. |
| 11:30.41 | maagic | yes, i have done that |
| 11:30.49 | Mavvie | then all calls from context one have to be dialed to the other group. |
| 11:30.53 | Mavvie | and visa versa. |
| 11:31.11 | maagic | i've tried the s -extension, but it didn't work? |
| 11:31.58 | maagic | in my case the extensions must be same all the way to second pbx |
| 11:32.19 | Mavvie | exten => _X.,1,Dial(Zap/G1/${EXTEN}) |
| 11:32.25 | Mavvie | that's all what is needed. |
| 11:33.15 | Mavvie | if it doesn't work, run asterisk in verbose so you can see what happens. |
| 11:33.21 | *** join/#asterisk aleff (n=alef@c9063318.virtua.com.br) |
| 11:33.24 | aleff | hi |
| 11:33.38 | aleff | how can I use h323 within users.conf? |
| 11:33.44 | jblack | can the FOP use the mini asterisk webserver, or will I need apache? |
| 11:33.53 | aleff | it looks like it only accepts options like hassip or hasiax but lacks h323 |
| 11:33.54 | maagic | ah. i thought it should be like that.. thanx a lot :) |
| 11:35.14 | maagic | i just forgot the beginning _ from my own configuration :) |
| 11:35.14 | Mavvie | maagic: but be careful... it's the first step to a 3500 line extensions.conf :-) |
| 11:35.42 | Mavvie | (of course 2500 of them are auto-generated :-) |
| 11:36.02 | maagic | :) |
| 11:36.39 | J4zen | I'm fairly desperate for Office(Outlook) Asterisk integration, does anyone know of a solution for it? |
| 11:37.15 | maagic | this is my first asterisk project and before i've got this ready employee already booked me in another asterisk project.. we'll see how it's going to work out :) |
| 11:37.18 | aleff | J4zen does office support sip connections as sip client? |
| 11:37.20 | Mavvie | J4zen: I'm pretty sure you can make a mighty buck with it. |
| 11:37.35 | J4zen | I know |
| 11:37.39 | J4zen | i have a ton of customers requesting it |
| 11:37.47 | J4zen | thats why im looking for it |
| 11:39.56 | Mavvie | J4zen: problem with these kind of things is that they require knowledge out outlook. |
| 11:40.02 | Mavvie | or whatever you are using. |
| 11:40.05 | J4zen | yup |
| 11:40.18 | J4zen | well i woudlnt be surprised if someone coded a plugin for it already |
| 11:40.41 | J4zen | asterisk being more populair than ever, combined with the popularity of Outlook |
| 11:40.57 | J4zen | id have guessed someone beat me to it |
| 11:41.07 | Mavvie | working together with somebody who knows outlook, and then write a simple RPC thingie which drops a call file inside asterisk. |
| 11:41.08 | Mavvie | easy. |
| 11:41.13 | Mavvie | oh, except for the outlook part. |
| 11:41.27 | J4zen | yep heh |
| 11:42.59 | Mavvie | so, invest in a windows programmer, and license your program to the people who want to use it. |
| 11:43.25 | J4zen | ah i see |
| 11:43.33 | J4zen | Fonality ( ofcourse ) have one |
| 11:43.41 | J4zen | only shipped with their PBXtra though |
| 11:43.46 | J4zen | 995$ for minimal package |
| 11:43.58 | Mavvie | told you you could make money out of it. |
| 11:44.02 | J4zen | hehe |
| 11:44.47 | Mavvie | people who run outlook often are in companies which can afford to buy these tools. |
| 11:45.14 | Mavvie | If we were using Asterisk as the PABX for customers we would surely be interested. |
| 11:45.24 | J4zen | yteah |
| 11:46.02 | Mavvie | specially since it gives us an extra over competitors. |
| 11:46.15 | Mavvie | upsell upsell upseel! |
| 11:47.02 | J4zen | hehe |
| 11:47.08 | J4zen | u run your own company or? |
| 11:47.16 | Mavvie | or? |
| 11:54.07 | tzanger | what's the A in PABX anyway? I always see that |
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| 11:58.46 | Mavvie | Access |
| 11:59.21 | Mavvie | automatic! |
| 12:01.37 | *** join/#asterisk zaur_pronet (n=zaur_pro@85.132.55.134) |
| 12:02.27 | Mavvie | versus manual. |
| 12:07.23 | mosty | where does menuselect save its options? |
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| 12:12.15 | tzafrir | menuselect.makeopts |
| 12:12.35 | tzafrir | this is the file to delete if you want 'make' to Just Work[tm] |
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| 12:14.31 | mosty | tzafrir, i want to run menuselect once, then save the settings and be able to drop the settings in before i compile next time, so i can build my packages non-interactively |
| 12:15.08 | tzafrir | sadly, menuselect is not smart enough. This will work only if your config will not change |
| 12:15.29 | tzafrir | A more robust method: |
| 12:15.50 | *** join/#asterisk Stefan1979 (n=stan@4806ds1-hl.0.fullrate.dk) |
| 12:16.00 | tzafrir | ./configure; rm menuselect.makeopts; make menuselect.makeopts; edit menuselect.makeopts; make |
| 12:16.13 | RockHound | good day... could someone help me debug why I can not get a context coming from mISDN to switch to DUNDi? SIP contexts work perfectly with same include. |
| 12:16.19 | tzafrir | and hopefully automatic editing of menuselect.makeopts will work |
| 12:16.28 | mosty | tzafrir, the config never changes, i just want it to build everything |
| 12:16.39 | tzafrir | skip the edit part |
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| 12:36.40 | JT | anyone knowledgeable in aastras about? |
| 12:36.51 | JT | does a factory reset key sequence even exist for these bloody things? |
| 12:36.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 12:44.15 | Stefan1979 | Help! Geek in need :-).. I'm trying to install a e100p pri card.. - but when "module load chan_zap" gives me to following error: chan_zap.c:903 zt_open: Unable to specify channel 1: No such device or address |
| 12:44.42 | Stefan1979 | (and genzaptel finds the cards; but it doesn't configure any channels) |
| 12:44.45 | mosty | Stefan1979, does ztcfg -vvvv succeed? |
| 12:44.58 | *** join/#asterisk af_ (n=getsmart@88-149-230-244.dynamic.ngi.it) |
| 12:45.36 | Stefan1979 | mosty: 0 channels configured. |
| 12:45.37 | Stefan1979 | Nope' |
| 12:46.09 | mosty | Stefan1979, what if you manually create zaptel.conf ? |
| 12:46.21 | Stefan1979 | 2 sec i'll try |
| 12:46.39 | yang | I have a problem with zaptel device - unable to open master device http://paste.debian.net/47498 ; lspci -v = 03:02.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 12:47.25 | Stefan1979 | mosty: much better.. |
| 12:47.45 | mosty | yang, does the device file exist? can you pastebin the output of lsmod |
| 12:47.58 | yang | mosty: lsmod | grep ... ? |
| 12:48.16 | mosty | just lsmod |
| 12:48.50 | yang | http://paste.debian.net/47502 |
| 12:49.38 | mosty | yang, the zaptel module is not loaded, obviously /dev/zap/ctl won't exist |
| 12:49.57 | skopii | yang: did you compile the module? |
| 12:50.22 | yang | mosty: modprobe zaptel ; FATAL: Module zaptel not found. |
| 12:50.35 | mosty | yang, you don't appear to have zaptel installed |
| 12:50.42 | skopii | http://downloads.digium.com/pub/zaptel/ |
| 12:50.49 | yang | skopii: with aptitude install zaptel & genzaptelconf -sdvM |
| 12:51.08 | skopii | apt-cache search zap | grep -i dummy |
| 12:51.27 | mosty | yang, the asterisk zaptel package only includes the userspace utils, it does not include the kernel modules |
| 12:51.37 | yang | oh |
| 12:52.05 | yang | so aptitude install zaptel-source & m-a a-i zaptel |
| 12:52.42 | atop | defswork, you around? I've been playing with that adding a zero prefix for inbound calls, it works but it's not "low level" enough, ie, it does not affect the number stored in the CDR etc |
| 12:52.53 | mosty | yang, yeah |
| 12:53.00 | atop | I recon I may go back to the network provider :/ |
| 12:54.06 | skopii | atop: not sure about this but I would guess the cdr just stores the CID information, so you can use Set() in the dialplan to set the cid |
| 12:54.27 | *** join/#asterisk WindBack (n=Jorge@proxy.unc.edu.ar) |
| 12:54.59 | mosty | atop, what are you trying to do? |
| 12:55.58 | *** join/#asterisk cbx33 (n=pete@ubuntu/member/cbx33) |
| 12:56.02 | cbx33 | hey guys |
| 12:56.02 | cbx33 | need a little help |
| 12:56.18 | cbx33 | I was running ubuntu version of asterisk fine.... |
| 12:56.34 | cbx33 | I apt-get removed, and then compiled the new version from source and make installed it |
| 12:56.40 | cbx33 | it runs |
| 12:56.42 | cbx33 | but it complains |
| 12:56.49 | cbx33 | Starting Asterisk PBX: /var/run/asterisk/asterisk.pid |
| 12:56.49 | cbx33 | Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 12:56.49 | cbx33 | asterisk. |
| 12:56.58 | nixguy | you stating it as root? |
| 12:57.02 | cbx33 | yes |
| 12:57.06 | nixguy | can you touch it |
| 12:57.10 | nixguy | touch /var/run/asterisk.pid |
| 12:57.16 | nixguy | if it doesent exist |
| 12:57.16 | cbx33 | the bit after starting asterisk ... is me echoing out the PID variable from the startup script |
| 12:57.30 | cbx33 | so it shouldn't be putting it there |
| 12:57.37 | cbx33 | plus if I cat out the config |
| 12:58.03 | cbx33 | astrundir => /var/run/asterisk |
| 12:58.03 | cbx33 | astlogdir => /var/log/asterisk |
| 12:58.10 | cbx33 | the run dir is set there too |
| 12:58.33 | skopii | cbx33: not sure about ubuntu but you may want to check /etc/default/asterisk to see if its configured to drop root after it starts |
| 12:58.44 | cbx33 | it is |
| 12:58.46 | cbx33 | and that's ok |
| 12:58.53 | cbx33 | /var/run/asterisk is set up for that |
| 12:59.00 | cbx33 | but asterisk seems to be ignoring this |
| 12:59.27 | skopii | /var/run/asterisk or /var/run/asterisk.pid? |
| 12:59.35 | Stefan1979 | mosty: everything seems to work; except that i still can't load chan_zap |
| 12:59.53 | cbx33 | Unable to open pid file '/var/run/asterisk.pid': Permission denied |
| 12:59.53 | cbx33 | Unable to bind socket to /var/run/asterisk.ctl: Permission denied |
| 13:00.02 | cbx33 | but as you can see....from the config |
| 13:00.08 | cbx33 | it should be trying to put it elsewhere |
| 13:00.27 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 13:00.43 | mosty | Stefan1979, so ztcfg -vvvv shows all the channels you configured? and what are the permissions on the device files? |
| 13:00.56 | cbx33 | and nixguy yes that does get the problem to go away, however I still can't connect to the console |
| 13:01.04 | tzafrir | yang, m-a a-i zaptel |
| 13:01.06 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:01.06 | cbx33 | so i want to find out why it's not honouring the config |
| 13:01.22 | tzafrir | if you don't have zaptel-modules-`uname -r` available |
| 13:01.28 | FlatFoot | can anyone help with this err ... ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 13:01.46 | FlatFoot | trying to get a te210p conf'd for E1 in the uk |
| 13:01.51 | tzafrir | FlatFoot, head /proc/zaptel/1 |
| 13:02.02 | tzafrir | is it actually the span you intended? |
| 13:02.02 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 13:02.03 | yang | tzafrir: thanks it works now |
| 13:02.10 | skopii | cbx33: check safe_asterisk ;] |
| 13:02.23 | cbx33 | safe_asterisk |
| 13:02.25 | cbx33 | a file? |
| 13:02.38 | FlatFoot | i am trying to get it to work but it does not seem to like span=1,1,0,ccs,hdb3,crc4 |
| 13:02.40 | skopii | /usr/sbin/safe_asterisk is a "wrapper" script that the ubuntu init script is probably calling |
| 13:02.48 | skopii | ASTPIDFILE=/var/run/asterisk/asterisk.pid |
| 13:03.02 | cbx33 | what is safe_asterisk? |
| 13:03.08 | FlatFoot | head /proc/zaptel/1 : No such file or directory |
| 13:03.11 | skopii | its a script in /usr/sbin/safe_asterisk |
| 13:03.17 | cbx33 | ahh ok |
| 13:03.19 | FlatFoot | on freebsd btw |
| 13:03.38 | skopii | FlatFoot: you can try mount -t procfs none /proc? |
| 13:03.43 | cbx33 | skopii, does it use that file? |
| 13:03.54 | atop | mosty, sorry was afk. you still around? |
| 13:03.54 | skopii | cbx33: I am guessing...yes |
| 13:03.58 | skopii | so it can't hurt to try |
| 13:04.11 | Stefan1979 | mosty: http://pastebin.com/d2fdfc141 |
| 13:04.18 | FlatFoot | skopii: does not return anything |
| 13:04.34 | cbx33 | nope |
| 13:04.45 | skopii | FlatFoot: well that means procfs is mounted on proc now..but I don't think FreeBSD drivers export information to /proc like some linux drivers do... |
| 13:05.42 | mosty | Stefan1979, what user is asterisk running as? |
| 13:05.50 | cbx33 | skopii, no effect |
| 13:05.53 | FlatFoot | aha i am waiting port cvsup to fininsh then i'm gonna portupgrade zaptel and * to latest port version , see if that helps |
| 13:06.07 | JT | bloody aastras |
| 13:06.12 | skopii | cbx33: and your *sure* the permissions are set correctly on /var/run/asterisk? |
| 13:06.17 | cbx33 | yes |
| 13:06.19 | JT | no non-password protected factory reset option |
| 13:06.23 | cbx33 | but it's not even looking there |
| 13:06.34 | JT | i had to packet dump an spoof the old tftp server it was looking for |
| 13:06.39 | Stefan1979 | mosty: root |
| 13:06.52 | cbx33 | drwxr-xr-x 2 asterisk asterisk 40 2008-01-23 12:31 asterisk |
| 13:06.58 | skopii | cbx33: maybe try "strace -f -s 16384 /etc/init.d/asterisk restart >out 2>&1" |
| 13:07.05 | cbx33 | ko |
| 13:07.05 | Stefan1979 | mosty: It's seems to be failure 40... |
| 13:07.18 | skopii | and paste the file "out" into a pastebin |
| 13:07.31 | Stefan1979 | mosty: i defined channel 1-9.. instead of 1-8 |
| 13:09.47 | cbx33 | skopii that is ludicrous |
| 13:09.48 | *** join/#asterisk af_ (n=getsmart@88-149-230-64.dynamic.ngi.it) |
| 13:10.10 | skopii | cbx if its a *lot* of text try changing the 16384 to 255 or something |
| 13:10.21 | cbx33 | well it shows very little |
| 13:10.40 | skopii | what does it show cbx33 ? |
| 13:11.19 | cbx33 | 22000 lines |
| 13:11.25 | cbx33 | too much for a paste bin? |
| 13:11.26 | cbx33 | ;) |
| 13:11.38 | skopii | very little lol? |
| 13:11.54 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:12.32 | skopii | cbx33: really the only stuff that is relevant is before the "permission denied" |
| 13:13.37 | skopii | you may be able to filter the output doing something like "grep -B 9999 -i "permission denied" > out2 |
| 13:13.49 | cbx33 | no no |
| 13:13.50 | cbx33 | I have it |
| 13:13.58 | cbx33 | just trying to find something before it that's useful |
| 13:14.29 | skopii | it would help if I could actually see the init script too |
| 13:14.44 | cbx33 | it should be the standard asterisk init script |
| 13:15.58 | skopii | and there's nothing in /etc/default/asterisk? |
| 13:16.07 | atop | clutching at straws: Anyone configured a system in the UK using a PRI line provided by Cable & Wireless ? |
| 13:16.13 | cbx33 | skopii, http://pastebin.ca/869680 |
| 13:16.38 | FlatFoot | skopii: sorted ! just upgraded to 1.4.8 and she fires up a treat . now upgrading * |
| 13:17.05 | *** join/#asterisk aikanaro79 (n={aikanar@89-180-85-152.net.novis.pt) |
| 13:17.08 | cbx33 | skopii, http://pastebin.ca/869681 |
| 13:18.03 | cbx33 | skopii, they weere the bits I found interesting |
| 13:18.33 | defswork | atop: isn't it just backed off onto BT ? |
| 13:18.41 | aikanaro79 | hi...I'm developing an application to interact with asterisk using SIP only...I'm able to register with success but when I try to dial someone asterisk replies that it cannot authenticate my user |
| 13:18.45 | atop | no idea really |
| 13:18.50 | defswork | atop: I've got 2 installs with BT PRIs - with the same cli deficit as yours ;) |
| 13:18.52 | aikanaro79 | ideas? if I try with another client program it works fine |
| 13:19.07 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 13:19.11 | atop | defswork, I also cant dial out |
| 13:19.24 | defswork | atop: what PRI card ? |
| 13:19.35 | atop | Sangoma A102 |
| 13:19.39 | defswork | snap |
| 13:19.47 | defswork | have you set the data channel correct ? |
| 13:19.51 | *** join/#asterisk ming_zym (n=ming_zym@124.14.238.104) |
| 13:19.51 | defswork | and |
| 13:20.13 | defswork | theres a setting you need |
| 13:20.17 | defswork | one sec |
| 13:20.27 | atop | I'll pastebin my configs |
| 13:21.08 | defswork | pridialplan=unknown |
| 13:21.08 | defswork | prilocaldialplan=unknown |
| 13:21.11 | atop | zapata.conf http://pastebin.ca/869683 |
| 13:21.12 | defswork | in zapata-auto.conf |
| 13:21.17 | cbx33 | skopii, what ya think? |
| 13:21.19 | defswork | without that you can't dial out :) |
| 13:21.30 | skopii | cbx33: it boggles my mind..which version are you using? |
| 13:21.43 | defswork | I had the very same probelm on my first install (when I was even more clueless than I am now) |
| 13:21.51 | cbx33 | 1.4.17? |
| 13:21.58 | atop | that was zaptel, this is zapata http://pastebin.ca/869688 |
| 13:22.08 | skopii | cbx33: like I said it boggles my mind |
| 13:22.17 | cbx33 | sh*t |
| 13:22.23 | cbx33 | i need this working fast |
| 13:22.29 | atop | the dialplan options are in there already |
| 13:22.34 | skopii | cbx33: you might want to try making a symlink |
| 13:22.49 | skopii | but it will probably get deleted |
| 13:22.51 | Stefan1979 | mosty: thanks for the helps.. it seems to work now :-D |
| 13:22.51 | cbx33 | but even when I made them, the ctl file, which is a sock bind by the looks of things didn't work |
| 13:22.57 | cbx33 | it didn't help |
| 13:23.10 | skopii | cbx33: you can't "make" a socket |
| 13:23.20 | cbx33 | you know what I mean |
| 13:23.26 | tzanger | some people would say they could make it with a socket... |
| 13:23.30 | defswork | atop: in my zaptel.conf I have dchan=16 |
| 13:23.31 | skopii | I would suggest probably pulling the source and compiling |
| 13:23.42 | skopii | well to make a sock you would need to make a hardlink to an existing socket |
| 13:23.52 | skopii | well actually |
| 13:24.06 | atop | defswork, I used to have that set, but the lastest wanpipe driver changed it - the card has hardware EC |
| 13:24.55 | defswork | atop: I have a 102D install too and dchan=16 in that too |
| 13:25.35 | mosty | atop, i've never seen an E1 provider where the D-channel wasn't 16 |
| 13:25.52 | defswork | mosty: he means he hasn't got dhcan=16 in there at all |
| 13:25.58 | mosty | but i think if you set it to 0 in wanpipe, it will autodetect |
| 13:26.14 | atop | is this not the same thing ? hardhdlc=16 |
| 13:26.24 | defswork | doesn't look like it |
| 13:26.31 | atop | wanpipe 3.2.2 |
| 13:26.49 | *** part/#asterisk admin0 (n=admin@116.90.228.34) |
| 13:26.52 | atop | inbound calls do work tho, so the D chan would be operational in order for that to work? |
| 13:27.20 | skopii | cbx33: http://rafb.net/p/UpQI5z95.html |
| 13:27.42 | tzafrir | hardhdlc is a d channel with a different way to move the data |
| 13:28.35 | defswork | atop - change it to dchan=16 |
| 13:28.41 | atop | I'll change it and try again |
| 13:28.44 | defswork | it's a documented feature of wansetup |
| 13:29.03 | atop | Thing is, I can only test first thing in the morning as I have to disconnect the line from the old phone system |
| 13:29.14 | defswork | :| |
| 13:29.14 | atop | cant commission the new asterisk until everything is working! |
| 13:30.02 | atop | just add the dchan line or remove the hdlc line also? |
| 13:30.10 | defswork | remove also |
| 13:30.15 | skopii | cbx33: you could always recompile asterisk and change the file "defaults.h" changing the #define AST_SOCKET and AST_PID |
| 13:30.21 | atop | done ;) |
| 13:30.22 | defswork | atop: http://www.trixbox.org/forums/vendor-moderated-forums/sangoma/howto-install-sangoma-drivers-source-tb-2-3-0-10-80 |
| 13:30.33 | cbx33 | ok |
| 13:30.41 | atop | um.... I'll get back to you in about 18 hours! |
| 13:30.45 | defswork | :D |
| 13:30.48 | defswork | good luck |
| 13:30.51 | cbx33 | I'll do that |
| 13:30.54 | defswork | atop: what handsets you using btw ? |
| 13:31.03 | atop | aastra 57i |
| 13:31.03 | cbx33 | would I have to make uninstall first skopii |
| 13:31.16 | defswork | atop: cool - I'm all aastra kit so far too |
| 13:31.27 | defswork | 9133i though (and a 480i and a 55i) |
| 13:31.34 | skopii | cbx33: don't see why you would the new install would overwrite everything |
| 13:31.40 | atop | were also using poly 320's in the remote site, smaller, cheaper etc |
| 13:31.52 | defswork | how are you linking sites ? |
| 13:31.56 | cbx33 | it's funny everything else was right |
| 13:32.14 | skopii | cbx33: what do you mean? |
| 13:32.23 | atop | remote site has asdl connection, and I use an IAX trunk to allow desk to desk calling |
| 13:32.32 | skopii | as far as the config and what not goes? |
| 13:32.41 | cbx33 | yes |
| 13:32.41 | defswork | atop: 2 asterisk installs then ? |
| 13:32.43 | cbx33 | very odd |
| 13:32.56 | atop | yes |
| 13:32.59 | cbx33 | just rebuilding now |
| 13:33.04 | cbx33 | skopii, maybe this is a bug |
| 13:33.07 | skopii | it is probably just some bug in * |
| 13:33.11 | atop | admin nightmare, but helps for resilience |
| 13:33.18 | defswork | atop: I have similar from home to customer sites over openvpn |
| 13:33.19 | skopii | I really doubt digium cares, you could always file a bug report though |
| 13:33.24 | atop | like if our system goes down, they can still make calls |
| 13:33.31 | skopii | I would actually be curious to see what they have to say about it |
| 13:33.35 | cbx33 | surely it must affect otherse too ;) |
| 13:33.41 | lilalinux | which variable holds the callerid? (asterisk 1.2) |
| 13:33.59 | mosty | lilalinux, look it up on the wiki |
| 13:34.01 | skopii | cbx33: works fine in gentoo |
| 13:34.20 | cbx33 | very odd |
| 13:34.20 | cbx33 | cos it's running as root too |
| 13:34.21 | cbx33 | i mean the init script |
| 13:34.36 | cbx33 | oh i just don't get it |
| 13:34.37 | *** join/#asterisk RoyKa (n=roy@91.149.26.242) |
| 13:34.40 | cbx33 | ok recompiling |
| 13:34.45 | skopii | cbx33: you could always just run * as root |
| 13:34.49 | defswork | lilalinux: CALLERID(number) |
| 13:34.52 | cbx33 | ummm.......no |
| 13:34.57 | skopii | ;] |
| 13:34.59 | cbx33 | too much of a sec risk |
| 13:35.28 | lilalinux | mosty, defsdoor I only find information about setting the callerid, I want to store the callerid of the caller |
| 13:35.34 | lilalinux | (for callback) |
| 13:35.36 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 13:36.24 | mosty | lilalinux, put ${ and } around it |
| 13:37.40 | atop | defswork, if you get a chance can you compare this wanpipe1.conf to yours ? http://pastebin.ca/869698 |
| 13:37.56 | naxxatoe | does anybody know whats causing theese errors? |
| 13:37.56 | defswork | doing it now |
| 13:37.57 | naxxatoe | [Jan 23 13:26:01] WARNING[2586] chan_sip.c: sip_xmit of 0xb63d0dcc (len 548) to 10.211.55.5:5067 returned -2: Network is unreachable |
| 13:38.12 | mosty | naxxatoe, the network is unreachable...? |
| 13:38.12 | naxxatoe | i am currently trying to get ekiga to work -.- |
| 13:38.17 | lilalinux | naxxatoe: try a traceroute to that ip |
| 13:38.18 | naxxatoe | shouldnt be |
| 13:38.30 | mosty | naxxatoe, test it with traceroute |
| 13:38.39 | skopii | naxxatoe: network unreachable means you dont have a default gw set and the host is on a diff subnet |
| 13:38.54 | naxxatoe | thx4info |
| 13:39.00 | naxxatoe | gonna get this fixed now |
| 13:39.45 | defswork | atop: - idenitickle to mine |
| 13:39.45 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:39.53 | atop | good to know, thanks. |
| 13:40.00 | defswork | mine works too :) |
| 13:40.01 | naxxatoe | nope |
| 13:40.04 | naxxatoe | i can ping both |
| 13:40.10 | naxxatoe | and traceroute both ways |
| 13:40.18 | naxxatoe | plus they are on the same subnet -.- |
| 13:40.23 | skopii | naxxatoe: do you have some kind of firewall? |
| 13:40.30 | *** join/#asterisk tiav (n=tiav@inv75-3-82-241-117-16.fbx.proxad.net) |
| 13:40.33 | naxxatoe | not that i knew |
| 13:40.34 | naxxatoe | -.- |
| 13:40.37 | cbx33 | skopii, |
| 13:40.41 | cbx33 | it's still doing it |
| 13:40.48 | cbx33 | even after the recompile |
| 13:40.50 | atop | defswork, are you doing anything to the outbound number format? I mean, do you make it send in international format, or any other changes? |
| 13:41.07 | skopii | lol cbx33 that just *BOGGLESMYMIND* |
| 13:41.10 | defswork | atop: you need to nag BT to alter the outgoing CLI |
| 13:41.13 | cbx33 | mine too |
| 13:41.28 | defswork | atop: they don't allow it by default |
| 13:41.50 | defswork | atop: on one site I change the cli to their 0870 number |
| 13:41.54 | atop | should be ok, as the current/old panasonic system would sent the correct CLI for the department being dialled form |
| 13:42.13 | skopii | cbx33: did you edit the defaults.h file after you ran configure ? |
| 13:42.21 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 13:42.26 | cbx33 | no |
| 13:42.43 | tzanger | I'm watching last week's rick mercer report |
| 13:42.50 | tzanger | where he did the polar bear dip |
| 13:42.51 | cbx33 | ahhh |
| 13:42.53 | cbx33 | ok |
| 13:42.53 | skopii | that could be why...but it could also be the AST_RUN_DIR |
| 13:42.55 | cbx33 | thanks skopii |
| 13:43.02 | tzanger | that's hilarious... he's feeling his throat... "oh, those are my testicles" hahaha |
| 13:43.03 | cbx33 | i changed that too |
| 13:43.07 | cbx33 | do i need to make clean? |
| 13:43.21 | cbx33 | i guess i do really |
| 13:43.40 | skopii | cbx33: you should be able to just do make clean && make && make install |
| 13:43.46 | cbx33 | ok |
| 13:44.54 | naxxatoe | fucking gateway error -.- |
| 13:45.41 | *** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr) |
| 13:45.53 | mamep | a quick question about dialplan |
| 13:45.57 | hmmhesays | anyone in here ever use buildroot? |
| 13:46.14 | mamep | how is it possible to redirect to a sip provider every call that starts with a given pattern 92? |
| 13:46.56 | mosty | mamep, not sure what you mean exactly, but yes |
| 13:47.38 | mamep | mosty : i want to send all calls starting with a pattern 92 to a sip provider and add before it 00302 |
| 13:47.50 | mosty | mamep, yes that is simple |
| 13:48.11 | mamep | an example please |
| 13:48.21 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 13:48.34 | mamep | exten => _92XXXXXXXXX,1,Dial(SIP/${EXTEN}@trunk_6,45,Rtj) |
| 13:48.40 | mamep | i was trying something like this |
| 13:48.43 | atop | exten => _92.,1,Dial(SIP/provider/00302$[exten]) |
| 13:49.04 | atop | but that's a guess |
| 13:49.42 | mamep | hmm |
| 13:49.43 | mamep | let me try |
| 13:49.51 | Dr-Linux | anybody tried vicidial? |
| 13:49.54 | cbx33 | skopii, |
| 13:49.56 | cbx33 | that fixed it |
| 13:50.29 | skopii | cbx33: it's probably the chicken before the egg thing... |
| 13:50.46 | ManxPower | Everyone knows eggs came first. |
| 13:51.35 | mamep | atop: it's not working something else i want to remove 92 from the beggining and replace it with 00302 |
| 13:52.01 | mosty | mamep, in your dial command, use SIP/trunk_6/00302${EXTEN} |
| 13:52.05 | skopii | mamep: 00303${EXTEN:0}? |
| 13:52.28 | cbx33 | thanks skopii |
| 13:52.29 | mamep | what's the difference skopii ? |
| 13:52.48 | skopii | mamep: the :0 part matches the part after _92 |
| 13:52.52 | ManxPower | exten => _92.,1,Dial(SIP/provider/00302${EXTEN:2}) |
| 13:53.09 | ManxPower | no, the :0 part does nothing. |
| 13:53.10 | skopii | ManxPower: whats the difference? |
| 13:53.12 | cbx33 | that was a very annoying bug |
| 13:53.18 | ManxPower | As it says "trip the first 0 digits" |
| 13:53.24 | mamep | let me try guys |
| 13:53.25 | ManxPower | trip=trin |
| 13:54.04 | *** join/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 13:54.50 | skopii | ManxPower: I have ${EXTEN:1} in mine actually... |
| 13:55.29 | skopii | ahhh trip=strip |
| 13:55.44 | ManxPower | If you don't know the difference between ${EXTEN:0} and ${EXTEN:2}, then you should me watching not helping. |
| 13:56.01 | ManxPower | skopii: I have not had my first cup of coffee yet. |
| 13:56.08 | nebojsajsimic | hi all |
| 13:56.09 | fiXXXerMet | Trying to get festival working with asterisk and am getting errors. Info @ http://pastebin.com/m200cc4d1 |
| 13:56.22 | nebojsajsimic | make[1]: Leaving directory `/home/admin/install/asterisk-addons-1.2.8/asterisk-addons-1.2.8/format_mp3' |
| 13:56.22 | nebojsajsimic | ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql `ls *.c` |
| 13:56.27 | ManxPower | ~pb |
| 13:56.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:56.35 | *** join/#asterisk grayhame (n=grayhame@ns2.dalcon.com) |
| 13:56.45 | nebojsajsimic | how to fix this???? |
| 13:56.59 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:56.59 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:57.01 | nebojsajsimic | i cant install ast-add |
| 13:57.04 | mosty | nebojsajsimic, there is no error there, nothing to fix |
| 13:57.29 | ManxPower | not many people use Festival or Asterisk addons, so you might have better luck looking on the mailing list archives or the Wiki |
| 13:57.31 | WindBack | We have an AEX800 with 2x4 FXO modules and a cancelation hardware. What module I have to enable in /etc/default/zaptel ?? |
| 13:57.46 | mamep | ok working with :2 |
| 13:57.46 | nebojsajsimic | ./mkdep: Permission denied |
| 13:57.46 | Qwell | WindBack: wctdm2400 I think |
| 13:57.47 | mamep | thx |
| 13:57.49 | ManxPower | WindBack: What is an AEX800? |
| 13:57.57 | mamep | something else how can i set caller id to hidden? |
| 13:58.02 | Qwell | ManxPower: pci-e 8 port analog card |
| 13:58.03 | mosty | nebojsajsimic, chmod +x mkdep |
| 13:58.08 | ManxPower | Qwell: Digium? |
| 13:58.09 | nebojsajsimic | thx |
| 13:58.11 | Qwell | yes |
| 13:58.20 | Qwell | ~tdm800, except pci-e |
| 13:58.29 | WindBack | ManxPower, http://www.digium.com/en/products/analog/aex800.php |
| 13:58.39 | ManxPower | Ah. I don't use analog |
| 13:58.57 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:59.05 | WindBack | Qwell, thank |
| 13:59.25 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 13:59.38 | Qwell | wctdm24xxp sounds more right, actually |
| 14:00.00 | WindBack | Qwell, And.. how i tell asterisk to use the echo cancelation hardware instead of the echo cancelation software ?? |
| 14:00.26 | ManxPower | Qwell: I see Digium has STILL not fired the person that comes up with these product model numbers. |
| 14:03.35 | ManxPower | WindBack: all the digital digium products automatically use the hardware EC if you have the hardware EC installed. Check the output of dmesg for mentioning VPM in the messages about the driver loading |
| 14:04.18 | tzanger | the PCIe x1 connectors look so silly |
| 14:04.44 | ManxPower | We use all external EC boxes, so I'm just repeating what I have learned after 4 years of using Asterisk. |
| 14:04.49 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 14:05.29 | WindBack | ManxPower, Exist echo in a digital chanel ??? |
| 14:05.47 | tzanger | WindBack: absolutely! |
| 14:06.29 | ManxPower | WindBack: ALL EC is digital. I was referring to EC on cards that support digital lines (T-1/E-1/PRI) |
| 14:07.01 | mamep | is it possible to make multiple calls from a sip providers ? |
| 14:07.04 | WindBack | ManxPower, yes, thank |
| 14:07.21 | ManxPower | WindBack: If the ENTIRE call patch is digital you cannot have echo (as echo is usually defined), but by "entire call path" I also mean the far end of the call. |
| 14:07.29 | ManxPower | gads, I can't type today. |
| 14:07.34 | ManxPower | call patch == call path |
| 14:07.51 | mamep | ? |
| 14:07.52 | mamep | ? |
| 14:08.01 | fiXXXerMet | Yikes, the Festival() application really eats up the cpu |
| 14:08.20 | fiXXXerMet | Asterisk is using 100% of my cpu now |
| 14:08.40 | tzanger | ManxPower: there's no such thing as an all-digital call, as it's got to hit ear+mouth eventually and you can get acoustic echo or echo through poor mechanical mounting of the two |
| 14:08.48 | *** join/#asterisk PepOSX (n=angeldav@190.79.246.105) |
| 14:08.52 | WindBack | ManxPower, yes, I understand you |
| 14:09.20 | ManxPower | tzanger: I know, but I do not call that "echo as it is usually defined), which is cause by the 2-wire hybrd. |
| 14:09.25 | tzanger | agreed |
| 14:09.37 | tzanger | I get too much echo bouncing off of other people's shitty-ass hybrids :-( |
| 14:09.47 | ManxPower | I call that "crappy phone, whos designers should be shot echo" |
| 14:14.11 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 14:14.33 | mamep | is it possible to make multiple calls from a sip providers ? |
| 14:14.53 | ManxPower | mamep: yes. |
| 14:15.07 | mosty | mamep, if the sip provider allows that, sure |
| 14:15.08 | mamep | ManxPower : how? |
| 14:15.37 | mosty | mamep, just dial twice to two different numbers using that sip peer |
| 14:15.42 | ManxPower | mamep: that is like asking how to make two cars use the same road. You just drive on it. |
| 14:15.55 | ManxPower | You just send calls to the provider and if the provider allows multiple calls the calls will work. |
| 14:16.07 | mamep | aha |
| 14:16.14 | mamep | so it'ws not working i've tried that |
| 14:16.15 | mamep | :( |
| 14:16.33 | codejunky | hi, I want to log the phone numbers of all incoming calls from sip. how could I accomblish this? |
| 14:16.37 | fiXXXerMet | Is Festival() supposed to sound so choppy? |
| 14:16.39 | mosty | mamep, how many simultaneous calls does your sip provider allow? have you asked them that? |
| 14:16.50 | ManxPower | codejunky: you look at /var/log/asterisk/cdr-csv |
| 14:17.23 | codejunky | ManxPower: nice! |
| 14:17.37 | ManxPower | codejunky: you would have known that if you had read the Asterisk Book |
| 14:17.39 | ManxPower | ~book |
| 14:17.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 14:17.54 | codejunky | ManxPower: thanks |
| 14:18.51 | *** join/#asterisk pepse (n=pepse@71-223-193-145.phnx.qwest.net) |
| 14:19.27 | mamep | mosty : no i think they allow 1 :( |
| 14:20.04 | mosty | mamep, then stop trying |
| 14:20.07 | BBHoss | can anyone suggest how i remedy this problem: http://pastebin.com/d112a9c4f |
| 14:20.15 | BBHoss | i have it set to GSM on both sides |
| 14:21.28 | ManxPower | BBHoss: pastebin the sip.conf parts |
| 14:21.41 | BBHoss | its iax |
| 14:21.53 | ManxPower | sorry, iax.conf |
| 14:23.24 | tzanger | whoa |
| 14:23.33 | BBHoss | http://pastebin.com/d3a67f5e |
| 14:23.39 | tzanger | I'm mentioned in TFOT, I didn't realize that |
| 14:23.45 | tzanger | haha my name's in print! |
| 14:24.27 | BBHoss | tzanger: now when skynet takes over they will know you |
| 14:24.32 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 14:24.34 | tzanger | BBHoss: that's okay, i'm writing skynet |
| 14:24.46 | BBHoss | sweet, put some cool easter eggs in there |
| 14:24.50 | tzanger | you bet |
| 14:25.14 | tzanger | if you grab one of the T-1 prototypes and wiggle their arm up, up, down, down, left, right, left, right and honk her nose, you'll be in ofr a surprise |
| 14:25.25 | BBHoss | heh |
| 14:25.36 | tzanger | there's some old Konami code in there :-) |
| 14:26.23 | BBHoss | awesome |
| 14:26.38 | Dr-Linux | anybody tried vicidial? |
| 14:27.06 | BBHoss | ManxPower: im thinking the ITSP is not wanting to be compatible with 1.4 |
| 14:27.09 | ManxPower | Dr-Linux: Yes, but we always kill them as a warning to other telemarketers |
| 14:27.32 | BBHoss | we'll put skynet on your ass! |
| 14:27.46 | Dr-Linux | hhm.. |
| 14:28.03 | *** join/#asterisk zeeesh (i=zeeesh@203.215.179.43) |
| 14:28.20 | Dr-Linux | ManxPower: my problem is connection between asterisk and VC database |
| 14:28.24 | ManxPower | We consider them to be not as bad as people killing babies, but much worse than people clubbing baby seals. |
| 14:28.44 | Dr-Linux | ManxPower: when i logged in to VC client on the web, i can't see anything on the CLI :S |
| 14:29.21 | ManxPower | Dr-Linux: I suspect you will need to find a channel specific to that software. Joking aside, so few people use that software here that it may be impossible to find help. |
| 14:29.45 | Dr-Linux | :) |
| 14:30.18 | *** join/#asterisk flujan (n=flujan@200.160.115.20) |
| 14:30.43 | BBHoss | any ideas ManxPower |
| 14:31.10 | ManxPower | BBHoss: none whatsoever. But if 1.4 didn't work with 1.2 the world would have ended, so I doubt your problem is that simple./ |
| 14:31.12 | errr | how do you read the help for modules from the asterisk cli? |
| 14:31.24 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:31.24 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:31.47 | ManxPower | errr: You can't. But you can for apps and functions. "show application appname" or "show function FUNCNAME" (caps are important) |
| 14:31.52 | BBHoss | ManxPower: its a bit odd that it recognizes GSM but then won't match up |
| 14:32.01 | BBHoss | like why would gsm on my side be 0xe000. |
| 14:32.06 | codejunky | are there any scripts out there to generate statistics from the master.csv file? |
| 14:32.13 | errr | ManxPower: ahh, thanks. I always get that syntax messed up |
| 14:32.14 | ManxPower | BBHoss: "show codecs" tells you the codec numbers |
| 14:32.41 | BBHoss | i have 0x2 of course |
| 14:32.51 | WindBack | Qwell, Yes you are right, the module is wctdm24xxp |
| 14:32.52 | BBHoss | but for some reason it won't work :( |
| 14:33.20 | ManxPower | find the option for video support in iax.conf and turn it off. look at iax.conf.sample Also, you have SIP specific options in your iax.conf (doesn't hurt, but are ignored) |
| 14:34.32 | BBHoss | heh evidentially i've had this problem before, i did a google search and this: http://fr.pastebin.ca/461580?srch=oseqno came up :) I can't remember what i did |
| 14:35.55 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:36.04 | ManxPower | IIRC high codec numbers mean video. |
| 14:36.24 | ManxPower | But I don't use video stuff. People would rather not see me. 9-_ |
| 14:36.39 | mosty | codejunky, grep? awk? perl? statistics of what? |
| 14:37.13 | codejunky | mosty: it was just a question whether something is around, otherwise I will code something :-) |
| 14:37.13 | fiXXXerMet | Can I call the Festival() application inside of the Background() application? |
| 14:37.18 | BBHoss | ManxPower: i cannot find any codecs that would be 0xe000 |
| 14:37.23 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 14:37.42 | tzafrir | codejunky, feed it to your near-by spreadsheat |
| 14:37.49 | codejunky | :-) |
| 14:37.58 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:41.01 | *** join/#asterisk Rudolf (n=omnioujy@189.35.25.241) |
| 14:41.11 | *** join/#asterisk oppassum (n=op@adsl-76-249-13-6.dsl.klmzmi.sbcglobal.net) |
| 14:41.24 | Rudolf | jow i can debug a high cpu usage of asterisk? |
| 14:41.44 | Rudolf | normally is transcoding, but i don't know how i can solve this issue |
| 14:41.58 | oppassum | anyone here know how to troubleshoot issues with dialong OUT to IVRs? |
| 14:42.13 | ManxPower | oppassum: You mean Asterisk steals the # |
| 14:45.16 | ManxPower | oppassum: it is considered very rude to ask a question and then ignore all responses. |
| 14:46.22 | lirakis | oppassum: or do your button presses seem to do nothing when you call an ivr? .. if this is the case, your phone is not configured to match your asterisk's dtmf setting. |
| 14:46.32 | drmessano | oppassums are blind |
| 14:46.35 | drmessano | Maybe he didnt see |
| 14:46.48 | lirakis | drmessano: really? .. i though they were just nasty |
| 14:46.52 | drmessano | lol |
| 14:47.09 | drmessano | Nasty? Ever heard of ketchup? |
| 14:47.41 | ManxPower | they are not blind |
| 14:47.42 | lirakis | i meant nasty as in .. ill tempered |
| 14:47.45 | drmessano | I know |
| 14:47.57 | drmessano | I was trying to help you improve the tatse |
| 14:47.59 | drmessano | taste |
| 14:48.23 | *** join/#asterisk SteveTotaro (n=root@pool-70-16-15-23.balt.east.verizon.net) |
| 14:48.28 | lirakis | and no they arent blind.. but their eyes dont adjust very quickly to light.. so they can be temporarily blinded by bright lights... which may lead to this misconception... |
| 14:48.35 | lirakis | okay.. no more about opposums ;) |
| 14:48.40 | ManxPower | and death |
| 14:48.48 | drmessano | I know they're not blind.. I was joking |
| 14:48.54 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:49.03 | ManxPower | As far as I can tell they get very depressed in the spring -- based on all the dead ones I see along the road. |
| 14:49.04 | lirakis | drmessano: .. its a literal crowd here he he |
| 14:49.04 | drmessano | Phew.. this channel need a serious load of app_humor |
| 14:49.53 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:51.04 | FlatFoot | anyone done an install for cdr_adaptive_odbc on freebsd ? i am tryning to make the Makefile run but it keeps moaning at me ( just like the wife ) |
| 14:53.19 | Rudolf | FlatFoot: heuheiuheiuehiue |
| 14:54.00 | FlatFoot | Rudolf: whats that when it's at home ? |
| 14:56.10 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:57.41 | Rudolf | FlatFoot: sorry, i cannot help you, i'm dummy in asterisk, but you has a funny phrase ("just like the wife") |
| 14:57.53 | *** join/#asterisk grayhame (n=grayhame@ns2.dalcon.com) |
| 14:58.02 | FlatFoot | Rudolf: aha yes english humour |
| 14:59.02 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 14:59.32 | WindBack | I'm compiling the zaptel drivers for an AEX808. I 'm using make menuselect. As Qwell tellme the module used by this board is wctdm24xxp. if I deselect all the others modules that apear in that list is ok?? |
| 14:59.33 | javar | hi |
| 14:59.59 | javar | someone know, how can i check if mpg123 is working with asterisk? |
| 15:00.23 | WindBack | or are there another important module that i have to keep |
| 15:00.27 | WindBack | ? |
| 15:00.40 | Qwell | WindBack: should be able to remove all of the modules besides that one |
| 15:01.24 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.227) |
| 15:01.37 | *** join/#asterisk javb (n=javb@190.80.194.189) |
| 15:01.44 | tmccrary | I have a Snom 320, is there any way to get voicemail working with asterisk? |
| 15:02.21 | mosty | just set your mailbox to the number you dial for voicemail, should work fine |
| 15:02.39 | WindBack | Qwell, thanks |
| 15:02.48 | javb | Hi, i have TE120P, trying to configure the same way i used to configure 110P, and now i get 'blue/yel/red' alarm, and when trying to start zaptel, i get "Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22)" any ideas? |
| 15:03.05 | tmccrary | I'm getting "Address incomplete" with the snom on any extension that goes to VoiceMailMain |
| 15:03.19 | tmccrary | but regular extensions work fine (eg I can call other phones) |
| 15:03.24 | tmccrary | and polycoms don't have a problem |
| 15:03.32 | mosty | tmccrary, what if you call the voicemail number manually on the snom? |
| 15:03.38 | tmccrary | same thing |
| 15:04.17 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583384.dsl.bell.ca) |
| 15:04.18 | mosty | pastebin that part of your dialplan |
| 15:04.27 | tmccrary | exten => 0,1,VoiceMailMain(${CALLERIDNUM}@default) |
| 15:05.03 | mosty | tmccrary, what i do is set a channel variable in sip.conf instead of relying on callerid, which the phone can change |
| 15:06.05 | tmccrary | Can you post an example or perhaps let me know what I should search for regarding that? |
| 15:07.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:07.57 | ManxPower | tmccrary: sounds to me like you have other extensions that start with 0 |
| 15:09.28 | nebojsajsimic | how to fix this execvp: ./configure: Permission denied |
| 15:09.35 | tmccrary | man, these snom phones are garbage |
| 15:10.09 | tmccrary | ManxPower: Nope, only one. And I just tried a crazy extension to make sure asterisk wasn't having issues reading a new config or something |
| 15:10.09 | SteveTotaro | ls -l conf* |
| 15:13.29 | nebojsajsimic | how to fix this execvp: ./configure: Permission denied any idea |
| 15:15.43 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 15:15.49 | *** join/#asterisk egypcio (n=vinicius@200.150.142.210) |
| 15:19.50 | Stefan1979 | nebojsajsimic: Maybe you should be root |
| 15:22.18 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 15:22.45 | nicox | Hi |
| 15:23.38 | nicox | Does anybody know how to make asterisk faster to resolve a incoming call on chan_zap in the dialplan |
| 15:23.40 | nicox | ? |
| 15:23.56 | nebojsajsimic | i am root |
| 15:24.28 | *** join/#asterisk CrashSys (n=kumba@62-209.187-72.tampabay.res.rr.com) |
| 15:24.40 | nicox | Does anybody know how to make asterisk faster to resolve a incoming call on chan_zap in the dialplan ? |
| 15:27.56 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 15:28.31 | *** join/#asterisk saftsack (n=saftsack@p4FC74881.dip.t-dialin.net) |
| 15:29.26 | drmessano | Root has a permission problem |
| 15:29.30 | drmessano | Hmmm |
| 15:29.35 | drmessano | root != root? |
| 15:29.56 | Stefan1979 | nebojsajsimic: Well... It sounds like a file-permission problem. Even root needs write permissions to files to be able to write to them. |
| 15:32.39 | Stefan1979 | nebojsajsimic: The permissions for the directory you're compiling from should at least be something along these lines: drwxr-xr-x |
| 15:33.50 | *** join/#asterisk Chris55 (n=chris@dhcp-0-30-48-92-0-88.cpe.quickclic.net) |
| 15:34.04 | *** join/#asterisk timeshell (n=Khoja@gw.lusi.on.ca) |
| 15:34.23 | timeshell | greetings all |
| 15:34.33 | Chris55 | Anyone know how to set Cisco IP (SIP) telephones to work outside of the local area asterisk server? |
| 15:34.37 | timeshell | I need a little help with 1.4.9 Queues and Agents |
| 15:34.55 | timeshell | Chris55: you need appropriate networking |
| 15:35.03 | timeshell | Open ports, natting possibly |
| 15:35.08 | CrashSys | Dont suppose anyone is familiar with sendmail? |
| 15:35.21 | Chris55 | its open just the external phone is trying to find something local |
| 15:35.25 | penguinFunk | why CrashSys, what you need? |
| 15:35.28 | Chris55 | and locally isnt local for it obviously. heh |
| 15:35.38 | CrashSys | trying to re-route e-mail stuck in a queue |
| 15:36.07 | timeshell | Chris55: What is the phone trying to find that is "local"? |
| 15:36.34 | timeshell | Chris55: It really shouldn't matter at all whether it is local or not. Same principles that Vonage uses |
| 15:36.53 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:37.06 | timeshell | If your server doesn't have a public IP address or your phone is behind a firewall, you may need to enable NAT |
| 15:37.16 | Chris55 | timeshell: trying to find the TFTP server to load the config but the Ip its searching for is local and I am unable to change it to an external IP via the phone |
| 15:37.23 | timeshell | If the server is on a private IP, then you'll have to do some NATing to it |
| 15:37.35 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:37.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:37.52 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 15:37.53 | timeshell | Chris55: Ah, I see. I can't help you there. I am not familiar with that phone. |
| 15:37.54 | Chris55 | timeshell: server has an external IP and can be accessed externally but the primary two phones are local and one or two will be external |
| 15:38.02 | Chris55 | oh ok |
| 15:38.03 | drmessano | ~sipnat |
| 15:38.03 | 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 |
| 15:38.11 | Chris55 | thanks but that isnt it. |
| 15:38.18 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 15:38.26 | Chris55 | its this cisco phones that are a pain should buy something else |
| 15:39.37 | timeshell | Ok, so I'm trying to set up an Agent and it appears I got it to log in to a queue yesterday. But when I call the Agent Login extension again, I get prompted for a login, but it doesn't appear to get what I enter. It seems to time out and just says goodbye |
| 15:39.44 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 15:39.47 | timeshell | Is this because it is already logged in? |
| 15:40.07 | timeshell | I have tried to "Log out" but it doesn't seem to let me do this either. |
| 15:41.28 | *** join/#asterisk mcinerney (n=john@peter.mcinerney.id.au) |
| 15:43.23 | FlatFoot | Corydon72-dig: are you available for a question ? |
| 15:43.47 | FlatFoot | Corydon76-dig: are you available for a question ? sorry wrong number earlier |
| 15:44.21 | Corydon76-dig | Ask in channel |
| 15:44.24 | timeshell | Should one be using Ring Groups instead of Agent settings? |
| 15:44.34 | timeshell | I can get the Ring Group to work. |
| 15:44.47 | timeshell | But, I'd prefer to use agents. |
| 15:45.17 | FlatFoot | i am trying to install cdr_adaptive_odbc on version 1.4.17 ( freebsd ) but it keeps moaning at me is the latest Makefile etc for that version or not ? |
| 15:45.52 | Corydon76-dig | Are you using svncommunity? |
| 15:46.14 | aikanaro79 | I'm developing an application to interact with asterisk using SIP only...I'm able to register with success but when I try to dial someone asterisk replies that it cannot authenticate my user...ideas? if I try with another client program it works fine |
| 15:46.19 | FlatFoot | yes thats where i got the files from |
| 15:46.21 | FlatFoot | of the web |
| 15:46.57 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 15:46.58 | Corydon76-dig | FlatFoot: do a direct checkout, set the environmental variable 'ASTSRC' to the location of your main Asterisk 1.4 compile, and just type 'make' in that directory |
| 15:48.05 | FlatFoot | so checkout to the modules folder ? |
| 15:48.21 | Corydon76-dig | What? No |
| 15:48.41 | FlatFoot | sorry not sure about the terms just learning all this unix stuuf |
| 15:48.45 | Corydon76-dig | svn co http://svncommunity.digium.com/svn/tilghman/branches/1.4 svncommunity-1.4 |
| 15:48.46 | *** part/#asterisk Stefan1979 (n=stan@4806ds1-hl.0.fullrate.dk) |
| 15:48.53 | Corydon76-dig | cd svncommunity-1.4 |
| 15:49.04 | Corydon76-dig | export ASTSRC=/home/flatfoot/asterisk-1.4 |
| 15:49.06 | Corydon76-dig | make |
| 15:49.28 | FlatFoot | aha that make sense thank-you very much i shall get to work |
| 15:49.39 | timeshell | Does anyone know if the new 1.6 branch is supported by chanskype? |
| 15:51.51 | Corydon76-dig | 1.6 is not a branch |
| 15:52.08 | timeshell | Version then |
| 15:52.40 | Corydon76-dig | Ask them if they have a version for trunk. If so, then it'll work for the 1.6 release cycle |
| 15:53.18 | drmessano | ewww |
| 15:53.23 | drmessano | Dont say Skype in here |
| 15:53.25 | BipBip | Hello all, ${SIPDOMAIN} is not giving me the domain but the IP, it there other this one? |
| 15:53.26 | timeshell | Ok. I'll do that. Can you offer any assistance on the Agent's / Call Queues? |
| 15:53.39 | Corydon76-dig | ~ask |
| 15:53.40 | 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. |
| 15:54.03 | timeshell | drmessano: Unfortunately, S**** is the most economical for me for North American LD right now |
| 15:56.03 | javb | Hi, i have TE120P, trying to configure the same way i used to configure 110P, and now i get 'blue/yel/red' alarm, and when trying to start zaptel, i get "Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22)" any ideas? |
| 15:57.16 | SteveTotaro | timeshell, check this out |
| 15:58.04 | SteveTotaro | http://voipcalls4free.blogspot.com/ |
| 15:58.33 | timeshell | K, how about Canada? |
| 15:59.30 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 16:00.11 | SteveTotaro | timeshell: take some time and read what i gave you |
| 16:00.15 | timeshell | lol |
| 16:00.16 | timeshell | I am |
| 16:00.25 | timeshell | But anyway, that's not why I came here today |
| 16:00.35 | timeshell | I need help with agents logging in and out |
| 16:00.57 | SteveTotaro | sorry, you used up your free wish for the day |
| 16:01.00 | *** join/#asterisk mintee (n=mintee@72-165-177-90.dia.static.qwest.net) |
| 16:01.11 | mintee | how do i kill a locked channel from the console? |
| 16:01.19 | timeshell | Err...I never wished for what I was given. :p |
| 16:01.26 | aikanaro79 | please does anyone have a clue on what the problem might be? |
| 16:01.35 | SteveTotaro | softhangup |
| 16:02.17 | mintee | SteveTotaro, not working |
| 16:02.26 | SteveTotaro | stop now |
| 16:02.31 | SteveTotaro | lol |
| 16:02.35 | SteveTotaro | that should do it |
| 16:02.43 | mintee | :( |
| 16:02.44 | timeshell | :| |
| 16:02.45 | mintee | lol |
| 16:04.37 | SteveTotaro | firstnsrvr01*CLI> soft hangup |
| 16:04.38 | SteveTotaro | Usage: soft hangup <channel> |
| 16:04.38 | SteveTotaro | <PROTECTED> |
| 16:04.38 | SteveTotaro | <PROTECTED> |
| 16:05.48 | mintee | yeah, that didn't work |
| 16:05.59 | mintee | it was stuck in a Festival() call |
| 16:06.00 | SteveTotaro | what is the channel? |
| 16:06.05 | mintee | Zap1/1 |
| 16:06.37 | SteveTotaro | is it an analog card? |
| 16:06.46 | mintee | sangoma PRI |
| 16:07.03 | SteveTotaro | if you do a show channels what is the channel name? |
| 16:07.40 | SteveTotaro | i wonder if it will reset, providing you did not turn off resetinterval |
| 16:07.42 | mintee | one sec |
| 16:08.10 | mintee | Channel Location State Application(Data) |
| 16:08.10 | mintee | Zap/1-1 s@custom-recording:1 Ring Festival(Please record your me |
| 16:08.40 | mintee | Jan 23 11:08:31 DEBUG[4221]: chan_zap.c:8560 pri_dchannel: Ring requested on channel 0/1 already in use or previously requested on span 1. Attempting to renegotiating channel. |
| 16:08.56 | SteveTotaro | so what command did you issue for soft hangup? |
| 16:09.06 | mintee | soft hangup Zap/1-1 |
| 16:09.25 | mintee | i'm not sure why it's not skipping to the next channel either.. since i have 23 |
| 16:09.40 | SteveTotaro | that was my next question |
| 16:10.02 | SteveTotaro | i suspect you have other live calls going on? |
| 16:10.19 | mintee | my telco may have screwed it up and pointed all the PRI numbers to a single DID... |
| 16:10.35 | mintee | SteveTotaro, no, just a test box. I'm the only one with the number |
| 16:11.00 | SteveTotaro | oh, then i guess it is not that big of a deal |
| 16:11.09 | SteveTotaro | not sure why soft hangup is not working |
| 16:11.14 | mintee | well it will be when i make the box go live |
| 16:11.27 | SteveTotaro | try setting the resetinterval to a low number |
| 16:11.31 | *** join/#asterisk Titanous (n=Jon335@unaffiliated/titanous) |
| 16:11.36 | mintee | *CLI> SIOD ERROR: unbound variable : tts_textasterisk |
| 16:11.49 | SteveTotaro | and see if you can recreate it |
| 16:11.59 | SteveTotaro | and if the reset clears it |
| 16:12.06 | Titanous | I just bought a G.729 licenses, and went to register it, but got 'Could not generate Host-ID' |
| 16:12.18 | Titanous | solutions? |
| 16:12.25 | SteveTotaro | contact digium |
| 16:12.30 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:12.39 | Titanous | I submitted a ticket two days ago, and no response |
| 16:12.49 | SteveTotaro | call them |
| 16:12.52 | mintee | SteveTotaro, not sure where the resetinterval is.. |
| 16:13.25 | timeshell | Agent logoff #### is not logging of my agent from CLI either |
| 16:13.26 | timeshell | :( |
| 16:14.13 | SteveTotaro | it is in /etc/asterisk/zapata.conf (it is a default so you may have to add resetinterval=60) |
| 16:14.46 | SteveTotaro | hopefully that would clear any stuck channels at a maximum of 60 seconds |
| 16:14.48 | mintee | damn, i just did a restart gracefully and it's looping now. can't release the channel |
| 16:15.00 | timeshell | Was there bugs in the agent / queue stuff in 1.4.9? |
| 16:15.48 | SteveTotaro | wouldn't know, i use 1.2 for call centers |
| 16:15.57 | SteveTotaro | it just works |
| 16:16.12 | timeshell | I'm trying to set up a new AsteriskNOW |
| 16:16.27 | timeshell | It's to demo a new system |
| 16:16.29 | SteveTotaro | it is still beta right? |
| 16:16.37 | timeshell | I thought it may be quicker, but so far it's been a pain in the rear. |
| 16:17.05 | tzanger | heh |
| 16:17.07 | timeshell | Yah, but it has updates |
| 16:17.08 | JT | hrm |
| 16:17.11 | tzanger | yeah that's been my experience with all guis |
| 16:17.12 | SteveTotaro | switchvox is an awesome product although it is not free |
| 16:17.17 | timeshell | Beta 6 was 1.4.2 I think |
| 16:17.22 | timeshell | the update updated to 1.4.9 |
| 16:17.24 | tzanger | what I really, relaly, really like about *now is how it will read in a conf file, change it and write it back out, comments and all |
| 16:17.28 | timeshell | Which fixed a couple things |
| 16:17.31 | JT | giving this aastra a dialplan of just + seems to make it get stuck booting at SIP 92% |
| 16:17.58 | timeshell | AsteriskNOW should also give you the ability to manually update it |
| 16:18.03 | timeshell | (or does it?) |
| 16:18.04 | timeshell | lol |
| 16:18.12 | tzanger | yeah I want to take a good look at switchvox |
| 16:18.25 | tzanger | ok |
| 16:18.34 | SteveTotaro | i installed a few switchvox systems over a year ago |
| 16:18.58 | SteveTotaro | best in breed by far, i was not surprised in the least that digium bought them |
| 16:19.45 | SteveTotaro | the biggest non-free piece of junk i had the displeasure of dealing with was signate on an SGI Altix 350 |
| 16:20.37 | timeshell | ok. so can anyone please suggest why I can't log out an agent? |
| 16:20.42 | timeshell | Either from the phone or CLI? |
| 16:21.00 | SteveTotaro | maybe an error message would help.... |
| 16:21.04 | SteveTotaro | ~pb |
| 16:21.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:21.34 | timeshell | no error |
| 16:21.44 | SteveTotaro | nothing on the console? |
| 16:21.48 | timeshell | nope |
| 16:22.26 | timeshell | One of the reasons I'm pulling my hair out. It's not saying anything |
| 16:22.28 | SteveTotaro | no idea then |
| 16:22.58 | SteveTotaro | it is beta, not sure what you expect |
| 16:23.05 | SteveTotaro | i would ditch it for now |
| 16:23.08 | timeshell | Well, the *NOW is beta |
| 16:23.19 | timeshell | CLI in 1.4.9 should do something |
| 16:23.30 | timeshell | Bah |
| 16:24.04 | SteveTotaro | go with freepbx |
| 16:24.10 | JT | ... |
| 16:24.14 | JT | wrong channel |
| 16:24.49 | timeshell | Ok, building new system with CentOS, newest * and SVN *NOW then |
| 16:24.51 | timeshell | :( |
| 16:25.26 | *** part/#asterisk Titanous (n=Jon335@unaffiliated/titanous) |
| 16:25.26 | timeshell | was really hoping to save some time |
| 16:25.40 | SteveTotaro | catch 22 |
| 16:25.47 | SteveTotaro | thirdlane makes a good product |
| 16:25.52 | SteveTotaro | at a good price |
| 16:26.03 | drmessano | Thats what I want.. a PBX someone rushed together |
| 16:26.16 | SteveTotaro | with beta software |
| 16:26.40 | drmessano | "I built this thing in 11 minutes.. Hope it works!" |
| 16:27.01 | timeshell | lol, drmessano, you really shouldn't speak without knowing the circumstance |
| 16:27.08 | SteveTotaro | i still don't see a business case for digium to continue *now, i see quite the opposite |
| 16:27.16 | timeshell | The *NOW was supposed to be a proof of concept...not the final. |
| 16:27.40 | SteveTotaro | make the magic happen in .conf files |
| 16:27.40 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 16:27.41 | timeshell | I don't have a heck of a lot of time to get it ready |
| 16:27.56 | SteveTotaro | great demo when things don't work |
| 16:27.59 | timeshell | lol |
| 16:28.06 | drmessano | timeshell, you're still rushing, I dont care about circumstances |
| 16:28.10 | timeshell | The Agent/Queue thing was the last thing I needed to get working |
| 16:28.14 | SteveTotaro | you should postpone it |
| 16:28.32 | timeshell | Not my call |
| 16:28.46 | SteveTotaro | get thirdlane or something |
| 16:28.49 | timeshell | Could lose out altogether. |
| 16:29.13 | timeshell | no biggie |
| 16:29.14 | drmessano | Doesnt sound to me like you would be able to deploy properly if you won |
| 16:29.27 | [TK]D-Fender | timeshell: You should be showing use your queue status, AstDB dump, logoff attempt, agents & queue configs, and CLI output of the attempt to log off. You have so shown very little. |
| 16:29.28 | timeshell | Whatever dude |
| 16:29.34 | SteveTotaro | if you demoed something to me and was buggy, i would quickly show you the door |
| 16:29.57 | mintee | ok, anyone in here using festival? |
| 16:30.04 | drmessano | If they gave me three days to build a rocket for a NASA contract, and somehow fucked up and made it work, am I the guy that should be building it? |
| 16:30.05 | SteveTotaro | nope |
| 16:30.05 | drmessano | Hell no |
| 16:30.12 | mintee | lol... |
| 16:30.19 | timeshell | Hrm |
| 16:30.22 | mintee | ok, is there another choice for text to voice? |
| 16:30.31 | SteveTotaro | yes |
| 16:30.50 | SteveTotaro | cepstral is very good |
| 16:31.01 | timeshell | Nothing in the queue |
| 16:31.16 | mintee | ah, commercial... |
| 16:31.16 | [TK]D-Fender | timeshell: PASTEBIN everything. |
| 16:31.26 | mintee | f it, i don't really need it anyway, just wanted to play around |
| 16:31.44 | SteveTotaro | you can do cepstral for free |
| 16:31.56 | mintee | with the demo? |
| 16:32.03 | mintee | fairly easy to setup? |
| 16:32.04 | SteveTotaro | there are work arounds and demo |
| 16:32.11 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:32.21 | dandre | hello, |
| 16:32.31 | SteveTotaro | the demo you just have to listen to some crap about it being a demo everytime it starts |
| 16:32.33 | mocker | I'm demoing the installation of asterisk at my next AUG, anyone have any cool ideas for getting it connected to something free after the install? |
| 16:32.50 | drmessano | FWD |
| 16:32.54 | mocker | I guess I can just setup a SIP phone and call Digium's IAX line. |
| 16:33.05 | SteveTotaro | http://voipcalls4free.blogspot.com/ |
| 16:33.17 | mocker | drmessano: That's a good idea. |
| 16:34.04 | drmessano | That site is lame |
| 16:34.16 | drmessano | How to use Gizmo with an ATA |
| 16:34.18 | drmessano | Yawn |
| 16:34.21 | [TK]D-Fender | mocker: IPKall |
| 16:34.41 | SteveTotaro | how to get free inbound and outbound calls |
| 16:34.49 | SteveTotaro | guess you cant read so good |
| 16:34.49 | dandre | I am trying to use Background() for creating a ivr. I don't see the keys hit (they don't do anything). |
| 16:35.21 | SteveTotaro | you need to go to the school for kids that can't read so good |
| 16:35.23 | drmessano | I guess you cant |
| 16:35.33 | [TK]D-Fender | dandre: check your DTMF mode, pastebin your dialplan and its CLI output at verbose 10, debug 10 |
| 16:35.39 | drmessano | Its a howto on gizmo with a sunrocket ATA and Grandcentral for incoming |
| 16:35.47 | mocker | Guh, first I need to figure out VirtualBox networking so I can connect from my local machine. |
| 16:35.49 | drmessano | Maybe you need to reread |
| 16:37.06 | SteveTotaro | click2call buddy |
| 16:37.40 | SteveTotaro | the school for kidz thats cants reads tso goo |
| 16:37.49 | drmessano | Yeah, Gizmo to Gizmo |
| 16:37.53 | javb | I`m getting "No functioning zap hardware found in /proc/zaptel, loading ztdummy" .. any ideas? |
| 16:38.01 | drmessano | You dont understand the material, so FAIL |
| 16:38.44 | drmessano | If all your friends have Grandcentral and Gizmo, you can use the click to call on Grandcentral to ring each others gizmo numbers |
| 16:39.01 | [TK]D-Fender | SteveTotaro: , drmessano : COOL IT. |
| 16:39.02 | drmessano | Its still Gizmo |
| 16:39.05 | drmessano | Read again |
| 16:39.41 | dandre | [TK]D-Fender: it's probably a dtmf problem because I don't see anthing with verbose and debug set to 10 |
| 16:40.05 | drmessano | [TK]D-Fender: Its one thing to have a convo, its another to be an insulting dick like SteveTotaro.. I wont stand for it |
| 16:40.26 | SteveTotaro | lmao |
| 16:40.46 | SteveTotaro | who is using profanity and calling things lame |
| 16:40.52 | timeshell | Ok, Queue status shows 2 calls in queue (I've only called with one phone) and the Agent that is logged in is showing Busy. However, the Agent's phone did not ring and is not actually in use. |
| 16:41.03 | drmessano | The post is lame.. is it YOURS? Did I insult YOU? No |
| 16:41.08 | SteveTotaro | you are probably sitting anyways, so don't stand for it |
| 16:41.20 | drmessano | So grow up dude |
| 16:41.26 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:41.31 | SteveTotaro | yes, you called me a bad word |
| 16:41.36 | SteveTotaro | so you grow up |
| 16:41.41 | drmessano | lol |
| 16:41.52 | SteveTotaro | learn to be professional |
| 16:42.05 | timeshell | what the heck? |
| 16:42.10 | drmessano | [11:32] <SteveTotaro> guess you cant read so good [11:32] <SteveTotaro> you need to go to the school for kids that can't read so good |
| 16:42.17 | drmessano | Thats what I read |
| 16:42.25 | drmessano | So grow ip |
| 16:42.27 | drmessano | up* |
| 16:42.35 | SteveTotaro | i have lots of ip |
| 16:42.39 | dandre | How should I set the dtmfmode on sip trunks? |
| 16:42.50 | SteveTotaro | rfc2833 |
| 16:43.02 | drmessano | If you can't understand the materials, dont insult those that do.. Thats my advice to you |
| 16:43.03 | SteveTotaro | ~trunk |
| 16:43.03 | jbot | trunk is, like, is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 16:43.18 | *** join/#asterisk af_ (n=getsmart@88-149-230-64.dynamic.ngi.it) |
| 16:43.18 | SteveTotaro | i understand just fine |
| 16:43.37 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 16:44.00 | timeshell | It seems as though when a call comes in, both the agent and the caller are appearing in the queue? It's as if the queue is adding the agent, not calling the agent. |
| 16:44.26 | [TK]D-Fender | SteveTotaro , drmessano : Enough already |
| 16:44.38 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:46.29 | *** join/#asterisk nDuff (n=ccd@user-387ocuv.cable.mindspring.com) |
| 16:48.11 | SteveTotaro | Grand Central lets you forward to your Gizmo Project SIP number = free inbound calls. To make free outbound calls, use click2call through Grand Central. Basically you will have everything Sunrocket gave you for FREE!!! |
| 16:48.16 | dandre | dtmfmode is already set to rfc2833 and I don't see key hit in background |
| 16:48.33 | nDuff | Grand Central supports SIP now? Sweet! |
| 16:48.46 | drmessano | Free Outbound to another users Gizmo number |
| 16:49.05 | drmessano | I set up Gizmo, someone else using the same setup clicks my "Click2Call" and it connects me over Gizmo |
| 16:49.11 | drmessano | you still dont get it |
| 16:49.14 | FlatFoot | Corydon76-dig: did all as you suggested but it still shows the same errors. I take it 1.4.17 will not use extra fields in a db on its own ? |
| 16:49.31 | SteveTotaro | i guess you are missing the grandcentral part |
| 16:49.35 | drmessano | Im not |
| 16:49.59 | SteveTotaro | ok guy |
| 16:50.17 | drmessano | Youre missing the whole thing.. Its a little hack for click2call from Gizmo to Gizmo |
| 16:50.43 | SteveTotaro | no you are missing the fact that grandcentral gives you a real phone number |
| 16:50.45 | SteveTotaro | lmao |
| 16:51.00 | drmessano | Yes, INBOUND |
| 16:51.07 | drmessano | Outbound they dont give me SHIT |
| 16:51.32 | drmessano | The click to call connects the calls via GIZMO |
| 16:51.46 | SteveTotaro | ok but if you use click to dial IN Grandcentral |
| 16:51.50 | drmessano | Otherwise it would be ZOMG Grandcentral is a completely free ITSP |
| 16:51.54 | SteveTotaro | then both legs get connected |
| 16:52.03 | drmessano | Click2Call has to be connected via some transport |
| 16:52.09 | SteveTotaro | ZOMG it is |
| 16:52.11 | drmessano | Grandcentral is NOT the transport |
| 16:52.23 | *** join/#asterisk sacitec (n=tobi@189.129.225.150) |
| 16:52.25 | timeshell | omg you guys, start a new channel |
| 16:52.28 | drmessano | Your PSTN number that GC forwards to, or in this case, GIZMO is |
| 16:52.36 | drmessano | forget it |
| 16:52.39 | timeshell | This has nothing to do with * |
| 16:52.49 | [TK]D-Fender | dandre: First "SIP Trunk" is not an appropriate term. Second the mode you should be using depends on your PROVIDER. Third you have not provided anything I have resquested and are wasting time |
| 16:53.18 | *** join/#asterisk jmls_work (n=jmls@81.138.42.77) |
| 16:53.29 | jmls_work | is the asterisk manager robust enough to handle 100+ connections, or should we use astmanproxy ? |
| 16:54.01 | [TK]D-Fender | jmls_work: best to proxy |
| 16:54.08 | Qwell | jmls_work: personally, I would use a proxy, just to keep it from flooding unnecessarily |
| 16:54.27 | Qwell | a proxy can filter out what you don't want, so the traffic goes down that much more |
| 16:54.31 | jmls_work | yeah, my problem is that my astmanproxy dies every two or three days |
| 16:54.46 | [TK]D-Fender | jmls_work: Then restart it daily. |
| 16:54.49 | jmls_work | heh |
| 16:55.07 | drmessano | Cron makes any app "stable" |
| 16:55.10 | drmessano | lol |
| 16:55.14 | jmls_work | I don't think it's time related : Something is causing it to crash |
| 16:55.26 | Qwell | restarting it hourly would increase performance by 24x |
| 16:55.32 | drmessano | ROFL |
| 16:55.54 | SteveTotaro | there are patches for astmanproxy that are supposed to address stability |
| 16:56.09 | SteveTotaro | http://www.fatwallet.com/forums/messageview.php?catid=18&threadid=758191&start=20 |
| 16:57.03 | jmls_work | SteveTotaro: "Sorry, no results were found for astmanproxy" |
| 16:57.18 | SteveTotaro | lol, two different conversations |
| 16:57.33 | jmls_work | hah |
| 16:57.35 | SteveTotaro | let me check the mailing list, that is where i read about patches for astman |
| 16:57.39 | SteveTotaro | one sec |
| 16:58.02 | jmls_work | thanks |
| 16:58.43 | tzafrir | SteveTotaro, it was in reply to a post by jmls |
| 16:59.00 | tzafrir | If you refer to the recent mail on asterisk-users |
| 16:59.15 | SteveTotaro | what is your email, the patch was attached to the mail |
| 16:59.20 | jmls_work | tzafrir_home: those patches seem to be for an earlier astmanproxy |
| 16:59.33 | SteveTotaro | or can i send a file through here |
| 16:59.45 | jmls_work | I've got 1.22pre |
| 17:00.24 | jmls_work | juliet mark lima sierra at dotr.com |
| 17:00.31 | jmls_work | thanks SteveTotaro |
| 17:01.24 | SteveTotaro | well the post was by you |
| 17:01.45 | SteveTotaro | and steve davies replied with a patch |
| 17:01.56 | SteveTotaro | but like you said, it was for 1.21 |
| 17:02.13 | jmls_work | yeah. I was hoping that there were some more patches around |
| 17:02.15 | SteveTotaro | maybe you can gain something in the diff |
| 17:02.19 | jmls_work | thanks anyway |
| 17:02.46 | jmls_work | I did look, but there didn't seem to be anything that was relevant |
| 17:04.53 | SteveTotaro | how about something like http://www.voip-info.org/wiki/view/MANAGERAPI-JAVA-AGI |
| 17:05.26 | SteveTotaro | what are you banging on the AMI so hard for? |
| 17:06.07 | jmls_work | I use the AMI for clients to originate calls from our application |
| 17:06.10 | *** join/#asterisk umay (n=chris@71-208-162-10.hlrn.qwest.net) |
| 17:06.14 | jmls_work | I don't want to use .call files |
| 17:06.20 | x86 | anyone know why an IAX2 peer would be showing UNREACHABLE when I can ping the other host just fine, and I know the authentication credentials are good |
| 17:06.47 | jmls_work | it's only one-way (I don't need data from AMI, only to send commands) |
| 17:06.54 | SteveTotaro | i had the same problem but used ftp as a workaround, ftp a call file from the CRM |
| 17:07.04 | SteveTotaro | it worked perfectly |
| 17:07.21 | jmls_work | hmm |
| 17:07.48 | SteveTotaro | took me a few days to come up with that one on a very good salary |
| 17:07.52 | jmls_work | I can be originating up to 20 calls per second, depending on the number of available agents |
| 17:08.05 | jmls_work | manual and auto-dial |
| 17:08.08 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 17:08.14 | SteveTotaro | that was not an issue for us |
| 17:08.31 | SteveTotaro | 96 agents all logging in at once |
| 17:08.58 | SteveTotaro | it may take a couple seconds for the phone to ring |
| 17:09.43 | SteveTotaro | we used it when the agents logged into the CRM to call their headsets |
| 17:10.14 | ZPertee | I am confused about NAT. If all of my devices are on the same local subnet and no outgoing/incoming traffic is received for it from the internet just local. However I know that my router uses NAT. Can someone please clarify the issue for me? |
| 17:10.58 | SteveTotaro | if your voice traffic is not going over your router then you don't need to worry about nat |
| 17:11.02 | jmls_work | SteveTotaro: thanks for the food for thought |
| 17:11.14 | SteveTotaro | jmls, let me know if that works for you |
| 17:11.32 | SteveTotaro | the manager killed me a few times so i figured out how to avoid it as much as possible |
| 17:11.52 | ZPertee | SteveTotaro ok thanks for the clarification |
| 17:11.58 | [TK]D-Fender | ZPertee: Read up : |
| 17:11.59 | [TK]D-Fender | ~sipnat |
| 17:12.00 | 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 |
| 17:12.16 | SteveTotaro | no, i get great satisfaction from hearing that something i suggested helped |
| 17:12.53 | sacitec | hello |
| 17:13.52 | [TK]D-Fender | ZPertee: But indeed if you have no need of the internet for your system you don't need to do any of this. |
| 17:14.13 | [TK]D-Fender | ZPertee: But its always good to be aware in case you do end up wanting to do that. |
| 17:15.07 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:17.09 | ZPertee | [TK]D-Fender ok thanks for your help and the ongoing dedication to support for this irc channel. It is greatly appreciated! |
| 17:18.22 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 17:21.48 | dandre | [TK]D-Fender: sorry I was busy at phone |
| 17:22.47 | dandre | If I dial from a zap channel, the keys are correctly recognized |
| 17:23.28 | dandre | so this must be a misconfiguration of my sip channel against my sip provider, I will try in this direction |
| 17:30.29 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:32.05 | *** join/#asterisk bmg505 (n=leon@196.209.181.112) |
| 17:35.20 | timeshell | Ok |
| 17:35.23 | timeshell | New info |
| 17:35.41 | timeshell | I deleted the "logged in" agent and was able to remove it that way |
| 17:35.49 | timeshell | After that, it allowed me to log in again |
| 17:35.51 | *** join/#asterisk teknoprep (n=chrisraw@unaffiliated/teknoprep) |
| 17:35.57 | timeshell | However, it only allows 1 agent to log in |
| 17:36.12 | timeshell | When I attempt to log in any other agent, it doesn't acknowledge the log in attempt |
| 17:36.24 | timeshell | It seems to just time out instead of accepting the entered extension |
| 17:36.31 | timeshell | and says good by |
| 17:36.55 | timeshell | Does not allow me to "log out" the agent |
| 17:37.47 | timeshell | Can an Agent log in from any soft phone? Or is special features required on a softphone for an agent? |
| 17:38.37 | SteveTotaro | you need to post your configs to pastebin |
| 17:39.02 | SteveTotaro | have you ever setup queues/agents in asterisk? |
| 17:39.11 | timeshell | No |
| 17:39.15 | timeshell | This is the first time |
| 17:39.29 | timeshell | I actually just got a call to work from the queue |
| 17:39.35 | timeshell | But still can't log out the agent |
| 17:40.08 | SteveTotaro | well without seeing your configs it is impossible for anybody to really help |
| 17:40.44 | timeshell | Ok, tell me about pastebin. I haven't done that before either |
| 17:40.53 | SteveTotaro | ~pb |
| 17:40.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:41.24 | timeshell | Which configs do you want? |
| 17:41.50 | SteveTotaro | not sure about *now |
| 17:42.09 | timeshell | Agents.conf only has persistentagents=yes |
| 17:42.27 | SteveTotaro | that sounds troublesome |
| 17:42.37 | timeshell | queues.conf has 3 static members |
| 17:42.50 | timeshell | strategy ringall |
| 17:43.42 | timeshell | What else would you like? |
| 17:43.47 | HeXeD | http://thevoipforum.org/forum_index/ |
| 17:45.24 | SteveTotaro | HeXed, bookmarked will check it when there are more postss |
| 17:45.41 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 17:45.53 | HeXeD | hehe catch22 cant get more posts untill more ppl join |
| 17:46.09 | HeXeD | how do you get a forum going that is the question lol? |
| 17:46.21 | SteveTotaro | third time today catch22 has been referenced in my world |
| 17:46.27 | HeXeD | lol |
| 17:46.33 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:46.42 | SteveTotaro | i used to have a very popular forum |
| 17:47.04 | HeXeD | cool |
| 17:47.17 | SteveTotaro | don't moderate too hard |
| 17:47.29 | *** join/#asterisk r0d3nt (i=nobody@pinky.ratman.org) |
| 17:47.36 | SteveTotaro | promote freedom of speech |
| 17:47.44 | HeXeD | well im hoping its not the type of place where moderating will be needed that much |
| 17:47.45 | *** join/#asterisk j__1 (n=j__1@c-98-195-149-98.hsd1.tx.comcast.net) |
| 17:47.55 | HeXeD | just a good community resource is my aim |
| 17:47.57 | timeshell | Ok, from what I've read, the way to log out is basically the same extension as logging in but entering # instead of an extension. Yes? |
| 17:48.02 | timeshell | Or is there another way to log out? |
| 17:48.42 | SteveTotaro | is there a manual for *now? |
| 17:48.47 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 17:48.52 | timeshell | dunno |
| 17:48.55 | timeshell | haven't found one |
| 17:50.31 | *** join/#asterisk smark (n=mdierolf@208-106-111-103.dsl.static.sonic.net) |
| 17:50.50 | smark | does anyone know how to clear the dial tone for fxotune other than using a single digit? On ATT in the bay area, dialing a single digit gives appx 10 sec and then repeatedly plays a message |
| 17:52.04 | *** join/#asterisk worgil (i=usta@88.231.40.16) |
| 17:54.42 | [TK]D-Fender | timeshell: APSTEBIN your configs. Less talk, more show. |
| 17:54.47 | [TK]D-Fender | ~pb |
| 17:54.48 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:55.44 | [TK]D-Fender | SteveTotaro>don't moderate too hard <- I'm a hard-line moderate, and a moderate extremist :) |
| 17:55.58 | *** join/#asterisk flujan (n=flujan@201-42-104-17.dsl.telesp.net.br) |
| 17:57.38 | worgil | ~book |
| 17:57.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:58.35 | worgil | how can i set gsm and ilbc codec ? |
| 17:58.57 | mvanbaak | disallow = all |
| 17:59.01 | mvanbaak | allow = gsm |
| 17:59.15 | mvanbaak | allow = ilbc |
| 17:59.35 | mvanbaak | latero all |
| 17:59.41 | timeshell | TKD: Which do you wish to see? I'll be pulling bits and pieces out of the *NOW File Editor unless you can tell me how to get the files directly from *NOW. |
| 18:00.40 | worgil | mvanbaak but typing gsm:20 and ilbc:20 how can i use low bandwidth ? |
| 18:01.05 | worgil | <PROTECTED> |
| 18:01.07 | [TK]D-Fender | timeshell: users.conf, extensions.conf, queues.conf, agents.conf |
| 18:01.11 | worgil | Codec Order : (ulaw:20,gsm:20,ilbc:30) |
| 18:01.37 | [TK]D-Fender | timeshell: log into your sever with an SSH client and copy/paste from the terminal |
| 18:01.47 | *** join/#asterisk UnixDog (n=unixdog@adsl-69-234-203-31.dsl.irvnca.pacbell.net) |
| 18:02.44 | [TK]D-Fender | worgil: Do as mvanbaak showed you and you will not get stuck on ULAW. Also you should set ONE codec per device. |
| 18:02.59 | flujan | hi all... |
| 18:03.53 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 18:04.52 | worgil | [TK]D-Fender when i use one per device using 22 Kb, how can i set it (gsm=13 Kb) bandwidth ? |
| 18:05.20 | [TK]D-Fender | worgil: disallow=all allow=gsm |
| 18:05.32 | *** join/#asterisk metfan2007 (n=metfan20@201.103.91.58) |
| 18:05.59 | worgil | [TK]D-Fender, whet i do so that, not dialing |
| 18:06.11 | metfan2007 | Hi all!!!, I want to call a mySQL store procedure from func_odbc, how can I do this? |
| 18:06.48 | timeshell | what's the root password on *NOW? |
| 18:06.58 | Qwell | timeshell: there is no root, it's admin |
| 18:07.06 | [TK]D-Fender | worgil: then you'd better PASTEBIN something useful for me to look at. |
| 18:07.08 | [TK]D-Fender | ~pb |
| 18:07.09 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:07.11 | timeshell | well my admin password didn't work |
| 18:07.34 | timeshell | nm |
| 18:07.36 | timeshell | I got in |
| 18:07.42 | [TK]D-Fender | timeshell: Then you got it wrong. |
| 18:07.52 | timeshell | no, the password didnt' get changed |
| 18:07.57 | timeshell | was still the default |
| 18:09.26 | metfan2007 | any idea about obdc and store procedure? |
| 18:12.26 | worgil | http://pastebin.com/m2e3cbb49 |
| 18:12.37 | worgil | can anyone tell me what is wrong ? |
| 18:13.58 | timeshell | TKD: http://pastebin.ca/869957 |
| 18:14.40 | [TK]D-Fender | worgil: you need to do this in you PEER entry, and FreePBX is not supported here! |
| 18:14.50 | [TK]D-Fender | worgil: Please go to their channel for support. |
| 18:14.52 | [TK]D-Fender | ~freepbx |
| 18:14.53 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:16.50 | *** join/#asterisk teknoprep (n=chrisraw@unaffiliated/teknoprep) |
| 18:17.00 | [TK]D-Fender | timeshell: pastebin the CLI output of an attempt to log out. |
| 18:18.38 | *** part/#asterisk jmls_work (n=jmls@81.138.42.77) |
| 18:19.38 | timeshell | there is no output |
| 18:19.58 | *** part/#asterisk teknoprep (n=chrisraw@unaffiliated/teknoprep) |
| 18:20.02 | timeshell | I need the password for root |
| 18:20.09 | timeshell | to change the logging level |
| 18:20.58 | [TK]D-Fender | timeshell: where are you looking for you "output"? |
| 18:21.20 | timeshell | The *NOW gui has a CLI via the web browser |
| 18:21.24 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 18:21.32 | timeshell | That is the only place where I know that I can run the agent logoff 5221 |
| 18:21.36 | [TK]D-Fender | timeshell: well thats what I want to see. "core set verbsoe 10" <-------- |
| 18:21.39 | timeshell | But it doesn't give any output when I do it there. |
| 18:22.00 | [TK]D-Fender | and I did not ask you to try any kind of login/out through CLI. |
| 18:22.09 | [TK]D-Fender | timeshell: well thats what I want to see. "core set verbose 10" <-------- |
| 18:22.38 | *** join/#asterisk smace (n=smace@201.18.17.43) |
| 18:23.04 | timeshell | trying logout throught phone does nothign |
| 18:23.18 | [TK]D-Fender | timeshell: and HOW are you trying to "login"? |
| 18:23.23 | smace | I am trying the tutorial: http://www.orderlyq.com/asteriskqueues.html but I do not understand the Item 4 ...exten => 12345,1,Dial(SIP/12345) does it call the same number? |
| 18:24.18 | [TK]D-Fender | smace: read the WHOLE guide and you'll understand what they're doing. |
| 18:25.01 | smace | [TK]D-Fender, I have read. I still not understand it. |
| 18:25.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:25.10 | smace | [TK]D-Fender, the numbers 12346 does not repeat. |
| 18:25.12 | [TK]D-Fender | smace: Go read #6 again. |
| 18:25.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:26.58 | smace | [TK]D-Fender, I have read again. I understand that is for Login and Logout agents, but how is the link with 12346? |
| 18:27.22 | smace | ops/exten => 12345 |
| 18:27.28 | [TK]D-Fender | smace: Keep reading it over and over till you realize some common things they figure you should have thought they were doing.... |
| 18:27.41 | timeshell | TKD: Ok. here is where I am right now. I have one phone logged in. That part seemed to work. So, 1. I can't log in another phone. It prompts for an extension, and stops there. When I do, it does not prompt for a password. It simply seems to time out and says goodbye. 2. I cannot log out. I enter the same extension as I did to log in, it prompts for an extension, but likewise doesn't acknowledge the extension I enter. Times out. Says good by |
| 18:27.45 | [TK]D-Fender | smace: #6 says it all. |
| 18:27.49 | *** join/#asterisk asterisk3700 (n=silver19@66.93.51.2) |
| 18:27.54 | asterisk3700 | Hello |
| 18:27.59 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:28.04 | timeshell | TKD: So far, the only way I can log out a phone is by deleting it's extension and reloading. |
| 18:28.09 | *** part/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 18:28.11 | drako | is there any way to boost the volumen on TDMs cards? |
| 18:28.11 | [TK]D-Fender | timeshell: What the hell are you doing EXACTLY to "log in" ? |
| 18:28.27 | drako | the sound is too low |
| 18:28.42 | asterisk3700 | Can * handle 65 agents in a queue set to ringall? |
| 18:28.45 | [TK]D-Fender | timeshell: and if you hve no CLI output from your "log in" attempt then you are doing something very wrong. |
| 18:29.39 | asterisk3700 | Will I have any issues by doing so? |
| 18:29.51 | [TK]D-Fender | asterisk3700: Go try and see. |
| 18:30.05 | drako | d-fender any idea how to boost the volumen on a TDM card? |
| 18:30.15 | timeshell | TKD: In *NOW, the AgentCallBack Login Extension is set to 5202. So, I dial 5202. I get prompted "Agent Login. Please enter your Agent Number followed by the # key". So I do. Then I get prompted for the password. So I enter it. Then finally I get prompted for a new extension. So I entered the same as my extension. Then it hangs up and I'm logged in. |
| 18:30.17 | [TK]D-Fender | drako: txgain/rxgain. Go look at the sample zapata.conf |
| 18:30.20 | asterisk3700 | well, I wanted to know if anyone had any experience with this |
| 18:30.36 | [TK]D-Fender | timeshell: and your logout attempt? |
| 18:31.20 | timeshell | TKD: I only get as far as the first prompt. After I enter the extension, nothing else happens. It appears to time out and says "Good Bye" |
| 18:31.58 | timeshell | TKD: It also does this same thing if I attempt to log in any other agent after I get the first one logged in. SO far I can only have one agent logged in. |
| 18:32.00 | [TK]D-Fender | timeshell: pastebin the attempt |
| 18:32.08 | smace | [TK]D-Fender, "The agent can enter a new extension to ring when calls come in" ... does it mean I would be forwarding calls to 12345 extension ? Is here the' link' between the extension and agent? |
| 18:32.16 | timeshell | TKD: NOTHING APPEARS. |
| 18:32.37 | timeshell | TKD: On the console it shows exactly what I've said. |
| 18:32.42 | *** join/#asterisk RoyK (n=roy@91.149.9.91) |
| 18:33.01 | [TK]D-Fender | timeshell: if you have no CLI output while dialing the extension you use to log off, then you are not looking at CLI properly, not at the right verbose level, etc. |
| 18:33.05 | timeshell | There are no errors. It doesn't even register that I'm entering the extension |
| 18:33.20 | timeshell | It does display the extension on the first login. |
| 18:33.25 | timeshell | It doesn't for any other |
| 18:33.26 | [TK]D-Fender | timeshell: I never said "errors" I said "show me the ENTIRE damn output". |
| 18:33.54 | [TK]D-Fender | timeshell: If you see nothing, then you are screwing up. Log into the consol properly and set verbose to 10. |
| 18:34.26 | [TK]D-Fender | smace: Keep reading #6 till your eyes bleed, because you should be realizing what its doing... |
| 18:35.13 | smace | [TK]D-Fender, it seems related to hotdesk... but I am not sure when the link between the agent and the extension is done. |
| 18:35.23 | smace | [TK]D-Fender, during logout ? |
| 18:35.44 | [TK]D-Fender | smace: #6 *IS* the link. You login to TELL it where you are and what agent you are. |
| 18:36.08 | timeshell | TKD: Ok. Look. *NOW doesn't do verbage on that web gui as far as I've been able to see. TO do that I believe I have to do it from an ssh login. I have tried to do this but it says access denied. This is probably because I am not root. I have set the verbage as you said to 10, and I watch the console of the thing but it doesn't show anymore info that I have already told you. AGAIN, probably because I am not root and the process is running as root. |
| 18:37.02 | timeshell | Command> core set verbose 10 |
| 18:37.02 | timeshell | Verbosity is at least 10 |
| 18:37.14 | timeshell | That is what I get when I set the verbosity |
| 18:37.14 | *** join/#asterisk steliosk (n=Stelios@athedsl-117730.home.otenet.gr) |
| 18:37.23 | [TK]D-Fender | timeshell: if you see any CLI for your other calls you should be seeing it for your logout attempt |
| 18:37.55 | timeshell | I see the call, I see it playing "PLease enter number". It doesn't register the number i enter after that!! |
| 18:38.02 | *** join/#asterisk ThoMe (n=tm@81.92.168.130) |
| 18:38.09 | [TK]D-Fender | timeshell: pastebin the entire attempt/ |
| 18:38.33 | steliosk | anyone knows what's the 'latest' zaptel that can compile under 2.6.20 ? |
| 18:38.35 | timeshell | I copy the console !! |
| 18:38.49 | timeshell | I mean I can't! |
| 18:38.56 | [TK]D-Fender | timeshell: and why not? |
| 18:38.59 | timeshell | I CAN"T copy the console |
| 18:39.09 | timeshell | IT's on a different computer? |
| 18:39.24 | timeshell | I can't get the asterisk console on a ssh screen without the root password |
| 18:39.29 | [TK]D-Fender | timeshell: you see it through the web console, no? |
| 18:39.33 | timeshell | NO |
| 18:39.36 | timeshell | I don't |
| 18:39.41 | timeshell | I've been trying to say this |
| 18:39.57 | [TK]D-Fender | timeshell: are you using a ray TTY session (text mode) on your server only? |
| 18:40.00 | timeshell | I can enter commands through *NOW |
| 18:40.01 | [TK]D-Fender | raw* |
| 18:40.05 | timeshell | I don't get the verbage |
| 18:40.39 | [TK]D-Fender | timeshell: Ok, you had better get proper access to your server or we're going to run in circles. |
| 18:40.42 | timeshell | IT's the asterisk now server. |
| 18:40.51 | timeshell | It doesn't let you do much |
| 18:41.01 | timeshell | I could do more if I could get in as root |
| 18:41.15 | [TK]D-Fender | timeshell: Then you'd better get a clue fast and get in. |
| 18:41.29 | timeshell | lol |
| 18:41.40 | timeshell | Ok |
| 18:41.54 | timeshell | Building server from scratch on CentOS |
| 18:42.09 | [TK]D-Fender | timeshell: And forget that POS GUI |
| 18:42.25 | timeshell | I would if I wasn't the only person who had to manage it |
| 18:42.42 | timeshell | That's the whole reason I was using *NOW |
| 18:43.00 | [TK]D-Fender | timeshell: maybe Trixbox is for you.... |
| 18:43.56 | *** join/#asterisk kyron_ (n=kyron@211-217-static-ppp.3menatwork.com) |
| 18:44.56 | *** join/#asterisk nohup_ (n=nohup@crack.nohup.nl) |
| 18:45.03 | nohup_ | evening :) |
| 18:45.10 | [TK]D-Fender | nohup_: afternoon |
| 18:45.31 | *** part/#asterisk asterisk3700 (n=silver19@66.93.51.2) |
| 18:46.07 | mvanbaak | back |
| 18:46.16 | timeshell | TKD: Yah maybe. . . was trying to avoid licensing costs |
| 18:46.33 | mvanbaak | hey Fender :) |
| 18:46.46 | drmessano | licensing? |
| 18:46.46 | [TK]D-Fender | timeshell: SE = free |
| 18:47.04 | timeshell | I know |
| 18:47.11 | timeshell | nm |
| 18:47.36 | timeshell | Building ... bbl |
| 18:48.18 | drmessano | [TK]D-Fender: I knew you were a closet TB supporter |
| 18:48.43 | bkruse | [TK]D-Fender: mokoiax is coming along nicely |
| 18:49.01 | bkruse | just got my neo1973 in the mail 3 days ago, passed an iax call (no audio) last night :] |
| 18:49.09 | smace | [TK]D-Fender, do you prefer Ht-503 or SPA-3102? |
| 18:49.24 | mvanbaak | hhmm, from the trixbox website: We've fixed dozens of serious bugs and crash cases that native Asterisk software suffers from |
| 18:49.28 | [TK]D-Fender | smace: SPA-3102 |
| 18:49.31 | mvanbaak | did they ever contribute that ? |
| 18:49.34 | [TK]D-Fender | smace: What do you need? |
| 18:49.39 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 18:49.41 | smace | [TK]D-Fender, why? |
| 18:50.05 | smace | [TK]D-Fender, I have both here, boss is going to buy some new one, and I'm newbie. :) |
| 18:50.18 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 18:50.50 | methods | any idea why when I dial *1 the phone says, "feature is not available".... I only have this issue in one setup. the rest of the polycom phones would allow you to keep typing more numbers... |
| 18:51.13 | tzafrir | The latest asterisk srpm package on trixbox's site seems to be http://yum.trixbox.org/centos/5/SRPMS/repodata/repoview/asterisk-0-1.4.16-1.html |
| 18:51.20 | [TK]D-Fender | smace: I asked you what your NEED was. I don't recommend either of those in all cases. |
| 18:52.02 | smace | [TK]D-Fender, forward calls and use the PSTN as extension. |
| 18:52.07 | Qwell | mvanbaak: no, of course not |
| 18:52.28 | [TK]D-Fender | smace: how many total lines to you have? |
| 18:52.29 | mvanbaak | Qwell: like all the others that claim it :( |
| 18:52.40 | smace | [TK]D-Fender, 5 lines in 5 city. |
| 18:52.42 | tzafrir | The latest rpm package is http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/asterisk-0-1.4.17-1.html , though |
| 18:52.49 | [TK]D-Fender | smace: 1 per? |
| 18:52.52 | timeshell | TKD: The Distributed Call Center option of TB Pro, isn't that the Agent/ Queue thing? |
| 18:53.00 | timeshell | Or is that something else? |
| 18:53.03 | smace | [TK]D-Fender, one city one line. |
| 18:53.08 | [TK]D-Fender | timeshell: don't need the "distriibuted" part. |
| 18:53.23 | [TK]D-Fender | smace: yeah, as a single POP the SPA-3102 is a great choice. |
| 18:53.40 | smace | [TK]D-Fender, why not HT503? Curiosity :) |
| 18:53.48 | [TK]D-Fender | ~gs |
| 18:53.48 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:53.49 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 18:53.51 | *** join/#asterisk UnixDog (n=unixdog@adsl-75-41-223-197.dsl.irvnca.sbcglobal.net) |
| 18:53.59 | timeshell | So TB CE and Pro unpaid does Agent / Queues and such? |
| 18:54.09 | smace | [TK]D-Fender, when using SPA-3102 I have problems with low quality calls, but I have found something at the linksys web site. I'll try later. |
| 18:54.11 | [TK]D-Fender | timeshell: why don't you go over an LOOK :) |
| 18:54.17 | timeshell | I AM LOOKING |
| 18:54.52 | timeshell | I've never used a TrixBOX, so I don't really know what these features mean |
| 18:54.58 | timeshell | I've only used CLI asterisk before |
| 18:55.29 | [TK]D-Fender | timeshell: Go download & try. |
| 18:55.50 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:55.50 | *** join/#asterisk monkeytype (n=Bill@216.207.245.1) |
| 18:56.01 | [TK]D-Fender | smace: www.voxilla.com <- go check out their forums for info on the quality results from various firmwares and ways to tweak your performance. |
| 18:56.31 | *** join/#asterisk asr33 (n=asr33@dsl-207-112-92-225.tor.primus.ca) |
| 18:56.45 | nohup_ | i'm trying to setup iax2 here to a remote box... cause my ISP is blocking all SIP stuff, seemingly... so i setup a remote asterisk server for 'outgoing' calls... |
| 18:57.08 | nohup_ | iax2 registering works... and the remote asterisk box registers at the sip provider nicely aswell.. |
| 18:58.04 | nohup_ | but when i try to make a call... (to my local asterisk box -> remote asterisk box (via iax) -> sip provider... it keeps waiting forever... but no errors in either the sip nor iax2 debug... |
| 18:58.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:58.26 | nohup_ | where should i be looking ? |
| 18:59.03 | *** join/#asterisk nDuff (n=ccd@user-387ocuv.cable.mindspring.com) |
| 19:00.06 | nohup_ | (i can call on the remote asterisk box using the console too) |
| 19:01.02 | mvanbaak | setup an exten on the remote IAX box that runs Echo |
| 19:01.05 | mvanbaak | see if that works |
| 19:01.18 | mvanbaak | if that works, it's not an issue in your IAX2 configuration |
| 19:01.44 | nDuff | I'm seeing asterisk try to jump from priority 5 [a Background() call] to priority 51 in my dialplan without any obvious reason as to why this could be happening. (Since priority 51 doesn't exist, folks get the invalid handler, so callers are greeted with "I'm sorry, that's not a valid extension"). What should I be looking at for reasons this could happen? It only happens on the first run... |
| 19:01.45 | nDuff | ...through -- after the invalid handler jumps back to s,1 everything runs correctly. |
| 19:01.45 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 19:03.02 | mvanbaak | can you pastebin CLI output of that happening ? |
| 19:03.24 | UnixDog | trixbox is not worth it |
| 19:03.31 | UnixDog | install the digium gui |
| 19:03.38 | UnixDog | or get a paid gui |
| 19:03.49 | nohup_ | mvanbaak: ok, i'll try that :) |
| 19:03.54 | nDuff | mvanbaak: there's not much to it, but gladly. http://pastebin.ca/870006 |
| 19:04.45 | [TK]D-Fender | nDuff: I think you'd better be pastebinning this stuff for us to see... |
| 19:05.27 | DarKnesS_WolF | if ur usingg729 all the way how many bandwidth it should take ? |
| 19:05.29 | mvanbaak | ok, now show us the extensions.conf |
| 19:05.41 | DarKnesS_WolF | i find that it takes like 32kb for up and down for one call |
| 19:05.46 | [TK]D-Fender | nDuff: Somebody DIALED 51. it did not jsut to a PRIORITY, but it rather tried to go to an EXTEN based on your IVR input. |
| 19:05.58 | fiXXXerMet | I guess I am looking for recommendations for this deployment. We have 1 central office, 3 remote office of a subsidiary and 1 remote office of another subsidiary. Each location has it's own name, it's own employees, and it's own extensions. How should I handle this? 1 big asterisk box at the central location, with a context for each location? Or a satellite box at each location? Or what? |
| 19:06.03 | mvanbaak | DarKnesS_WolF: roughly 30 kx/s per call |
| 19:06.14 | nDuff | [TK]D-Fender: looks that way, yes, but it's my telephone, and I sure didn't press any buttons... and it tries to go to 51 every time. |
| 19:06.36 | nDuff | dialplan at http://pastebin.ca/870008 |
| 19:06.51 | nohup_ | mvanbaak: ok... seems like it IS a iax problem... but nothing the debug thing outputs looks like an error... |
| 19:07.18 | mvanbaak | nohup_: pastebin both servers iax.conf, extensions.conf and cli log |
| 19:07.58 | [TK]D-Fender | nDuff: Its coming from somewhere... |
| 19:08.17 | nohup_ | mvanbaak: ok, a sec... |
| 19:08.37 | *** join/#asterisk tlosey__ (n=tlosey@70.91.120.177) |
| 19:08.43 | mvanbaak | nohup_: be sure to only obscure the username/secret for security reasons |
| 19:09.14 | nDuff | [TK]D-Fender: also, it jumps there before any of the Background()'ed message is audible; I don't think I could dial that quickly. What's that option that enables the old behavior where asterisk jumps to {priority+$NUMBER} to indicate an error? I'm tempted to see if that's on. |
| 19:09.28 | [TK]D-Fender | fiXXXerMet: What resources at each office? |
| 19:09.36 | nDuff | (the Answer() is likely to be error'ing, as the call is already answered in this case before it gets there) |
| 19:09.59 | *** join/#asterisk teknoprep (n=chrisraw@unaffiliated/teknoprep) |
| 19:10.27 | fiXXXerMet | [TK]D-Fender: A number of users with SIP phones, possible 1 conference room per location as well. |
| 19:10.33 | [TK]D-Fender | nDuff: -- Invalid extension '51' in context 'custom-Mainmenu' on IAX2/catalis-pstn-forward-2 <-- this is pretty explicit. |
| 19:10.41 | *** join/#asterisk PepOSX (n=angeldav@190.72.154.234) |
| 19:11.15 | fiXXXerMet | [TK]D-Fender: The issue that I see with 1 central box is, what happens when users just want to call someone down the hall? It'll have to route the call outside of their lan, to our network, and back, using up a lot of bandwidth |
| 19:11.17 | [TK]D-Fender | fiXXXerMet: I generally advise 1 server a site with some kind of redundency in case centra goes down and you want a central server for termination. |
| 19:11.35 | [TK]D-Fender | fiXXXerMet: another reason I advise site-servers |
| 19:11.39 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 19:11.43 | fiXXXerMet | Makes sense. |
| 19:12.00 | mvanbaak | it depends on the size of the location though |
| 19:12.15 | nDuff | [TK]D-Fender: yes, it *is* indeed pretty explicit, which makes it so unusual that it's trying to do something else. |
| 19:12.16 | mvanbaak | if it's only like 5 ppl we connect them directly to the main pbx |
| 19:12.20 | fiXXXerMet | [TK]D-Fender: I don't know if asterisk supports 'satellite' system or whatever, but if I have a system at each location, how would I make them aware of the extensions and users at other locations? |
| 19:12.30 | fiXXXerMet | mvanbaak: Aye, makes sense |
| 19:12.32 | Qwell | fiXXXerMet: iax2 switch |
| 19:12.36 | Qwell | or dundi |
| 19:12.43 | mvanbaak | fiXXXerMet: we use dundi for that |
| 19:12.49 | teknoprep | fiXXXerMet, dundi |
| 19:12.56 | teknoprep | fiXXXerMet, or just setup your dial plans |
| 19:13.05 | nDuff | s/trying to do something else/claiming it sees digits that the user didn't press/ |
| 19:13.16 | fiXXXerMet | So, aside from redundancy, I wouldn't need to have each context for each location on each box? |
| 19:13.40 | Qwell | fiXXXerMet: dundi handles that kind of stuff |
| 19:13.42 | [TK]D-Fender | nDuff: perhaps a noisy line passing things on? shared line with people butting in? Gotta be something... |
| 19:13.44 | teknoprep | fiXXXerMet, check out dundi on www.voip-info.org |
| 19:13.49 | fiXXXerMet | Awesome. Thanks. |
| 19:13.56 | fiXXXerMet | I've heard of dundi but haven't used it yet. |
| 19:14.21 | mvanbaak | fiXXXerMet: for me the pdf by jr was very helpfull when I first started with dundi |
| 19:15.29 | fiXXXerMet | mvanbaak: Which pdf? There is a 'dundi-intro.pdf' linked at the voip-info wiki |
| 19:15.42 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:15.56 | nDuff | [TK]D-Fender: it's a clean POTS line over a PRI to one asterisk server, which forwards it over iax2 to another. Doesn't iax2 do DTMF out-of-band (so it'd be detected by the first host with the PRI going in)? |
| 19:16.27 | [TK]D-Fender | nDuff: well I'm not sure what to say at this point... |
| 19:18.33 | *** join/#asterisk guillote_GNU (n=guillote@host214.190-31-88.telecom.net.ar) |
| 19:18.49 | defsdoor | I've got a sangoma remora all connect up but get nothing on dialling out and see nothing on incoming call either |
| 19:18.55 | defsdoor | any pointers where to look ? |
| 19:19.05 | nohup_ | mvanbaak: done the pasting... http://www.pastebin.ca/870021 |
| 19:19.13 | nohup_ | i hope that's enough... cause i have no idea where to look :) |
| 19:19.22 | defsdoor | asterisk knows it's there etc.. |
| 19:19.42 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 19:19.44 | [TK]D-Fender | defsdoor: your configs & CLI would be a good start. |
| 19:20.42 | J4k3 | nDuff's line looks like a bunch of random telco acronyms shoved together |
| 19:23.27 | mvanbaak | # |
| 19:23.29 | mvanbaak | [Jan 23 20:16:50] WARNING[17448] chan_iax2.c: I don't know how to authenticate crack to 10.0.5.1 |
| 19:23.37 | drmessano | exten ==> sip,1,trunk,IAX3,skype,centraloffice(exchange) is NOT a valid extension? |
| 19:23.53 | drmessano | aww crap |
| 19:24.20 | [TK]D-Fender | mvanbaak: His crack must be labeled "exit only". |
| 19:24.31 | [TK]D-Fender | mvanbaak: ask bkw about opening a few ports for him! |
| 19:24.39 | mvanbaak | ;) |
| 19:24.47 | [TK]D-Fender | DAMN I'm good ;) |
| 19:24.56 | J4k3 | goatse ports. |
| 19:24.57 | drmessano | Open port 6969 to 0.0.0.YES |
| 19:25.00 | mvanbaak | ur on a roll today man |
| 19:25.36 | mvanbaak | nohup_: what's the extensions.conf line look on the local server ? |
| 19:25.56 | drmessano | I screwed my example up |
| 19:25.59 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 19:26.01 | drmessano | I left out BRI and Google |
| 19:26.27 | mvanbaak | and h323 |
| 19:26.48 | mvanbaak | and most important: skinny ;) |
| 19:26.49 | nohup_ | mvanbaak: hmm.. how come it can't authenticate... since i DO see it when i type "iax2 show registry" ? |
| 19:27.06 | drmessano | I know better than that.. h323(pedestal,google) sends calls to voicemail |
| 19:27.09 | drmessano | Duh |
| 19:27.14 | nohup_ | and... that line.. exten => 500,1,Dial(IAX2/crack@netb.local.nohup.nl/0031528272772@default) (just a test) |
| 19:27.16 | [TK]D-Fender | nohup_: because registering has nothing to do with authing calls you place |
| 19:27.42 | nohup_ | [TK]D-Fender: oh, i see... |
| 19:27.59 | drmessano | WARNING[65536] chan_smack.c: crack is not for everyone |
| 19:28.45 | mvanbaak | nohup_: you should put a type=peer in 'local' servers iax.conf |
| 19:29.20 | mvanbaak | like: [tocrack] |
| 19:29.46 | x86 | drmessano: hahahaha |
| 19:29.47 | mvanbaak | and in your dialplan use: Dial(IAX2/tocrack/${EXTEN}) |
| 19:29.49 | x86 | 65536 |
| 19:29.51 | x86 | that's great |
| 19:30.19 | nohup_ | mvanbaak: ohh... ok... so only the registry entry isn't enough, so to say... :) |
| 19:30.24 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 19:30.35 | a1fa | damn.. PAP2 is so unatable with Asterisk 1.14.17 |
| 19:30.52 | a1fa | I am loosing the peer every couple of minutes |
| 19:31.21 | drmessano | Working here |
| 19:31.24 | drmessano | 1.14.17 |
| 19:31.26 | a1fa | its ridicilous |
| 19:31.28 | drmessano | LOL |
| 19:31.32 | drmessano | 1.4.17 |
| 19:31.37 | a1fa | heh.. crapphone 101 is more stable |
| 19:31.48 | drmessano | I think its your box or line |
| 19:31.54 | a1fa | Toerkeium: <sip:202@192.168.1.3:5060>;tag=85530c16f22ca6cai0 |
| 19:31.58 | a1fa | no.. box is fine |
| 19:32.00 | drmessano | 1.4.17 is working fine with all my PAP2s |
| 19:32.03 | a1fa | line is fine |
| 19:32.16 | drmessano | Ok, then you have a version of 1.4.17 with the PAP2.c issue |
| 19:32.16 | a1fa | may be the pap2.. i am running the latest firmware on the pap |
| 19:32.24 | a1fa | drmessano : ?? |
| 19:32.30 | drmessano | Dude, seriosuly |
| 19:32.36 | drmessano | Check the PAP2 |
| 19:32.38 | twisted | CHEAP ASTERISK SERVER: http://www.pacificgeek.com/nooner.asp?P=SS |
| 19:32.53 | a1fa | PAP2 reports no errors |
| 19:33.02 | a1fa | Server: Linksys/PAP2-3.1.22(LS) |
| 19:33.21 | drmessano | Then nothing is wrong a1fa |
| 19:33.24 | drmessano | Youre right |
| 19:33.47 | a1fa | i dont know if pap2 is chocking on nating |
| 19:33.54 | a1fa | it looks like it is |
| 19:34.25 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 19:34.29 | drmessano | Hmmm |
| 19:34.39 | a1fa | <PROTECTED> |
| 19:34.46 | a1fa | i am seeing my internal ip in sip header |
| 19:34.49 | [TK]D-Fender | twisted: Yup, I always love those SFF boxes. |
| 19:35.04 | [TK]D-Fender | twisted: Room for 2 PCI cards.. |
| 19:35.14 | drmessano | Its easier to get to the blown caps to smell them |
| 19:35.29 | twisted | [TK]D-Fender :) |
| 19:35.49 | b1ch0 | hi everybody, how can i see wich codes is in use during a conversation ? |
| 19:35.51 | drmessano | I have two GX270s in the room here with me |
| 19:36.02 | a1fa | [TK]D-Fender : hello friend |
| 19:36.02 | drmessano | 3rd mobo for each |
| 19:36.04 | [TK]D-Fender | twisted: Mind you mine is an Antec HTPC case and runs jsut about the biggest scress on any * server :) |
| 19:36.06 | *** join/#asterisk dkatz334 (n=guest@128.238.35.151) |
| 19:36.12 | [TK]D-Fender | a1fa: Read up : |
| 19:36.14 | [TK]D-Fender | ~sipnat |
| 19:36.15 | 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 |
| 19:36.16 | [TK]D-Fender | ^^^ |
| 19:36.20 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 19:36.39 | drmessano | Heading to the office.. be back from there |
| 19:36.46 | a1fa | [TK]D-Fender : did that.. |
| 19:38.13 | b1ch0 | sorry codecs |
| 19:38.28 | [TK]D-Fender | a1fa: then pastbin your sip.conf, and CLI SIP debug. |
| 19:38.42 | [TK]D-Fender | b1ch0: "sip show channels" |
| 19:38.50 | a1fa | [TK]D-Fender : the phone works fine, except goes in and out of conciousness |
| 19:38.59 | [TK]D-Fender | a1fa: PASTEBIN <------- |
| 19:39.05 | a1fa | my BT101 phone rock solid stable behind 2 nats.. to make things worse |
| 19:39.50 | a1fa | the only issue i see is PAP2 sending sip headers "To: <sip:202@192.168.1.3:5060> |
| 19:40.12 | a1fa | but * is smart enough to reply to the actual ip:source-port |
| 19:41.29 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 19:42.01 | mintee | I have a extension setup to go straight to a voicemailbox, but i would like it to emulate a few rings first. How can I do that? |
| 19:43.11 | a1fa | ok.. pap2 is rock solid stable at the moment.. i told it to re-register every 30s |
| 19:43.52 | a1fa | my * is generating 200 mb/day in sip registration |
| 19:45.17 | [TK]D-Fender | a1fa: ....... |
| 19:45.25 | mintee | heh, nevermind |
| 19:45.41 | a1fa | [TK]D-Fender : =) |
| 19:45.42 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 19:45.50 | a1fa | [TK]D-Fender : i just changed nat keep alive to 5s |
| 19:45.51 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 19:45.55 | a1fa | on the pap2 |
| 19:46.37 | a1fa | this feels good ;P |
| 19:47.21 | dkatz334 | mintee, jsut saw your Q, did you get the answer? Ringing is the app you want. |
| 19:47.35 | mintee | dkatz334, yep, sorry i asked too early. |
| 19:47.42 | dkatz334 | np |
| 19:48.02 | *** join/#asterisk janinge (i=j@ninge.net) |
| 19:48.14 | mintee | but i do have another question... I have 100 DID's and some of them I just want to go straight to a voicemail box. Do i have to setup an extension just for that? |
| 19:48.17 | b1ch0 | here me again, there is a stuff over codecs that i cant understand |
| 19:48.28 | b1ch0 | please can anyboby help me ? |
| 19:48.39 | dkatz334 | mintee, are the DIDs consecutive? |
| 19:48.48 | *** part/#asterisk janinge (i=j@ninge.net) |
| 19:48.52 | mintee | yeah |
| 19:49.06 | mintee | and they are coming in 10 digits from the lec |
| 19:49.16 | b1ch0 | i have a phone using g729 and a softphone using GSM and they can comunicate, how is it posible ? |
| 19:49.30 | *** join/#asterisk janinge (i=j@ninge.net) |
| 19:49.57 | dkatz334 | so how about in your "inbound" context, how about something like exten => _12345678XX,1,Voicemail(mailbox@context) |
| 19:50.44 | dkatz334 | That would cover anything which changed the last two numbers, you could also create individual items for stuff you DONT want to go to VM. |
| 19:51.27 | mintee | dkatz334, ok I kinda see what your saying, but I don't see the variable to goto XX mailbox |
| 19:51.45 | b1ch0 | that means that i can have 729 phones and 711 able to communicate each other ? |
| 19:51.52 | dkatz334 | b1ch0, the softphone is doing g729, the phone is doing gsm or someone is transcoding. |
| 19:52.05 | b1ch0 | that means that asterisk is doing transcoding ? |
| 19:52.22 | dkatz334 | b1ch0, yep, it'll do that if you have sufficient g729 licenses. |
| 19:52.39 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 19:52.46 | dkatz334 | mintee, you want different mailboxes for each DID? |
| 19:53.19 | dkatz334 | b1ch0, a warning about transcoding, it's something you want to avoid. It can have bad effects on system performance and call quality. |
| 19:53.27 | mintee | dkatz334, yeah |
| 19:54.06 | b1ch0 | ok i understand, but checking i got -- Format for call is GSM |
| 19:54.27 | b1ch0 | and on sip show channels i got ulaw !!! |
| 19:54.39 | mintee | so 12345678XX will goto 12345678XX voicemail box specifically |
| 19:54.51 | *** join/#asterisk Olobola (n=casper_s@c-24-23-198-187.hsd1.mn.comcast.net) |
| 19:54.55 | mintee | and 12345678XY will goto 12345678XY voicemail box |
| 19:54.56 | dkatz334 | exten => _12345678XX,1,Voicemail(${EXTEN:8:9}@context) |
| 19:55.01 | b1ch0 | and on iax2 show channels i got gsm |
| 19:55.14 | b1ch0 | ... very strange, isnt it ? |
| 19:55.33 | mintee | may i ask what the leading underscore is for? |
| 19:55.40 | dkatz334 | No you may not. |
| 19:55.43 | dkatz334 | jk |
| 19:55.43 | mintee | lol |
| 19:56.16 | dkatz334 | Sure, it tells asterisk that what follows is to be evaluted as possibly meaning more than one extension. |
| 19:56.26 | *** part/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 19:56.30 | mintee | ok |
| 19:56.36 | dkatz334 | _ at the beginning is required if you're going to use X, N, or the . |
| 19:56.48 | mintee | N being? |
| 19:56.54 | mintee | the same as X |
| 19:57.06 | dkatz334 | X = 0123456789 |
| 19:57.10 | dkatz334 | N = 23456789 |
| 19:57.32 | *** join/#asterisk detch (n=PIERO@unaffiliated/detch) |
| 19:57.45 | mintee | oh, ok. interesting.. little over my head but i'm just starting... thanks. I'm going to go tinker now. ;) |
| 19:57.46 | dkatz334 | . = {X} (X repeated indefinitely) |
| 19:57.50 | *** part/#asterisk detch (n=PIERO@unaffiliated/detch) |
| 19:57.57 | dkatz334 | Tinkering is the best way to learn. :) |
| 19:58.04 | [TK]D-Fender | dkatz334: Close, but not quite. |
| 19:58.04 | dkatz334 | Don't tell my CS students that though. |
| 19:58.09 | dkatz334 | oops |
| 19:58.16 | mintee | lol |
| 19:58.19 | dkatz334 | what'd I get wrong? |
| 19:59.42 | [TK]D-Fender | dkatz334> . = {X} (X repeated indefinitely) <- no, not "x" |
| 20:00.01 | [TK]D-Fender | dkatz334: thats 1 or more chards of ANY kind. |
| 20:00.05 | [TK]D-Fender | chars* |
| 20:00.10 | mintee | and the $EXTEN:8:9 is calling digit placement i assume? |
| 20:00.48 | dkatz334 | $EXTEN:8 |
| 20:00.50 | dkatz334 | er |
| 20:00.58 | mintee | exten => _12345XXXX,1,Voicemail(${EXTEN:6:7:8:9}@context) |
| 20:01.09 | [TK]D-Fender | mintee: No. |
| 20:01.18 | mintee | or am i just making sh*t up now, lol |
| 20:01.19 | dkatz334 | $EXTEN:8:9 means only the 8th and 9th digit, er 9th and 10th digit (start wiht zero) |
| 20:01.21 | [TK]D-Fender | mintee: Go read the chapter on variable handling again |
| 20:01.44 | dkatz334 | mintee: $EXTEN:6:9 would work if you want four digit mailbox numebrs. |
| 20:01.55 | [TK]D-Fender | dkatz334: nope |
| 20:03.12 | SteveTotaro | ~pb |
| 20:03.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:04.04 | SteveTotaro | anyone know why this would occur |
| 20:04.11 | SteveTotaro | or how to troubleshoot |
| 20:04.16 | SteveTotaro | http://pastebin.com/d760f7caf |
| 20:05.09 | dkatz334 | Steve, looks like your DB doesn't exist or mysql server is offline. |
| 20:05.25 | SteveTotaro | everything is fine and then suddenly sip peers become unavailable |
| 20:05.52 | SteveTotaro | there are 3 servers in the same rack |
| 20:05.57 | dkatz334 | mintee, I think you want to have a look at: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf |
| 20:06.18 | SteveTotaro | two asterisk using realtime hitting the same db |
| 20:06.28 | SteveTotaro | only one is having issues |
| 20:06.44 | dkatz334 | where does the DB reside? |
| 20:06.47 | mvanbaak | connectivity issues? routing borked ? |
| 20:07.47 | b1ch0 | even changing codec on the phone (to 723) i still have ulaw and gsm |
| 20:08.55 | SteveTotaro | on a server in the same rack |
| 20:09.06 | *** join/#asterisk oej_ (n=olle@soll4-125.cust.blixtvik.net) |
| 20:09.26 | SteveTotaro | if i do a restart everything is good for a while |
| 20:09.51 | hmmhesays | I'm not sure if I can go on without heath ledger |
| 20:09.57 | mvanbaak | max connection limit ? |
| 20:10.02 | SteveTotaro | then sip show peers status gets high in ms |
| 20:10.22 | SteveTotaro | then they go unavailable and then mysql drops |
| 20:11.24 | phix | my nokia phone is weird, I call a number, doesn't matter what number, and it calls s,1, |
| 20:11.28 | phix | well s |
| 20:11.46 | phix | unless that is some fallback thing doing that :\ |
| 20:13.07 | dkatz334 | phix, give us a little more to go on. |
| 20:13.15 | *** join/#asterisk ZX81 (n=ZX81@202.20.97.211) |
| 20:13.50 | mvanbaak | SteveTotaro: maybe you hit some limit on the box? like open files or something ? |
| 20:13.57 | dkatz334 | Steve, might you have a networking issue with that server? soulds like you might be loosing connectivity. |
| 20:14.13 | hmmhesays | sounds like a dialplan problem |
| 20:14.30 | mvanbaak | hmmhesays: a dialplan that makes mysql connections drop ? |
| 20:14.43 | hmmhesays | mvanbaak: sorry that was for phix |
| 20:14.44 | phix | dkatz334: yeah I will later on, need to go to work now :( |
| 20:14.50 | mvanbaak | mocker: gheh, trying to get bridged networking setup eh ? |
| 20:15.11 | phix | brctl is your friend :) |
| 20:15.18 | hmmhesays | yeah, vmware is your friend |
| 20:15.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:15.37 | mvanbaak | nothing wrong with virtualbox |
| 20:17.39 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 20:18.19 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 20:18.28 | phix | I am using virtualbox as we speak :) |
| 20:18.49 | phix | I would like GL support though :( |
| 20:19.14 | phix | hardware accellerated so I can get rid of my Windows partition for good |
| 20:19.16 | SteveTotaro | it is some sort of connectivity problem |
| 20:19.40 | SteveTotaro | but i am sshed into the box and the connection stays up |
| 20:19.58 | SteveTotaro | and i am not losing pings halfway around the world |
| 20:20.29 | dkatz334 | Steve: Might be a transient problem, how about leaving a ping session up to the other box and periodically checking it. |
| 20:21.29 | dkatz334 | Steve: In other words, it might go down for long enough to loose your SIP sessions but not long enough to loose your SSH. |
| 20:22.37 | SteveTotaro | possibly, strange issue |
| 20:22.38 | SteveTotaro | i have to think it is not an asterisk issue |
| 20:22.57 | SteveTotaro | they did have srvlookup=yes in sip.conf |
| 20:23.11 | SteveTotaro | i thought that was the problem but it didn't make a difference |
| 20:23.53 | dkatz334 | Steve: I'd agree, the loss of connectivity to the mysql server and the SIP messages are probably indicative of a larger problem. Just my thinking though. |
| 20:24.21 | mvanbaak | maybe the switchport is dying |
| 20:24.27 | mvanbaak | I seen it happen. looks the same |
| 20:24.39 | SteveTotaro | hmmmm |
| 20:24.46 | dkatz334 | mvanbaak, I was thinking along the same lines, switch, NIC, even cable. |
| 20:25.04 | mvanbaak | indeed |
| 20:25.36 | SteveTotaro | hard to troubleshoot that since the server is in germany |
| 20:25.44 | SteveTotaro | and i am in washington DC |
| 20:26.01 | SteveTotaro | but yeah, always check cabling and ports first |
| 20:26.08 | mvanbaak | SteveTotaro: I'm in .nl |
| 20:26.17 | mvanbaak | if the price is right I can drive to germany |
| 20:26.37 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:26.37 | mvanbaak | nice invoice: check cable - 1200 euro |
| 20:26.38 | SteveTotaro | well they have guys to check cables |
| 20:26.44 | SteveTotaro | right |
| 20:26.46 | SteveTotaro | lol |
| 20:26.47 | mvanbaak | ;) |
| 20:29.47 | *** join/#asterisk jsmith- (n=john@peter.mcinerney.id.au) |
| 20:30.44 | defsdoor | ~paste |
| 20:30.44 | jbot | well, paste is http://rafb.net/paste/, or see also pb |
| 20:31.28 | mvanbaak | pastebin.ca |
| 20:33.08 | *** join/#asterisk AndyGraybeal (i=minDlabs@node48.34.251.72.1dial.com) |
| 20:33.32 | Shaun2222 | with ringall on a queue after priority 1 is rang when it tries priority 2 does it also include priority one in that ring? |
| 20:33.55 | defsdoor | I've installed a sangoma a200, seems to be setup ok, zaptel sees it - is there a way to test (see ring perhaps) before asterisk ? |
| 20:34.13 | [TK]D-Fender | Shaun2222: it keeps going at a given priority unless no agents are available (kicked, paused, or on-call already) |
| 20:34.54 | *** join/#asterisk jblack (n=jblack@pool-71-181-145-13.sctnpa.east.verizon.net) |
| 20:35.27 | Shaun2222 | [TK]D-Fender: say it rings all of priority one.. no one answeres... so it goes ot priority 2, does it also run priority one while it's ringing 2? |
| 20:35.42 | defsdoor | include pri 1 extensions in pri 2 |
| 20:35.45 | Shaun2222 | s/run/ring/ |
| 20:35.56 | [TK]D-Fender | Shaun2222: No, as long as anyone is "available" in prio 1 #2 should never get called |
| 20:36.08 | Shaun2222 | oh... hmmm |
| 20:36.47 | Shaun2222 | has i always worked like this... i could have sworn i used to set cell phones as priority 2 and if none of the members assigned to priority 1 answered it would go to priority 2. |
| 20:37.41 | b1ch0 | still here botering with codecs, now i have 2 phones with 722 codecs (sip) enabled, on my show channels i still have ulaw .... i am confused |
| 20:39.29 | b1ch0 | anybody can help me to understand this ? |
| 20:39.38 | [TK]D-Fender | Shaun2222: Keep swearing :) |
| 20:40.03 | [TK]D-Fender | b1ch0: go look at your sip peer setup |
| 20:41.04 | mvanbaak | b1ch0: make sure you have set 'disallow=all' and 'allow=g722' in all peers in your sip.conf |
| 20:41.16 | mvanbaak | make sure you have the disallow before the allow |
| 20:45.31 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 20:46.36 | *** join/#asterisk bmcghee (n=brentmcg@S01060015e916bf8b.ok.shawcable.net) |
| 20:47.02 | bmcghee | for asterisk dialing prefix for outgoing. for example when i dial 554 XXXX i want it to add 1+250 infront |
| 20:47.37 | b1ch0 | i put it, but i got channel unavailable |
| 20:47.53 | jblack | woot. Daughter is bringing home straight As for another quarter. I get to keep my title as "great dad" |
| 20:48.04 | defsdoor | jblack: \o/ |
| 20:48.46 | *** join/#asterisk bhima (n=gopi@72.19.4.237) |
| 20:49.10 | mvanbaak | lol jblack |
| 20:49.18 | mvanbaak | jblack: can she configure an asterisk box ? |
| 20:49.50 | b1ch0 | what i am to tring to test is if phones are able to work with a different codec than 711 |
| 20:50.26 | mvanbaak | if you get channel unavail it prolly means not all endpoints support g722 |
| 20:50.34 | [TK]D-Fender | b1ch0: pastebin your configs and the CLI output of your call attemp with SIP debug enabled. |
| 20:50.36 | [TK]D-Fender | ~pb |
| 20:50.36 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:50.37 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 20:50.39 | bhima | So, I know how to dial multiple people with an '&', but I'm wondering how to achieve what I want: Incoming call goes to three "people", each of which I will dynamically decide which one or more channels should be used to reach. |
| 20:51.10 | *** join/#asterisk ZX81 (n=ZX81@202.49.106.158) |
| 20:51.38 | bhima | Basically, for each DID I have a list of people whose phones should ring. |
| 20:51.54 | bhima | Their phones may be cellphones or SIP hardphones. |
| 20:52.06 | [TK]D-Fender | bhima: then on each exten for each DID put only those people. |
| 20:52.40 | [TK]D-Fender | bhima: and including PSTN numbers in a "group dial" is hard in many case, unworkable in others. |
| 20:52.48 | bhima | [TK]D-Fender: If one person is supposed to get calls from two extensions, then I'll need to carefully go through and edit a dial string... |
| 20:53.21 | bhima | [TK]D-Fender: I'd like to have one place to adjust the numbers for each specific person. |
| 20:53.26 | dkatz334 | Bhima: What you might want is to setup so the SIP phones ring first, then timeout, then you go on to dial the Cell phones. |
| 20:53.46 | dkatz334 | much easier. |
| 20:53.56 | bhima | dkatz334: These people are in and out a lot, and it often takes a few seconds for cellphones to start handling the call... |
| 20:53.58 | [TK]D-Fender | agreed |
| 20:54.42 | dkatz334 | Maybe a "DEVSTATE" option for them to autoforward to the cell phones... They push a button on the SIP phone and it bypasses rining the SIP phone? |
| 20:54.52 | dkatz334 | Might even be able to do it with the DND option. |
| 20:54.58 | bhima | dkatz334: Also, that doesn't really solve the initial problem: if I do SIP then cellphones, I'll still have a series of dial-strings with cellphone numbers that will need to be changed if the cellphone number changes... |
| 20:55.17 | [hC] | Hey is it still the case where if i have an old ass wildcat digium te410 PRI card, if i send it back to them, will they send me a fancy new card? |
| 20:55.22 | dkatz334 | I'd solve that with global variables at the top of extensions.conf |
| 20:55.26 | dkatz334 | very easy to change. |
| 20:55.39 | bhima | (We have people traveling a lot and using local SIM cards) |
| 20:55.57 | bhima | [TK]D-Fender: Why is using PSTN numbers in a group dial hard...? |
| 20:55.59 | dkatz334 | So put it in the ASTDB and allow them to change it by dialing a specific number? |
| 20:56.43 | bhima | dkatz334: Right, I was just wondering if there was a good clean way to do it. I'm a programmer; globals make my hair stand on end. :) |
| 20:57.04 | mvanbaak | bhima: use the ASTDB |
| 20:57.12 | *** join/#asterisk jameysharp (n=jamey@71.216.22.202) |
| 20:57.15 | [TK]D-Fender | bhima: because of call progress, etc. Remember multiple entries in a dial stop on the first one that answers. On analog channels a call is considered answered the moment it is *placed*. |
| 20:57.16 | mvanbaak | and local channels |
| 20:57.42 | mvanbaak | somethig like: Dial(Local/joe@ppl/n&Local/luke@ppl/n) |
| 20:57.43 | bhima | [TK]D-Fender: Is it possible to do something like have the receiver dial a digit to indicate they are alive? |
| 20:57.50 | [TK]D-Fender | bhima: and if you are forwarind to a cell and the cell is off, or out of range, the cell's VM will pick it up instantly meaning noone will ever get to really answerit |
| 20:57.51 | mvanbaak | use the ppl context to call macros |
| 20:57.57 | *** join/#asterisk DrkShadow_ (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 20:58.04 | mvanbaak | in that macro check the ASTDB what number to dial |
| 20:58.05 | [TK]D-Fender | bhima: There are ways, but this is whree things get "hard" |
| 20:58.40 | DrkShadow_ | hey, I'm getting "ztdummy: Unknown symbol rtc_register, rtc_unregister, rtc_control" when I try to modprobe ztdummy.. anyone know a fix? I saw one, but crc_ccitt is loaded as a module, zaptel is loaded and using it, so I dunno. |
| 20:59.06 | bhima | mvanbaak: Ok, I'll take a look into that, thanks. |
| 20:59.31 | jameysharp | my Nokia E70 works with Asterisk 1.2, but seems to ignore all SIP packets coming from Asterisk 1.4. the Twinkle softphone works with my 1.4 setup. any clues? |
| 21:01.15 | *** join/#asterisk nephfl (n=none@wsip-70-168-186-225.ga.at.cox.net) |
| 21:01.57 | jblack | mvanbaak: Nope. she's only gotten as far as keeping her ubuntu laptop up to date. |
| 21:02.10 | mvanbaak | jblack: no A there then ;) |
| 21:02.24 | jameysharp | the only interesting differences I can see in the SIP traffic (from 'sip debug ip' on both Asterisk servers) are that 1.4 adds a "Supported: replaces" header to its responses, and the realm name changed (so I told the phone to use the new realm). |
| 21:03.29 | jblack | mvanbaak: Heh. She decided that she wants to be a neurologist... as soon as she decided that was a field she liked, she stopped paying much attention to computers as more than a device. |
| 21:05.01 | jblack | Leaves me kinda stuck. What should I do? "You should be a admin/programmer like me for 70/hrs a week and $60-120K a year, instead of someone that works 30 hours a week and pulls in a quarter million" |
| 21:05.51 | *** part/#asterisk bmcghee (n=brentmcg@S01060015e916bf8b.ok.shawcable.net) |
| 21:05.57 | b1ch0 | great it finally works (between 2 codecs), but now what can i procede if i want a phone working with g729 (or other like g722 o 723) and another with g711 ? will asterisk transcode the call using a preferred codec ? |
| 21:06.48 | [TK]D-Fender | b1ch0: * will transcode if needed. |
| 21:06.59 | dkatz334 | b1ch0 in sip.conf you can sepcify disallow=all then allow=g711 ; asterisk will worry about the transcoding... But performance/quality may still be an issue. |
| 21:07.19 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 21:07.25 | bhima | jblack: Good neurologists are *really* useful, but they end up being *insanely* specialized. My wife saw one who specialized in movement disorders. |
| 21:08.26 | jblack | bhima: A shitty neurosurgeon easily clears $180k. The best neurosurgeon can get whatever he wants because rich people with brain disorders don't care how much it costs. |
| 21:08.37 | [TK]D-Fender | bhima: Really? I know a few people that have passed this way that were full of shit... |
| 21:09.01 | [TK]D-Fender | bhima: Oh wait... thats NEUROTIC , and in need of a BOWEL MOVEMENT. |
| 21:09.07 | jblack | Anyways, point being, she picked a better field than I'm in, so ... I can't really complain. |
| 21:09.21 | [TK]D-Fender | bhima: easily mistaken! |
| 21:10.01 | bhima | jblack: I wonder how many fail to actually get into the profession... |
| 21:10.22 | [TK]D-Fender | jblack: Neuro-surgery .... it ain't Raw-Cat Science! |
| 21:10.34 | jblack | Pardon, neurologist, not neurosurgeon. |
| 21:10.52 | bhima | (I've only seen one neuro-surgeon, but somewhat more neurologists) |
| 21:11.02 | jblack | same for me. |
| 21:11.22 | jblack | anyways, I think she can do it. She's a bright child and knows how to buckle down at 13. |
| 21:11.38 | nhuisman_work | what is that like 12 years of school |
| 21:11.53 | nhuisman_work | plus 4 years of being under someone |
| 21:12.22 | jblack | I guess that's why not everyone's a neurologist. :) |
| 21:12.36 | nhuisman_work | yeah it's a long track |
| 21:12.38 | jblack | There's no way I could have done the medical route. |
| 21:12.57 | nhuisman_work | yeah I dunno I could see myself doing it but I probably would have been A LOT of work |
| 21:13.00 | b1ch0 | TK, thanks i dont want to flood the channel anymore ... where can ifind info or howto over transcoding |
| 21:13.01 | nhuisman_work | you have to study like mofo |
| 21:13.08 | jblack | I could now, but I didn't catch onto the important stuff until I was much too old for the field. |
| 21:13.09 | nhuisman_work | and know ur shit people's live are in your hands |
| 21:13.19 | [TK]D-Fender | jblack: no, no, no, you misunderstand her. Whats she's really trying to tell you is that she just likes to "play doctor", and its about time you had "the talk" ;) |
| 21:13.27 | nhuisman_work | ooooo |
| 21:13.31 | nhuisman_work | lols |
| 21:13.34 | jblack | [TK]D-Fender: I'll tell her that. |
| 21:13.42 | [TK]D-Fender | b1ch0: there's notihng to "find out" * just DOES IT. |
| 21:13.44 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:14.09 | nhuisman_work | [TK]D-Fender, hey is there some tests I can run to stress my * and servers? Like making lots of concurrent calls and using features. |
| 21:14.14 | grandpapadot | Hi all. In asterisk 1.2.24, is there a way to have asterisk log the queue "Log" into a mysql database as with the CDR logging? |
| 21:14.15 | b1ch0 | :-) |
| 21:14.22 | [TK]D-Fender | nhuisman_work: "For everything else, there's malpractice insurance" |
| 21:14.29 | nhuisman_work | [TK]D-Fender, heh. |
| 21:14.36 | *** part/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 21:14.39 | [TK]D-Fender | nhuisman_work: |
| 21:14.44 | [TK]D-Fender | nhuisman_work: "SIPP" |
| 21:14.49 | *** part/#asterisk jameysharp (n=jamey@71.216.22.202) |
| 21:14.54 | nhuisman_work | k i'll google it |
| 21:15.39 | nhuisman_work | hmm |
| 21:16.16 | [TK]D-Fender | grandpapadot: Yes. |
| 21:16.50 | jblack | the talk? |
| 21:16.58 | jblack | that's just messed up man |
| 21:17.20 | bhima | My T1 to a cellphone reports call progress information pretty well. |
| 21:17.26 | grandpapadot | TK: Thanks. What config file controls that? |
| 21:18.05 | [TK]D-Fender | grandpapadot: There is a section of THE BOOK dedicated to this and the Wiki. Go take a look. |
| 21:18.14 | grandpapadot | Ok, thanks again. |
| 21:18.38 | [TK]D-Fender | jblack: and you are a little slow today :) |
| 21:18.47 | [TK]D-Fender | jblack: You should have caught that instantly |
| 21:19.13 | grandpapadot | TK: I found some info for 1.4, to your knowlege will 1.2 do this logging? |
| 21:20.38 | [TK]D-Fender | grandpapadot: Yes, thats part of asterisk-addons... |
| 21:20.49 | grandpapadot | Cool. Thanks again. |
| 21:20.59 | jblack | I went for a smoke. |
| 21:21.02 | jblack | what did I miss? |
| 21:21.06 | mvanbaak | me |
| 21:21.42 | jblack | "The talk" ? No. We just did the birds and bees talk... not about the birds and bees, but about the level of indirection. |
| 21:22.33 | jblack | The sound effects made her laugh so hard, that I think she'll never be able to have sex without giggling. :) |
| 21:24.06 | [TK]D-Fender | jblack: Time to start being the "fated in-laws" you're destined to become :) |
| 21:24.14 | [TK]D-Fender | hated* |
| 21:24.27 | [TK]D-Fender | wow, temporally convenient typo :) |
| 21:24.51 | kyron | LOL |
| 21:25.01 | jblack | "Dad, I don't understand the bee part." "Well, hon, I guess it's becuase parents can talk about the pollen getting from the male part of the flower to the female part. Its easier to discuss than [repeatedly punches palm with a "HURH!" sound]" |
| 21:25.28 | mvanbaak | just be honest to them |
| 21:25.34 | mvanbaak | tell them the real story |
| 21:26.00 | [TK]D-Fender | yeah, in the same vein as the "Creation Museum"! |
| 21:26.19 | jblack | Her mom is a slut. She understands sex fine. |
| 21:26.30 | mvanbaak | SteveTotaro: you still here ? |
| 21:26.33 | jblack | As her mom is the worst kind of slut.. the verbose kind |
| 21:26.35 | [TK]D-Fender | jblack: lol. |
| 21:26.56 | SteveTotaro | yes |
| 21:27.03 | [TK]D-Fender | BBIAB |
| 21:27.09 | jblack | Her mom actually got thrown out of a plasma-selling clinic for being too nasty. |
| 21:27.11 | mvanbaak | SteveTotaro: you following asterisk-users ? |
| 21:27.30 | mvanbaak | mailinglist that is ? |
| 21:27.34 | SteveTotaro | i have been trying to fix this realtime issue for a customer |
| 21:27.38 | *** join/#asterisk tristanbob_ (n=tristanr@oalug/member/tristanbob) |
| 21:27.41 | SteveTotaro | why, am i being bashed or something? |
| 21:28.09 | jblack | Who would bash you? Surely you jest. |
| 21:28.12 | mvanbaak | SteveTotaro: no, but it shows I was right about calls being pushed through asterisk |
| 21:28.20 | Olobola | I'm trying to get Cepstral installed, tts: http://pastebin.com/d5a439fd |
| 21:29.30 | mvanbaak | SteveTotaro: |
| 21:29.32 | mvanbaak | By the way, I have a client with a four-core Xeon box doing SIP to IAX |
| 21:29.37 | mvanbaak | conversion - that box can handle 1000 concurrent calls. |
| 21:29.38 | jblack | Olobola: I don't know how much luck you'll get with cepstral here. Ethics asside, the only people that can help you are those that have purchased a license. |
| 21:30.01 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-48-245.pskn.east.verizon.net) |
| 21:30.46 | SteveTotaro | cepstral works fine without purchasing, it just nags you |
| 21:30.51 | jblack | Oh look. Richard Branson has a new toy. http://www.blogsmithmedia.com/www.engadget.com/media/2008/01/ssttop.jpg |
| 21:30.56 | Olobola | can I install from any directory? I just created a random folder and tried installing from there. |
| 21:31.14 | jblack | SteveTotaro: Oh, ok. Well, perhaps you'll be willing to help him then. =) |
| 21:31.20 | grandpapadot | SteveTotaro: And you can use sox to trim off the nag audio automatically.. |
| 21:31.24 | *** join/#asterisk Kimitaka (n=swiceje@cpe-065-184-219-014.ec.res.rr.com) |
| 21:32.19 | SteveTotaro | there yougo |
| 21:32.20 | defsdoor | if I run ztmonitor 4 -v and call the line my plugged into the sangoma a200 I see audio level change on RX - but it doesn't seem to register as a call - am I missing something in wanpipe1.conf perhaps for uk lines (anyone know?) |
| 21:32.55 | SteveTotaro | i am getting paid a premium right now |
| 21:33.05 | SteveTotaro | if he will pay more then sure |
| 21:33.08 | Kimitaka | When running asterisk on a machine that's doing nat and has a real ip and hobbyist ip, is there something I need to add to sip.conf so that it knows all the traffic for that account should take place on the outside ip? |
| 21:33.26 | SteveTotaro | ~nat |
| 21:33.27 | jbot | well, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:33.59 | mvanbaak | ~sipnat |
| 21:33.59 | 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 |
| 21:34.49 | Kimitaka | I have asterisk on an openwrt router and sipphone.com is showing it registered but it's not answering the calls |
| 21:35.17 | Kimitaka | on the s extension that is, sipphone and that are both in incoming context |
| 21:36.00 | *** join/#asterisk mmmouse (n=mmouse@office.haefft.de) |
| 21:37.28 | grandpapadot | Hey TK, sorry to bug you again but I'm coming up dry on logging the queue.log to mysql. Clearly, i could do something clever like import it ever hour, chase it wil tail and a script, etc. But I'd prefer to just have asterisk insert the records. I found a couple patches and some info for 1.4, can you point me to the 1.2 info you described? I really appreciate your help. |
| 21:37.31 | puppet | tomorrow im getitng a N95 |
| 21:38.43 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 21:40.37 | SteveTotaro | people tend to over state things, funny how they run exactly 1,000 calls |
| 21:41.35 | SteveTotaro | everyday i get on I95 |
| 21:42.01 | outtolunc | "yeah, when i was testing... i made 3571 calls and my averages were funcky" <G> |
| 21:42.48 | mmmouse | Hi! I'm using zaptel/libpri with a digium TE220 PRI card. Is there any way to get a trace of d-channel communications? |
| 21:44.37 | errr | when I am getting a call in on my zap trunk is there a global variable of some kind that will contain the DID they dialed to get in? |
| 21:45.00 | mvanbaak | ${EXTEN} |
| 21:45.31 | errr | hm |
| 21:46.06 | errr | I didnt think of that. thanks |
| 21:47.04 | mvanbaak | :) |
| 21:47.13 | mvanbaak | that's 10 euro |
| 21:47.27 | nhuisman_work | are you folks' international patterns something like "_011X." ? |
| 21:49.00 | mvanbaak | no |
| 21:49.14 | mvanbaak | our ITSP needs calls in E164 format |
| 21:49.29 | *** join/#asterisk krondorl (n=chatzill@207.245.216.9) |
| 21:49.36 | mvanbaak | we do _31XXXX for dutch calls |
| 21:49.37 | krondorl | Hello! |
| 21:49.42 | nhuisman_work | hmm |
| 21:49.48 | nhuisman_work | i'm in us |
| 21:49.50 | mvanbaak | and _<countrycode><area><number> |
| 21:49.58 | SteveTotaro | outtolunc, do pri intense debug or whatever it is |
| 21:50.15 | nhuisman_work | i'm just wondering how to catch all country codes at once |
| 21:50.27 | nhuisman_work | i've seen a few other configs like "_011X." |
| 21:50.35 | SteveTotaro | pri intense debug span N |
| 21:52.33 | bhima | So far I haven't had any luck figuring out how to show a person's extension as in-use if they are on a single call but still allow incoming calls to ring to them... |
| 21:52.57 | mvanbaak | bhima: what asterisk version ? |
| 21:53.04 | *** part/#asterisk Kimitaka (n=swiceje@cpe-065-184-219-014.ec.res.rr.com) |
| 21:53.32 | SteveTotaro | time to see if Torchwood is half as good as the Dr. Who? |
| 21:53.57 | mvanbaak | it isn't |
| 21:54.11 | SteveTotaro | not even half? |
| 21:54.41 | SteveTotaro | i have the first three dvds of season one |
| 21:54.46 | bhima | mvanbaak: 1.4.17 |
| 21:54.57 | *** join/#asterisk jakenl (n=jake@gryphon.xs4all.nl) |
| 21:55.13 | nhuisman_work | you need 3 dvds for a single season |
| 21:55.16 | jakenl | Anyone using/used VoiceOne ? |
| 21:55.17 | nhuisman_work | wtf format is this in |
| 21:55.18 | nhuisman_work | hd? |
| 21:55.42 | SteveTotaro | six dvds for the season |
| 21:55.42 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:55.54 | nhuisman_work | long season I guess |
| 21:55.56 | SteveTotaro | i just wonder if they are worth "backing up" |
| 21:56.03 | nhuisman_work | could watch it first |
| 21:56.13 | SteveTotaro | each one is an hour |
| 21:56.28 | SteveTotaro | so three episodes fill a dvd |
| 21:57.03 | nhuisman_work | oh not half hour episodes |
| 21:57.20 | krondorl | is anyone have issues where phones randomly register with the * box? |
| 21:57.31 | SteveTotaro | plus they probably have alot of bonus features and junk |
| 21:57.47 | drmessano | krondorl: All my phones randomly register |
| 21:57.52 | mvanbaak | bhima: add a call limit to the sip peers |
| 21:57.56 | SteveTotaro | randomly like at a regular interval |
| 21:58.01 | mvanbaak | doesn't matter at what you set it |
| 21:58.44 | mvanbaak | call-limit=99 |
| 21:58.48 | mvanbaak | something like that |
| 21:58.58 | krondorl | drmessano: That''s not what I mean.. They will be properly registered for a few hours then just no longer register and get "no service".. Then sometime late may get connected again.. |
| 21:59.01 | mvanbaak | bhima: read this: http://www.voip-forum.com/asterisk/2008-01/sip-subscriptions/#more-217 |
| 21:59.42 | drmessano | Are they external? |
| 21:59.45 | drmessano | or internal |
| 21:59.47 | drmessano | or both |
| 22:00.03 | nhuisman_work | if I don't care what a user dials is it bad to just make the outgoing pattern "." |
| 22:00.13 | mvanbaak | yup |
| 22:00.18 | krondorl | internal.. it's only a couple of phones... we have 26 phones.. All the externals are fine... |
| 22:00.23 | mvanbaak | should at least use _X. |
| 22:00.30 | nhuisman_work | err yeah _X. |
| 22:00.44 | drmessano | krondorl: How many phones is it? |
| 22:01.00 | SteveTotaro | do you have qualify set? |
| 22:01.01 | nhuisman_work | are they any issues with doing that? |
| 22:01.07 | nhuisman_work | besides it lets everyone dial wherever they want |
| 22:01.26 | jblack | nhuisman_work: With _X. ? Other than being really permissive, no. |
| 22:01.26 | krondorl | 4 phones are acting up.. One of them is brand new out of the box. |
| 22:01.28 | mvanbaak | besides that, no issues |
| 22:01.56 | nhuisman_work | it just seems like there is a lot of outgoing rule I will have to make |
| 22:02.04 | jblack | nhuisman_work: With _. ? Definitely. It does bad things to the s, i and t extensions, among other things. |
| 22:02.05 | drmessano | Maybe connectivity issues.. |
| 22:02.15 | bhima | mvanbaak: Reading and poking at config, thanks. |
| 22:02.19 | nhuisman_work | yeah i'll stick with _X. if I do it. |
| 22:02.25 | drmessano | Hmm |
| 22:02.26 | drmessano | NM |
| 22:02.30 | krondorl | We checked that.. The computers that are connected to the same line never get inturrputed.. |
| 22:02.31 | drmessano | Wouldnt be just 4 specific |
| 22:02.31 | nhuisman_work | do you guys forward *XX stuff? |
| 22:02.34 | nhuisman_work | like *69? |
| 22:02.38 | jblack | There's a page on the wiki that can explain in more detail as to why _. is a disaster. |
| 22:02.45 | *** join/#asterisk atisss (n=atisss@193.238.212.171) |
| 22:02.55 | drmessano | What kind of phones, krondorl? |
| 22:03.07 | krondorl | Aastra 9133i. |
| 22:03.52 | jblack | nhuisman_work: I don't have a pstn (zapata) line, so those services don't exist for me (I use voip instead, as it's much, much cheaper). Instead, I provide them interally |
| 22:04.10 | defsdoor | krondorl: daisy chained ? |
| 22:04.10 | nhuisman_work | hmm |
| 22:04.39 | drmessano | krondorl, Do they work plugged in somewhere else? |
| 22:04.41 | krondorl | All phones go directly to a switch.. Computers plug into phone. |
| 22:04.42 | nhuisman_work | how do you match *, just put it in there ? |
| 22:04.43 | defsdoor | krondorl: I use 9133i and have had no problems so far - 40 or so extensions |
| 22:04.58 | jblack | I don't have *69 setup, but I do have 3 way calling, call waiting, call forwarding, etc etc. |
| 22:05.08 | jblack | nhuisman_work: Yup. |
| 22:05.11 | nhuisman_work | i found a nice page with some nice macros |
| 22:05.20 | nhuisman_work | http://www.nslu2-linux.org/wiki/Optware/AsteriskStdextenMacro |
| 22:05.20 | krondorl | Doesn't matter where they are plugged in.. they lose service sooner or later.. |
| 22:05.22 | nhuisman_work | was going to add these |
| 22:05.26 | jblack | You may or may not need them. |
| 22:05.26 | nhuisman_work | let the users setup their own forwards |
| 22:05.29 | nhuisman_work | with *XX numbers |
| 22:05.32 | jblack | Are you using an ATA? |
| 22:05.37 | defsdoor | krondorl: network congestion ? |
| 22:06.02 | krondorl | defsdoor: But then why only those 4 and none of the others? |
| 22:06.16 | defsdoor | krondorl: do the pc users copy 5GB files around a lot ? |
| 22:06.41 | krondorl | Nope.. Mater of fact the one computer is not used all that much.. |
| 22:06.53 | jblack | nhuisman_work: Are you using an ATA? |
| 22:07.18 | jblack | damnit. I'm on hold with verizon, and it's 17:06. I bet I made it into their queue right before everyone switched off the lights and went hom. |
| 22:08.00 | jblack | nhuisman_work: do you know what an ATA is? |
| 22:09.45 | drmessano | Thats the hard drive with all the pins |
| 22:09.48 | drmessano | Right? |
| 22:09.58 | drmessano | :D |
| 22:11.14 | krondorl | if I do a hard restart (pull power) it will connect sometimes.... |
| 22:11.50 | [TK]D-Fender | jblack, I've been on hold with HP for over an hour before. |
| 22:12.34 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 22:12.42 | krondorl | defsdoor: What version you running?? We are using 1.4.11 |
| 22:12.44 | drmessano | I hate HP |
| 22:12.50 | drmessano | HATE HATE |
| 22:12.58 | Nugget | Torque is more important than HP. ;) |
| 22:13.02 | drmessano | lol |
| 22:13.33 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 22:13.39 | drmessano | HPs network managent on Windows knows nothing but how to team NICs and it never will |
| 22:13.47 | drmessano | So they can kiss my...... HAPPY THOUGHT |
| 22:13.54 | drmessano | ahem.. better now |
| 22:14.07 | defsdoor | krondorl: 1.2.17 on one site |
| 22:14.09 | drmessano | s/whateverthatwas/management |
| 22:14.46 | defsdoor | krondorl: I have 20 9133i to put in on a 1.14.17 shortly (if I can ever get the a200 working) |
| 22:15.12 | drmessano | The trick to buying HP is always buy an extra server.. then you can always rob drives later when they become extinct.. lol |
| 22:15.14 | defsdoor | krondorl: 1.2.23 on the other site |
| 22:16.01 | grimsy | is there a CLI command I can use to list all defined extensions? |
| 22:16.15 | krondorl | defsdoor: lol.. It just all of a sudden started doing this to us.. Even with a brand new phone... (now 1 week old) Two of the phones even have firmware .3000 while the others are still at .1081.. |
| 22:17.32 | [TK]D-Fender | grimsy, "dialplan show". |
| 22:17.40 | krondorl | Well, I gotta go home now.. I'll try again tomorrow morning, maybe a different crowd might be on and be able to help. Night all.. |
| 22:17.50 | [TK]D-Fender | grimsy, or you could just look at your extensions.conf like the rest of us. |
| 22:18.37 | grimsy | [TK]D-Fender: thanks :) |
| 22:18.52 | grimsy | looking at extensions.conf doesn't give me a list of iax/sip users though |
| 22:19.19 | [TK]D-Fender | grimsy, Nor should it |
| 22:19.32 | grimsy | i'm trying to write a script to find an unused extension within a range |
| 22:19.42 | grimsy | is there a better way to do it? |
| 22:20.06 | [TK]D-Fender | grimsy, And you nned to stop mixing up "extension" and "SIP/IAX/other device" |
| 22:20.19 | [TK]D-Fender | grimsy, extensions = extensions.conf |
| 22:20.48 | [TK]D-Fender | grimsy, if you want a summary of DEVICES thats another matter. |
| 22:20.50 | grimsy | true, ok. i guess in my case, all my users are numbers rather than names, so i think of them as extensions, sorry |
| 22:20.56 | [TK]D-Fender | grimsy, "sip show users" |
| 22:21.10 | [TK]D-Fender | grimsy, good time to STOP that thought |
| 22:22.09 | grimsy | [TK]D-Fender: i'm using sip show users now, however our system also has some conference rooms set up within the same range, and they don't show up in the sip show users (naturally) |
| 22:22.35 | [TK]D-Fender | grimsy, What does a SIP Device have to do with a "conference room"? |
| 22:22.37 | grimsy | dialplan show will probably do the trick though |
| 22:22.44 | grimsy | it doesn't, hence naturally |
| 22:22.49 | [TK]D-Fender | grimsy, Got some more apples & oranges to compare? |
| 22:23.00 | defsdoor | he means internal diallable numbers :) |
| 22:23.19 | [TK]D-Fender | defsdoor, Which would lead back to "read your extensions.conf" |
| 22:23.43 | [TK]D-Fender | grimsy, "help meetme" |
| 22:24.04 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:24.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:24.48 | grimsy | thanks [TK]D-Fender, i wasn't asking about meetme, your original answer with dialplan show will work, as it lists everything i need |
| 22:25.33 | [TK]D-Fender | grimsy> [TK]D-Fender: i'm using sip show users now, however our system also has some conference rooms set up within the same range, and they don't show up in the sip show users (naturally) <-- |
| 22:25.40 | km- | any ideas: |
| 22:25.41 | km- | Jan 23 17:24:58 NOTICE[13796]: channel.c:1954 ast_read: Dropping incompatible voice frame on Local/917273484130@local-b157,2 of format ulaw since our native format has changed to slin |
| 22:25.57 | [TK]D-Fender | grimsy, well you say you want "A" and then no, you want "B" and now you're back to "A" again. make up your mind. |
| 22:26.04 | km- | the phone call was fine then wham, started gettign choppy. |
| 22:26.09 | *** part/#asterisk fiXXXerMet (n=kjohnson@dsl092-156-002.wdc2.dsl.speakeasy.net) |
| 22:26.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:28.44 | *** join/#asterisk ghento (n=ghento@75.157.224.194) |
| 22:28.47 | ghento | join #symfony |
| 22:28.57 | ghento | whoops :) |
| 22:34.25 | Olobola | so cepstral tts voice is choppy.. any suggestions? CPU usage is <10% |
| 22:38.36 | drmessano | flite |
| 22:40.18 | keith4 | isn't cepstral non-free? |
| 22:42.25 | mvanbaak | some of it isn't |
| 22:42.44 | hmmhesays | seems as though the latest iksemel is not compatible with the latest gnutls |
| 22:46.10 | mvanbaak | the debian and openbsd packages are ;) |
| 22:48.23 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 22:49.17 | hmmhesays | time to see if 1.6 is |
| 22:49.18 | hmmhesays | I hope |
| 22:50.19 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 22:50.50 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 22:51.39 | mvanbaak | latero all |
| 22:59.52 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:00.33 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583384.dsl.bell.ca) |
| 23:04.06 | defsdoor | has anyone got any clues for my a200d problems before I go to bed ? Asterisk is passing calls over to it as I see the TX gain change, and the RX change on incoming - it's as if the card isn't working properly on a BT analog line |
| 23:06.19 | *** join/#asterisk BipBip (n=BipBip@bl8-131-248.dsl.telepac.pt) |
| 23:07.01 | [TK]D-Fender | mvanbaak, Hey got a situation perhaps you can help me with : What combination of settings should I use for Cisco's behind a remote NAT? Asterisk (public IP) > internet > NAT > Cisco 7940 |
| 23:07.02 | jblack | SteveTotaro: Ping |
| 23:12.37 | SteveTotaro | pong |
| 23:13.16 | jblack | SteveTotaro: Hey... I was looking back at an earlier conversation that you had about grandcentral, and you said that they support sip. |
| 23:13.34 | SteveTotaro | no, i don't think so |
| 23:13.39 | jblack | I've really appreciate it if you could provide me more details, because I'd love that |
| 23:13.56 | jblack | Oh, you didn't? |
| 23:14.03 | SteveTotaro | one second, i will give you a thread on a forum that will explain everything |
| 23:14.15 | jblack | Nah. you're right. I misread what you said. |
| 23:14.20 | SteveTotaro | maybe they do, i have yet to set it up |
| 23:14.45 | SteveTotaro | http://www.fatwallet.com/t/18/758191/ |
| 23:15.16 | jblack | Aww. I don't have a sunrocket gizmo. |
| 23:15.26 | SteveTotaro | i don't think you need one |
| 23:16.11 | jblack | I set gc up.. yesterday I think. I was able to get them to forward their DID to a free IP Kall did. |
| 23:16.15 | jblack | Trivially. |
| 23:16.34 | jblack | So I was able to set up the inbound with no problem. |
| 23:16.40 | SteveTotaro | http://voipcalls4free.blogspot.com/ |
| 23:16.46 | jblack | How do I set up the free outbound calls to DIDs? |
| 23:17.06 | SteveTotaro | i guess you use click2dial on grandcentral |
| 23:17.14 | SteveTotaro | that is the piece i am not sure of |
| 23:17.30 | SteveTotaro | everyone says they are doing it but i personally have not yet |
| 23:17.58 | SteveTotaro | so if you figure it out and get it working, please let me know |
| 23:18.05 | jblack | That click2call stuff doesn't work for me. I'm restrictive in the software that I allow. |
| 23:18.18 | _charly_ | does anyone here have a h323 trunk from asterisk to a siemens hipath3000? |
| 23:18.27 | SteveTotaro | is there a box that you can try it on somewhere? |
| 23:18.42 | SteveTotaro | i have a box that i format every month or so |
| 23:19.13 | SteveTotaro | anyways, that's what i have, gonna get back to watching torchwood |
| 23:19.25 | SteveTotaro | if you do anything with it, let me know |
| 23:19.32 | SteveTotaro | please |
| 23:20.03 | jblack | Avoiding things like "Monthly reinstalls" is part of the reason I'm restrictive on proprietary software. I don't feel keen on handing over access to machines that contain crucially private data (bank account numbers, my gpg key, etc) to third parties with inauditable code. |
| 23:20.13 | [TK]D-Fender | SteveTotaro, strangley thats EXACTLY what I'm doing now... |
| 23:20.23 | [TK]D-Fender | SteveTotaro, Torchwood.2x01.Kiss Kiss Bang Bang.avi |
| 23:20.34 | [TK]D-Fender | strangely* |
| 23:27.24 | *** join/#asterisk sniperd (n=josh@cp120.cardplayer.com) |
| 23:28.08 | sniperd | I am having a wierd problem that im wondering if anyone has heard before, when calling softphone to softphone it works fine, and when calling from a real phone to realphone, and realphone to softphone it works fine |
| 23:28.31 | sniperd | but calling from softphone to realphone, it automatically goes to voicemail, if voicemail is enabled, if voicemail isnt enabled it works. |
| 23:28.48 | sniperd | im using trixbox 2.2.9 |
| 23:28.57 | [TK]D-Fender | sniperd, Trixbox is NOT supported here. |
| 23:29.06 | [TK]D-Fender | sniperd, go ask in their channel(s) |
| 23:29.11 | [TK]D-Fender | ~trixbox |
| 23:29.11 | jbot | extra, extra, read all about it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 23:29.33 | [TK]D-Fender | ~freepbx |
| 23:29.33 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:29.47 | sniperd | ok thanks, no need to get yourself all worked up into a heart attack man |
| 23:30.08 | *** part/#asterisk sniperd (n=josh@cp120.cardplayer.com) |
| 23:30.10 | [TK]D-Fender | sniperd, just giving you both sets of support links |
| 23:35.01 | nhuisman_work | what format list for your voicemail do you guys use? |
| 23:35.11 | nhuisman_work | just wondering if I should change it from wav49|gsm|wav |
| 23:35.14 | nhuisman_work | or if that is ok |
| 23:35.42 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:41.05 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 23:41.32 | *** part/#asterisk DrkShadow_ (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 23:41.44 | *** join/#asterisk DrkShadow_ (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 23:42.05 | DrkShadow_ | so.. I have the asterisk server behind NAT, and I _can_ call in to voicemail.. but.. I don't get any sound from the server. Ideas? |
| 23:42.53 | DrkShadow_ | port 5060 is forwarded to the server.. but that's all. |
| 23:42.57 | jblack | DrkShadow_: You are confusing me. |
| 23:43.14 | jblack | Connecting to voicemail works, but sound does not, correct? |
| 23:43.18 | jblack | sounds like it. |
| 23:43.23 | DrkShadow_ | correct.. I get no sound. |
| 23:43.24 | jblack | SIP needs a lot more ports than just 5060 |
| 23:43.32 | nhuisman_work | sounds like no sound |
| 23:43.34 | DrkShadow_ | such as? is there a page? |
| 23:43.37 | jblack | 5060 is the channel that brokers connections. |
| 23:44.11 | jblack | There's probably a few pages about, but what you want to know is actually listed in your rtp.conf |
| 23:44.38 | DrkShadow_ | hmm. Starts at 10k, ends at 20. ok... |
| 23:44.41 | jblack | You'll see a range of udp listed. |
| 23:44.54 | jblack | Yeah. You can cut those back if you wish. I only define 50. |
| 23:45.59 | jblack | ~ipnat |
| 23:46.03 | jblack | ~sipnat |
| 23:46.04 | 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 |
| 23:46.09 | jblack | DrkShadow_: There's docs for you |
| 23:46.15 | DrkShadow_ | ty |
| 23:46.35 | nhuisman_work | how many formats do you folks save your voicemail iN? |
| 23:46.50 | jblack | Just one. |
| 23:46.54 | nhuisman_work | wav49? |
| 23:47.12 | jblack | wav49 is the default here. |
| 23:47.40 | nhuisman_work | the sound quality is acceptable I assume, i'll try it. |
| 23:48.12 | jblack | Grin. Yeah, I generally recommend people try things before they change them --- especially steak. |
| 23:48.31 | nhuisman_work | heh |
| 23:48.44 | nhuisman_work | steak = salt, pepper, and optionally rubbed in minced garlic |
| 23:48.52 | nhuisman_work | a1 sauce/bbq sauce = not steak |
| 23:48.54 | jblack | That sounds good. |
| 23:49.08 | nhuisman_work | preferably rock salt |
| 23:49.10 | DrkShadow_ | hmm. I do steak a bit differently.. some orange peal is _great_ |
| 23:49.12 | nhuisman_work | and fresh cracked pepper |
| 23:49.15 | jblack | Heh. A1 has come to my rescue from time to time with really low quality steak. |
| 23:49.31 | DrkShadow_ | A1 is... I dunno. I kinda used to like it, but I think it's too sour now. |
| 23:49.34 | jblack | Such as comes with breakfast. |
| 23:49.40 | nhuisman_work | if i'm buying below tri-tip i mostly don't do a normal steak preperation |
| 23:49.40 | tzanger | I never liked A1 much |
| 23:49.47 | nhuisman_work | sometimes chuck can be ok |
| 23:49.54 | nhuisman_work | cook it fast and keep it medium rare |
| 23:50.06 | nhuisman_work | round steak = braise it |
| 23:50.15 | nhuisman_work | do swiss style steak |
| 23:51.11 | nhuisman_work | I use worcestershire sauce, but in marinades with other stuff for chicken and other cooking applications. |
| 23:51.33 | nhuisman_work | i guess I better join #asterisk-cooking before my ranting pisses people off :P |
| 23:51.52 | jblack | Nah. It's fine as long as nobody's looking for help |
| 23:52.08 | nhuisman_work | cooking is my hobby |
| 23:52.20 | jblack | This is the ice cream reward(random chatter) for eating one's veggies (helping others) |
| 23:54.03 | jblack | Oh, lesson learned today... |
| 23:54.33 | jblack | Dial(${lblack}&${kitchen}&${kitchen}) has unfavorable results. =) |
| 23:55.42 | nhuisman_work | question: what does the & do? |
| 23:55.56 | jblack | When you dial multiple numbers at once, the first one to pick up wins. |
| 23:55.57 | nhuisman_work | is that like / |
| 23:56.00 | nhuisman_work | oh |
| 23:56.12 | nhuisman_work | so putting kitchen twice isn't good? |
| 23:56.33 | jblack | Certainly not bright. |
| 23:56.48 | nhuisman_work | what does it do? |
| 23:56.51 | jblack | I saw odd things here. Like call waiting indications after the phone call was initiated. |
| 23:56.53 | nhuisman_work | just not dial at all? |
| 23:57.43 | jblack | I'm not quite sure all the wierd things that happened. I know my ex-wife got pissed after trying to call the kid a few times. |
| 23:58.21 | jblack | I say it fell well into undefined land |
| 23:59.10 | nhuisman_work | what a useless option |
| 23:59.13 | nhuisman_work | pbxskip in voicemail.conf |
| 23:59.39 | jblack | Pardon? |
| 23:59.49 | nhuisman_work | seriously who thought up that hack |
