| 00:01.55 | RobZombie | I just wish I was stronger with going from a blank system to installing linux and then asteriska and basic configuration. If I had that, I'd have no issues... But my friends that know all this, that i have worked on there boxes for ages, all either too busy with school or work, millitary or something. Leaving me reading,and not getting too far. |
| 00:02.48 | WIMPy | It's only the installation. |
| 00:02.58 | RobZombie | And this ubuntu desktop i installed on the netbook is cool, but I am not learning anything from it. |
| 00:03.13 | WIMPy | The configuration will never be for the faint hearted, no matter what package or distro. |
| 00:03.58 | RobZombie | I'm clueless, if I had a link to an is id slap it on a usb right now installt he linux version, like my friends use gentoo, but id use whatever, and the try for the asterisk all manually, but, im just lost. LOL |
| 00:05.05 | RobZombie | Guess my path of thinking was off. LOL.. I figured its the 4th of july, if I find someone online today, they gotta be a brain, and can pull me thru, lol. |
| 00:05.31 | RobZombie | Honestly, I have everything setup and running now onthe server with elastix, problem is I dont have a gui database solution. |
| 00:05.35 | WIMPy | I't the 5th already here :-) |
| 00:05.45 | RobZombie | Hehehe, right. |
| 00:05.57 | WIMPy | Uhh, elastix. |
| 00:06.02 | WIMPy | ~elastix |
| 00:06.03 | infobot | well, elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 00:06.06 | RobZombie | I'm a day late and dollar short as usual. |
| 00:06.31 | WIMPy | So that might not be the best choice. |
| 00:07.05 | RobZombie | Like i said I only went with elastix because it had everything asterisk needed to run from the get go, other than that I dont even use the elastix gui for nothing but backing up and restoring. |
| 00:07.07 | WIMPy | At least FreePBX has some sort of support. But not in here, either. |
| 00:07.33 | RobZombie | I want a asterisk only solution, was my point of coming here |
| 00:07.40 | RobZombie | been trying to get asterisknow in on the usb |
| 00:07.43 | RobZombie | no dice. |
| 00:07.52 | RobZombie | as i mentioned earlier and now i know thats not gonna work |
| 00:07.55 | RobZombie | im stuck again |
| 00:08.48 | RobZombie | What about trixbox? it have a ready to run solution with gui database support preinstalled? |
| 00:09.42 | WIMPy | I think your best bet is AsteriskNOW. |
| 00:09.47 | WIMPy | ~trixbox |
| 00:09.47 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 00:10.03 | WIMPy | Always the same. |
| 00:10.09 | RobZombie | Thats what I thought, guess I wont be doing no work today as I had hoped, ill have to get usb-cd first to get asterisknow installed |
| 00:11.06 | RobZombie | Figured after seeing 2 + yrs of usb install issues with asterisknow I'd find a solution here, guess they figured, buy a cd. LOL |
| 00:12.42 | RobZombie | I'm about ready to go with a pay support solution, lol, literally, killing me over here. |
| 00:12.46 | TechSmurf | Trix is dead |
| 00:12.51 | RobZombie | Yeah |
| 00:14.29 | RobZombie | thanx everyone. Happy 4th to some and belated 4th to others. |
| 00:14.35 | *** part/#asterisk RobZombie (cce48afa@gateway/web/freenode/ip.204.228.138.250) |
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| 02:17.25 | fling | hello! |
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| 03:56.42 | adeel | anyone happen to know how often * will do an SRV lookup? |
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| 04:37.19 | carrar | moo |
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| 04:57.26 | mcf3782 | I am creating MeetMe conference rooms dynamically on the fly for some users. They're prompted to enter a five digit number for the conference they want to create. I want to check to see if the number the user enters, is already an active conference. Something like 'asterisk -x "meetme list concise" | cut -d \! -f 1 |grep $ROOM_NUM'. Is there a way to do that from within the dialplan? |
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| 05:38.26 | ChannelZ | mcf3782: try looking at the MEETME_INFO function |
| 05:38.45 | Jo_Bell | From the SIP RFC 3665, I understand that to implement "registration" with asterisk, a SIP client needs to send first a registration, followed by another registration with encrypted user information. Here, what is the type of the encryption that Asterisk is looking for? |
| 05:39.01 | ChannelZ | or MeetMeCount() application |
| 05:39.56 | ChannelZ | it's just an MD5 digest I think |
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| 05:42.37 | mcf3782 | ChannelZ: I hadn't seen MEETME_INFO yet. Thanks for that pointer. I haven't found a way of calling MeetMeCount() that gives me the data I want; but I'll look at it again. Thanks. |
| 05:44.30 | ChannelZ | Reading the help MeetMeCount looks a bit wacky so MEETME_INFO is probably better anyway |
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| 05:52.54 | mcf3782 | The way I interpret the help info for MEETME_INFO, it appears to be for getting more information about a specific conference 'confno'. I don't really need any detailed information about a conference number. I just want to know if it exists, so I can tell the user that the conference number they entered (i.e. 56789) is already in use; and prompt them to pick a different number. |
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| 05:55.20 | Beltechs | is it possible to format a callerid to be out put like (xxx) xxx-xxxx ? |
| 05:56.20 | mcf3782 | I'll tackle this again after some sleep. My eyes are going crossed. |-) |
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| 05:58.05 | Jo_Bell | Channelz: Fine. Is there any document which mentions the exact parameter list of individual commands to Astersisk, the possible values etc? Such a document will be handy for a SIP client development. The RFC's just say Registration with User details. Nothing is mentioned about the header values and the payload structure. |
| 06:00.09 | ChannelZ | I assume there are probably hundreds of docs on the net, search for "sip digest authentication" or something |
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| 06:00.52 | ChannelZ | http://alexkr.com/memos/66/digest-authorization-in-sip-with-md5/ |
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| 06:07.29 | v0lZy | lo |
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| 06:13.04 | Jo_Bell | ChannelZ: thank you. This serves the purpose for SIP MD5 authentication. Since there are a large number of SIP commands/responses possible, I would like to know if there is a single document covering the structure of all the commands to the Asterisk and the responses back to the SIP clients. Ex: Invite F1 - the client developer needs to know How to construct the Invite payload. Is there |
| 06:13.05 | Jo_Bell | anything serving this? Else, we have to see how exactly the Asterisk expects individual client commands? |
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| 06:15.57 | din3sh | hello all |
| 06:17.35 | din3sh | exten => _xx,n,Goto(testing,${newexten},1} ==> do i need to have exact matching for ${newexten} in [testing]? |
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| 06:46.06 | ChannelZ | Jo_Bell: the SIP rfcs tell all. SIP is a standard |
| 06:47.18 | ChannelZ | din3sh: never tried but I imagine not, if there are extension patterns that match in the Goto'd context |
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| 07:25.17 | Jo_Bell | ChannelZ: I got what you said. But Does the standard doesn't mention how the From, To, Call-ID, CSeq, Contact etc are packaged together in the Register command? I didn't see any such spec. How does the Asterisk expect the field parameters? Are the fields given like x=y, where x is the name of the field, y is the value. Or any other format? Sorry if my query is primitive. |
| 07:33.20 | wdoekes | Jo_Bell: ngrep and tcpdump and wireshark are your friends |
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| 07:35.12 | _zoom_ | hi, |
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| 07:35.39 | _zoom_ | am looking for call center suite (asterisk-based) that support arabic?/ |
| 07:43.11 | Jo_Bell | wdoekes: fine. If there is no documentation on that, yes wireshark is my friend :-( |
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| 07:55.07 | wdoekes | Jo_Bell: there *is* documentation, but you're not reading it |
| 07:55.10 | bulkorok | hi... I want to receive faxes with ReceiveFax and FFA. but the receivefax is not sending a CED tone at the beginning of the transaction. how can I force this!? |
| 07:55.33 | wdoekes | points to rfc3261 some more.. specifically the abnf portions |
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| 08:18.27 | itsurkg | Dear All, I am trying to play a sound file on outgoing call when all lines are busy. I found CHANUNAVAIL is what i need but haven't got any luck. |
| 08:20.14 | itsurkg | Here is my extension.conf file look like: http://fpaste.org/c6Ik/ |
| 08:23.03 | shadebob | hi. Which softphone support sendurl (not Zoiper biz …) ? |
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| 10:22.26 | MariusKarthaus | Hi, I've set up a queue and agents log in to it using AddQueueMember() from the dialplan. There are multiple agents in the queue. Agents can call eachother by extentions like this: 6,1,dial(SIP/phone6,30). Now we have a problem when a caller is transferred by an agent to another agent. The caller is dropped back into the music-on-hold and is never picked up. Any ideas where I should start to fix this issue? |
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| 10:29.35 | MariusKarthaus | It appears that the agent that is tranferred to, is also put in music-on-hold. |
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| 13:37.27 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 13:37.27 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.1 (2012/06/14), 1.8.13.0 (2012/06/04), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 13:38.23 | philfine | I have just notice that obviously context in lua cannot have '-' sign :D |
| 13:38.42 | philfine | At least without " |
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| 14:01.00 | philfine | 71 |
| 14:01.04 | philfine | oups |
| 14:01.05 | philfine | :S |
| 14:03.53 | Dovid | morning TK |
| 14:04.05 | Dovid | flew over ur home yesterday |
| 14:06.03 | [TK]D-Fender | Thankfully enough cover to limit your approach ;) |
| 14:09.57 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 14:13.19 | philfine | Is there a debug function that we can use in the diaplan |
| 14:13.31 | philfine | Want to print caller id on asterisk log |
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| 14:15.51 | philfine | print is good enough |
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| 14:23.37 | Lantizia | Hey how are you meant to debug callpickup issues? I have a peer 3649 ringing, so 3648 dials *8 and nothing happens (both are in callgroup & pickupgroup 71) |
| 14:24.06 | Lantizia | if I turn on sip debug for 3648 I see it does dial *8@<ip of pbx> in the To: line - but ultimately hangs up |
| 14:24.31 | Lantizia | pickupexten is set to *8 in features.conf |
| 14:25.03 | GeoGeek | We have Ubuntu 12.04 server and asterisk 10 vanilla and want to build an install cd so that we can easily roll this configuration out to our other radio stations. Anyone have suggestions on the best way to do this? |
| 14:25.17 | philfine | How do I access caller id from Lua dial plan |
| 14:26.06 | philfine | Isn't it channel.CALERID(num):get() ? |
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| 14:34.26 | pithagorians | hi. sip trunk provider gets <sip:402:myIP> where 402 is my internal number instead of the number they provided for us. same for Contact and Remote-Party-ID. question - what it the right method to send all of this parameters ? i have only one DDI from them so it's gonna be static for all users. |
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| 14:37.19 | carrar | names |
| 14:37.20 | carrar | err |
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| 14:44.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:50.42 | philfine | Can anyone give me a clue if it is possible to override caller id on a varix business pbx with asterisk. I have to transfer calls through that pbx, but don't know if I am doing everything to override the caller id. |
| 14:51.35 | [TK]D-Fender | phithen loko at your CALL gong to it |
| 14:52.42 | leifmadsen | if the other end doesn't let you control the caller ID there is nothing you can do to force a specific callerID through that system |
| 14:53.17 | philfine | Thats most likely the case right ? |
| 14:53.48 | leifmadsen | if you've set the callerID on your side and are attempting to pass it through, and it doesn't work, then I suspect yes |
| 14:54.05 | leifmadsen | via SIP, you'd want to use sendrpid and try with 'yes' and 'pai' |
| 14:54.13 | leifmadsen | if those don't work, then you're pretty much skuppered |
| 14:54.13 | *** join/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 14:54.16 | philfine | How exactly do I set the caller ID ? |
| 14:54.23 | leifmadsen | using the CALLERID() function |
| 14:54.32 | treborsux | Someone calls in and the operator answers and hears a ringing like shecalled someone. They can hear her she only hears ringing. She puts call on hold and resume and ringing goes away and call is normal from there. |
| 14:54.33 | philfine | I did Dial with "f" on the 3rd parameter |
| 14:54.50 | treborsux | <PROTECTED> |
| 14:55.04 | treborsux | <PROTECTED> |
| 14:55.19 | treborsux | Anyone want to see the log ans see if they can solve this mystery? |
| 14:55.37 | philfine | leifmadsen: So I do it before Dial function ? |
| 14:55.53 | leifmadsen | that's how things work, yes |
| 14:55.58 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:56.13 | philfine | Isn't it suppose to be set initially for the caller ? |
| 14:56.36 | philfine | I will override it |
| 14:57.16 | treborsux | I dont understand why all my systems do this. I am not doing anything special just installing |
| 14:57.56 | treborsux | both my dahdi and pri systems do this |
| 14:58.03 | treborsux | could it be my swiches? |
| 14:58.12 | treborsux | the network? |
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| 15:01.58 | treborsux | Will someone look at this log for me and see if they know whats up? |
| 15:05.52 | mcf3782 | I'll take a stab at it. Where's the log? |
| 15:06.20 | treborsux | its a zip how can i get it to you too big to pastebin |
| 15:06.46 | treborsux | http://pastebin.com/3XGUqJsP here is her discription with times and numbers that were calling this happened on |
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| 15:07.05 | treborsux | i have had sip debug on all and stuff so it is huge |
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| 15:15.31 | treborsux | howw can I get log to you?> |
| 15:15.35 | mcf3782 | From the description, you have one-way audio on some calls. Have you looked at any firewalls and/or NAT devices in the path from end-to-end? That's gonna be tough to debug from just log files. |
| 15:15.52 | treborsux | i have sip debug on all |
| 15:16.11 | treborsux | no the 2 phones hooked up on the same switch poe |
| 15:16.23 | treborsux | are there setting on the switch that could cause this? |
| 15:16.56 | treborsux | why does she heare a ring like she called someone |
| 15:17.19 | treborsux | the pbx is hooked to same switch the phones are |
| 15:17.54 | treborsux | trendnet pos 24 port switch |
| 15:18.27 | treborsux | on the other systems they are linksys/cisco poe switches. Same issues |
| 15:18.44 | treborsux | not everytime just sometimes |
| 15:21.23 | mcf3782 | What's the commonality between times when it happens? Always the same phone, or the same number? Are all the phones the same type? Do the phones themselves have any configuration settings that need to be checked? |
| 15:22.03 | treborsux | polycom 501s and 601s happens on any extension |
| 15:22.18 | treborsux | mostly when they pic up an outside call coming in |
| 15:22.31 | treborsux | put has happened extesion to extension before |
| 15:22.47 | treborsux | happens on all systems i have built both with dahdi and with pri |
| 15:23.20 | treborsux | i cant find anything to connect instances seems to be random |
| 15:23.58 | treborsux | the 501s and 601s are on the highest firmware they are allowed and highes sip software they are allowed |
| 15:24.06 | treborsux | all on same firmware |
| 15:24.09 | treborsux | and sip app |
| 15:24.59 | treborsux | What could I be doing wrong on 4 systems that causes this on all of them |
| 15:27.38 | treborsux | For more information about why Polycom phones don't seem to like type=friend, and to explore a fix for a known Polycom bug, the "One Way Communication" issue, |
| 15:28.01 | treborsux | http://web.archive.org/web/20050926170148/http://www.southwestfcu.org/tech/polycomsip.html fender seemed to think this was unrelated |
| 15:29.32 | treborsux | I dont understand what this document wants me to change to fix it? |
| 15:29.45 | treborsux | or if I should try to do what it says? |
| 15:31.09 | treborsux | I cant tell if that document solved this issue or what? |
| 15:31.30 | treborsux | mcf3782 does that document make sense to you? |
| 15:32.37 | treborsux | only thing I could find describing something close to my syptoms |
| 15:33.02 | treborsux | Do you want to look at log since i have the times and numbers it happened on? |
| 15:35.07 | treborsux | should i make the changes he suggests at the top of the ducument to sip.conf and extension.conf? |
| 15:35.50 | mcf3782 | That's a long article. Looks like it documents and fixes several problems that particular user was having. It may or may not be relevant to your situation. I'd print off some sections of your configs, and compare them to that article, to see if there are similarities and try the config changes that person did. Just document what you change before you change it; so you can put it back if it doesn't help. |
| 15:38.23 | treborsux | type=friend is what he acts like is the issue |
| 15:38.38 | treborsux | but freep[bx will just make it back that way when i make a change? |
| 15:39.25 | pabelanger | Find out who is generating the ringing, if setup properly, then it will be the polycom phone which means it is their issue. If you are using asterisk to generate audio inband, then that is the issue too |
| 15:39.27 | Qwell | treborsux: #freepbx |
| 15:39.51 | pabelanger | FFIW: I had an issue with a polycom 331 and ringing recently, upgrading to their latest firmwares fixed the problem |
| 15:40.09 | *** join/#asterisk shaiton (~shaiton@fedora/shaiton) |
| 15:40.47 | treborsux | i dont get what type=friend is |
| 15:41.44 | mcf3782 | http://www.voip-info.org/wiki/view/Asterisk+sip+type |
| 15:42.05 | pabelanger | ~book |
| 15:42.06 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:42.13 | pabelanger | treborsux, ^ |
| 15:42.43 | Lantizia | callpickup question bump? |
| 15:44.54 | mcf3782 | I am creating MeetMe conference rooms dynamically on the fly for some users. They're prompted to enter a five digit number for the conference they want to create. I want to check to see if the number the user enters, is already an active conference. Something like 'asterisk -x "meetme list concise" | cut -d \! -f 1 |grep $ROOM_NUM'. Is there a way to do that from within the dialplan? |
| 15:47.57 | treborsux | only thing i can find about my issue suggest type=friend is not usable forpolycom 501 and 601s which would mean freepbx cant be used with them to make extensionsso this does not seem to be an issue since others are using them with freepbx so why is this happening |
| 15:48.20 | treborsux | is anyone else using 501s or 601s with asterisk on type=friend |
| 15:48.34 | Qwell | treborsux: Pick a channel and discuss your issue there. |
| 15:49.06 | Qwell | Hint: I picked #freepbx for you. |
| 15:49.08 | treborsux | i was trying to get the asterisk side and the freepbx saide |
| 15:49.23 | Qwell | There is no Asterisk side. You are not configuring Asterisk. You are configuring FreePBX. |
| 15:49.34 | treborsux | since the only thing i can find has to do with manually editing sip.conf |
| 15:49.57 | Qwell | If you would actually pay attention in the channels you've asked your question, you'd see that you were given an answer. |
| 15:50.18 | Qwell | This is why it's incredibly rude. |
| 15:50.21 | treborsux | fI am an idiot |
| 15:55.05 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-127.mobile.uci.edu) |
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| 16:03.05 | *** join/#asterisk aviationprinc3ss (aviationpr@calpolyswift.org) |
| 16:17.58 | *** join/#asterisk RubyRails (~jaiken@209.33.214.243) |
| 16:24.51 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 16:27.17 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:30.14 | mcf3782 | Well. That's annoying. My 'Definitive Guide' book showed up today. From page 151 - page 217, the pages have been cut diagonally from top left to bottom right. The cut sections are just not there. |
| 16:31.00 | Qwell | cut diagonally? |
| 16:31.04 | mcf3782 | yep |
| 16:31.05 | Qwell | like, half the page is just missing? |
| 16:31.10 | mcf3782 | yep |
| 16:31.13 | Qwell | awesome |
| 16:31.26 | Qwell | yell at O'Reilly. I'm sure they'll replace it. |
| 16:31.42 | mcf3782 | very clean cut. Like a big break press just had a bad day. :) |
| 16:33.54 | drmessano | It's because the authors take turns printing them off, and one of them is left handed and using a right handed cutter |
| 16:33.57 | drmessano | It happens :? |
| 16:37.37 | Greenlight | Howdy folks. Quick question - I want to use the AMI to set a channel variable to the current Asterisk system time (accurate to ms), but I can't see an obvious way to do this - Any ideas? |
| 16:38.58 | pabelanger | mcf3782: screenshot |
| 16:41.47 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:43.41 | Greenlight | So no easy way to set a channel variable to the current time using AMI ? |
| 16:46.47 | mcf3782 | The nice folks at O'Reilly are sending me a replacement copy. :) |
| 16:48.17 | mcf3782 | pabelanger: I don't have a camera handy. I'll try and take one tonight. I put the belt-clip half of my iPhone's OtterBox case on my belt this morning….and left the phone sitting on the dresser, plugged into the charger. |
| 16:48.23 | mcf3782 | *headdesk* |
| 16:49.23 | *** join/#asterisk dijib (~dijib@bas10-kitchener06-1176139363.dsl.bell.ca) |
| 16:49.35 | dijib | p3nguin: i have mailed that thing as of 20min ago |
| 16:50.04 | mcf3782 | Maybe that's a hint that AGI isn't the way to do what I'm trying to do. |
| 16:54.23 | dijib | p3nguin: you around? |
| 16:54.50 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 16:56.14 | mcf3782 | I am creating MeetMe conference rooms dynamically on the fly for some users. They're prompted to enter a five digit number for the conference they want to create. I want to check to see if the number the user enters, is already an active conference. Something like 'asterisk -x "meetme list concise" | cut -d \! -f 1 |grep $ROOM_NUM'. What's the cleanest way to do this inside the dialplan? |
| 16:59.49 | Greenlight | You could use that with http://www.voip-info.org/wiki/view/Asterisk+func+shell |
| 16:59.56 | Greenlight | But it does seem messy |
| 17:00.14 | *** join/#asterisk Patraulea (~Patraulea@195.113.171.130) |
| 17:00.16 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:00.47 | Patraulea | hi there |
| 17:01.12 | Greenlight | mcf3782: What about http://www.asterisk.org/docs/asterisk/trunk/functions/meetme_info |
| 17:01.20 | Patraulea | somebody cand please help me with my asterisk ? ... i have an error that i try to fix for 3 hours .... :((( |
| 17:01.29 | Greenlight | mcf3782: Or maybe even http://www.asterisk.org/docs/asterisk/trunk/applications/meetmecount |
| 17:02.11 | Patraulea | <PROTECTED> |
| 17:02.12 | Patraulea | <PROTECTED> |
| 17:02.38 | Patraulea | nobody ? :-( |
| 17:02.43 | Greenlight | Umm |
| 17:02.49 | Greenlight | And what is the problem you're getting ? |
| 17:03.22 | Patraulea | when I answer to the phone the AGI does'nt lunch the IVRB.php .. |
| 17:03.58 | mcf3782 | MeetMe_Info seems to be for getting more detailed info about a specific conference. In this case, I don't really care what the conference is doing; I just want to know if it exists. So if user1 picks '56789' for his conference number, and later user2 comes along and picks '56789' while user1's conference is still going; user2 gets told that number is already in use. |
| 17:03.58 | Greenlight | According to what you pasted it did... |
| 17:04.22 | Patraulea | he did'nt play the sound's ... |
| 17:04.47 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:04.51 | Greenlight | mcf3782: I was thinking that if you called MeetMe_Info on a non existing conference compared to an existing one it would give you enought detail to tell |
| 17:05.04 | mcf3782 | Maybe I could run MeetMe_Info and check for return codes and see if it errors if the number doesn't exist...... |
| 17:05.18 | Greenlight | mcf3782: Yes |
| 17:05.37 | mcf3782 | That's a path I hadn't considered. |
| 17:06.12 | Greenlight | Patraulea: Do you have the sound files there, can you play then normally from the dialplan? Are permissions set correctly on both the php script and the sounds? |
| 17:06.16 | mcf3782 | Thanks, Greenlight. I'll go try that route and see if I can code something up that works. |
| 17:06.20 | Patraulea | yes |
| 17:06.23 | Greenlight | mcf3782: Hope it works :) |
| 17:06.30 | mcf3782 | me too |
| 17:06.42 | Patraulea | i have the same files to another PBX |
| 17:06.49 | Patraulea | and it's working perfect |
| 17:07.44 | Patraulea | :( |
| 17:07.49 | Greenlight | You should add some debugging to your AGI script to spit stuff out to the CLI and maybe you will see where it's failing |
| 17:08.06 | Greenlight | I presume the script excutes okay when you run it from the shell ? |
| 17:08.13 | Patraulea | yes |
| 17:08.37 | Patraulea | maybe because are 2 diferent's linux version's ? |
| 17:08.46 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 17:08.52 | Greenlight | There could be a million different things |
| 17:09.07 | Greenlight | That's why you gotta find out where/why its failing |
| 17:09.42 | Patraulea | one is CentOS release 5.8 that is working perfect and the another one is CentOS Linux release 6.0 that is the same asterisk and script and he's not working :( |
| 17:10.11 | Patraulea | how can i do that ? because i'am not so expert... |
| 17:10.12 | Greenlight | You've not got selinux enabled? |
| 17:10.18 | Patraulea | i'am a little guffy |
| 17:10.53 | Patraulea | Greenlight can we talk on private? |
| 17:11.09 | Patraulea | if is okay with you... |
| 17:11.11 | Greenlight | No, sorry, I've gotta go in a moment |
| 17:11.20 | Patraulea | :O( |
| 17:11.27 | Greenlight | The Mrs will have my dinner on the table, I must leave the office |
| 17:11.42 | Patraulea | ok no problem |
| 17:11.54 | Greenlight | Make sure you've disabled selinux: setenforce off |
| 17:12.00 | Greenlight | @ shell |
| 17:12.07 | Greenlight | Then try it |
| 17:12.11 | Patraulea | 360by247*CLI> |
| 17:12.12 | Patraulea | ? |
| 17:12.21 | Greenlight | Naa, not at the Asterisk CLI |
| 17:12.24 | Greenlight | At your shell |
| 17:12.26 | Patraulea | [root@360by247 xpldial]# setenforce off |
| 17:12.27 | Patraulea | -bash: setenforce: command not found |
| 17:12.27 | Patraulea | [root@360by247 xpldial]# |
| 17:12.44 | Patraulea | like i say'd iam a little guffy :P |
| 17:12.57 | Greenlight | Guess it's not that then - was just a hunch |
| 17:13.06 | Patraulea | oh... |
| 17:13.19 | Greenlight | So, when you executed the script manually whats the output? |
| 17:13.36 | Patraulea | it's playing a message |
| 17:13.44 | Patraulea | a song |
| 17:13.47 | Greenlight | Eh |
| 17:14.08 | Patraulea | i try everyting |
| 17:14.26 | Greenlight | SO you ran it from the shell like ./IVRB.php |
| 17:14.30 | Patraulea | i have like 4 hours try to fix it ... |
| 17:14.34 | Greenlight | And... it played a song |
| 17:14.48 | Greenlight | That makes no sense |
| 17:14.57 | Patraulea | [root@360by247 agi-bin]# ./IVRB.php |
| 17:14.57 | Patraulea | PHP Parse error: syntax error, unexpected T_GOTO, expecting T_STRING in /var/lib/asterisk/agi-bin/lib/asagi.php on line 1208 |
| 17:14.58 | Patraulea | [root@360by247 agi-bin]# |
| 17:15.07 | Greenlight | sighs |
| 17:15.12 | Greenlight | [06:08pm] <Greenlight> I presume the script excutes okay when you run it from the shell ? |
| 17:15.12 | Greenlight | [06:08pm] <Patraulea> yes |
| 17:15.24 | Greenlight | Does that look like "ok"? |
| 17:15.34 | *** join/#asterisk rubberneck (~chatzilla@ext-100.sagetelecom.net) |
| 17:15.38 | Patraulea | nop :) like I say'd i'am guffy guf :P |
| 17:16.15 | rubberneck | Nothing jumps out at me from a Google search.. Is there a open source video conference bridge for asterisk? |
| 17:17.13 | Greenlight | Not sure, but that's where your problem is - if it's working elsewhere then maybe different php versions, or configs |
| 17:24.02 | Greenlight | Good luck, an away. Laters all |
| 17:24.02 | Patraulea | aha |
| 17:24.03 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 17:24.10 | Qwell | rubberneck: confbridge |
| 17:24.22 | rubberneck | Qwell: Do you know if it is only available for v10? |
| 17:24.43 | Qwell | yes, Asterisk 10 or higher |
| 17:26.23 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 17:26.45 | rubberneck | Qwell: Thanks, have any experience with it by any chance? Curious to know how well it works. |
| 17:27.42 | Qwell | it works well |
| 17:34.21 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:43.16 | *** join/#asterisk gusto (~gusto@DSL01.83.171.190.103.ip-pool.NEFkom.net) |
| 17:45.53 | rubberneck | Qwell: Thanks. |
| 17:47.07 | *** join/#asterisk PaybackTony (~Adium@75-148-84-182-Oregon.hfc.comcastbusiness.net) |
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| 18:12.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 18:27.48 | *** join/#asterisk dorphalsig (be93997b@gateway/web/freenode/ip.190.147.153.123) |
| 18:29.24 | dorphalsig | Hi Im running * 1.8 and I'm trying to configure cdr and queue log in postgres via odbc. ODBC is ok, and asterisk can connect to it. However when writing the cdr record I get this message: WARNING[2037]: res_odbc.c:605 ast_odbc_direct_execute: SQL Execute error! Verifying connection to PostgreSQL [asterisk]... |
| 18:33.20 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 18:36.18 | wdoekes | dorphalsig: could be that constraints on the table don't accept the inserted cdr |
| 18:37.13 | wdoekes | look at the query (core set debug .. or postgres logging) being performed and check if you can run it manually on the postgres db |
| 18:39.28 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 18:39.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:39.44 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 18:47.35 | gusto | hey |
| 18:47.55 | gusto | when i have asterisk - asterisk (IAX2 - IAX2) do i have to create a user on both sides? |
| 18:48.21 | dorphalsig | wdoekes: Thanks, it was just a permissions stuff. The user had no permissions to insert :s |
| 18:48.40 | dorphalsig | ~wiki |
| 18:48.46 | p3nguin | gusto: http://pastebin.com/Ag7tknm2 <--- |
| 18:49.25 | gusto | p3nguin: you gave me the wrong one |
| 18:49.30 | gusto | p3nguin: that is SIP-SIP |
| 18:49.37 | p3nguin | No I did not give you the wrong one. |
| 18:50.05 | p3nguin | The example is SIP, but the concept is channel agnostic. |
| 18:50.09 | gusto | well, then something differnt must be wrong, because i did it according to that howto |
| 18:50.30 | gusto | so i do not have to have users on both sides? |
| 18:50.42 | p3nguin | Pastebin what you've done. Two pastebins. One from system A, one from system B. |
| 18:50.53 | gusto | <PROTECTED> |
| 18:51.02 | p3nguin | You don't have "users" in any place. You create peers. |
| 18:51.52 | gusto | so what does that mean? "no authority found", ha? |
| 18:52.32 | p3nguin | I'll wait for the pastes, but I won't wait long. |
| 18:53.32 | gusto | well, i have to look it up first |
| 18:54.03 | gusto | i ve found out with google that it can happen when the secrets are wrong, but reigstry to each other works |
| 18:54.24 | gusto | also iax2 show peers is looking good |
| 18:55.25 | gusto | [Jul 5 20:55:08] NOTICE[26896]: chan_iax2.c:10965 socket_process: Host 83.171.190.103 failed to authenticate as norimberg |
| 18:55.31 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 18:55.32 | gusto | says the other part |
| 19:06.10 | gusto | i have the problem |
| 19:06.16 | gusto | he is sending the wrong username |
| 19:06.28 | gusto | but it's not my mistake, he sends the username from another peer |
| 19:08.39 | gusto | well, maybe i should just restart asterisk |
| 19:08.58 | p3nguin | That isn't a fix. |
| 19:12.34 | *** join/#asterisk luckman212 (~irc@2001:470:8abb:0:4f9:27f7:61fa:ba45) |
| 19:13.07 | gusto | no, it was not |
| 19:16.30 | gusto | the interesting thing is, that the authentification works |
| 19:16.39 | gusto | the md5 result is always the same on both sides |
| 19:16.46 | gusto | also now the usernames are right |
| 19:17.17 | gusto | <PROTECTED> |
| 19:17.18 | gusto | <PROTECTED> |
| 19:18.10 | gusto | maybe there should be the same context? |
| 19:18.27 | gusto | i ll try it with context! |
| 19:20.21 | gusto | that does not help either |
| 19:20.27 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 19:21.55 | p3nguin | Still waiting on the pastebins. Time is wasting. |
| 19:22.27 | *** join/#asterisk Ralcala (~demo@201.127.143.106) |
| 19:22.45 | edge | In asterisk 1.8 , I use the Set(CALLERID(all)="Some name" <1234567890>) , then i jump to a Dial() function (to my cellphone) and i don't see the changed Callerid. Is there something i'm missing? |
| 19:23.24 | [TK]D-Fender | edge, Yes, proof of what actually got sent out, and details as to what you are sending it over. |
| 19:23.53 | *** join/#asterisk JonMR (~jon@sperg.p1mp.org) |
| 19:23.55 | gusto | This error also occurs when the context requested in the incoming call is different from one in the section of incoming peer. Multiple contexts in the peer section can be defined. |
| 19:23.59 | gusto | http://www.voip-info.org/wiki/view/Asterisk+No+authority+found |
| 19:24.45 | edge | [TK]D-Fender: How to I capture what actually got sent out? |
| 19:24.57 | edge | [TK]D-Fender: i'm sending this over DAHDI , POTs line |
| 19:25.07 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 19:25.08 | p3nguin | edge: Quatations are not valid there. Set(CALLERID(all)=Some name <1234567890>) |
| 19:25.44 | p3nguin | s/Quatations/Quotations/ |
| 19:25.45 | edge | p3nguin: you mean these <> |
| 19:25.51 | Ralcala | hi fellas |
| 19:26.04 | Ralcala | i cant register to my sip trunk provider |
| 19:26.04 | p3nguin | <p3nguin> edge: Quatations are not valid there. Set(CALLERID(all)=Some name <1234567890>) <-----------What I mean. |
| 19:26.07 | edge | p3nguin: oh ic |
| 19:26.35 | Ralcala | soft: * 1.8.11 |
| 19:26.51 | [TK]D-Fender | edge, You can't send callerID over POTS |
| 19:26.56 | p3nguin | 1.8.13.0 is current |
| 19:26.57 | [TK]D-Fender | ^^^66 |
| 19:27.01 | [TK]D-Fender | DOA <- |
| 19:27.29 | Ralcala | mmmm |
| 19:27.43 | [TK]D-Fender | edge, tht was for you |
| 19:27.49 | Ralcala | i got response 400. Bad Request. error |
| 19:27.56 | [TK]D-Fender | Ralcala, www.pastebin.com <-------------- |
| 19:28.01 | Ralcala | k |
| 19:29.20 | edge | [TK]D-Fender: What would i need so i could set caller id? See we're an office that shares two busineses. and we were going to pool lines together. So if a call went out from a business A set, we'd tag it with their return number, and dido with Business B |
| 19:29.42 | [TK]D-Fender | edge, get different connectivity to the PSTN |
| 19:30.11 | edge | [TK]D-Fender: Ok. So long as i understand WHY it doesn't work. Thank you very much. |
| 19:30.13 | p3nguin | Depending on your connection type, the telco could allow you to set some predefined caller ID info. |
| 19:30.47 | [TK]D-Fender | edge, Because POTS is dumb analog and there isn't a real signalling standard for it. |
| 19:31.22 | edge | [TK]D-Fender: But we still receieve it. So is the telco setting that CID for us on the way to the receiving party? |
| 19:31.40 | [TK]D-Fender | edge, Receiving CID has nothing to do with being able to send it |
| 19:31.58 | [TK]D-Fender | edge, And yes, the telco sends you the incoming info |
| 19:32.33 | [TK]D-Fender | edge, Standard CID is OOB FSK between the first & second rings. |
| 19:32.45 | [TK]D-Fender | edge, there is no "send" standard |
| 19:33.07 | [TK]D-Fender | edge, While inventing one is possible, it simply isn't a standard. |
| 19:33.16 | gusto | p3nguin: i was right! IAX wants to have a user! |
| 19:33.30 | p3nguin | You should have followed my example. |
| 19:33.32 | [TK]D-Fender | gusto, Not needed |
| 19:33.36 | gusto | p3nguin: i just changed it from peer to friend in type and now it goes |
| 19:33.52 | gusto | [TK]D-Fender: with type=peer it does not work |
| 19:34.01 | p3nguin | It would if you follow my example.` |
| 19:34.05 | edge | [TK]D-Fender: I'm glad to know. Thank you for the knowledge. I'm going to see what is offered in our area for maybe a t1 or PRI link |
| 19:34.09 | [TK]D-Fender | gusto, You aren't listening, looking, or showing |
| 19:34.10 | gusto | p3nguin: i followed that |
| 19:34.13 | drmessano | You don't need a user |
| 19:34.29 | gusto | yes, but a friend in type |
| 19:34.34 | p3nguin | type=peer handles BOTH directions. |
| 19:34.40 | [TK]D-Fender | edge, If you're small, an ITSP may be a better option as cost-effectiveness goes |
| 19:34.47 | gusto | and that one does create a user then |
| 19:34.50 | drmessano | friend is just peer+user and you don't need a user here, just a peer |
| 19:35.01 | edge | [TK]D-Fender: we are very small (currently only 5 lines between two busineses) |
| 19:35.15 | gusto | http://www.voip-info.org/wiki/view/Asterisk+No+authority+found <- has type=friend as well |
| 19:35.20 | [TK]D-Fender | edge, ITSP has a high likelyhood of being a much better value for you then |
| 19:35.34 | [TK]D-Fender | gusto, And that wiki is full of decrepit crap. |
| 19:35.44 | drmessano | gusto, you created a problem with an inproper setup, you resolved it with more duct tape. |
| 19:35.49 | edge | [TK]D-Fender: is a ITSP a VOIP trunk? |
| 19:35.53 | [TK]D-Fender | ~itsp |
| 19:35.53 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 19:35.58 | p3nguin | ~trunk |
| 19:35.58 | infobot | i heard 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. the nose of an elephant |
| 19:36.36 | drmessano | If adding a user or changing to type=friend fixes it, you set up the peer incorrectly. You see it as tomato/tomato.. It's more like tomato/rhinocerous |
| 19:36.40 | gusto | so does anybody have a working IAX-IAX setup? |
| 19:36.44 | edge | <PROTECTED> |
| 19:36.44 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 19:36.54 | drmessano | gusto: What p3nguin gave you is just that |
| 19:37.02 | gusto | that was SIP-SIP |
| 19:37.08 | gusto | and that worked |
| 19:37.08 | [TK]D-Fender | gusto, You aren't showing anything and are wasting our time until you do... |
| 19:37.14 | p3nguin | Like I told you, the concept is the same from IAX2. |
| 19:37.38 | *** join/#asterisk Osage_ (~Osage@71-81-57-248.dhcp.slid.la.charter.com) |
| 19:38.22 | gusto | well, w8 |
| 19:38.22 | gusto | i have something |
| 19:38.31 | gusto | i once set up a IAX2 to IAX2 and i could call the other side |
| 19:39.35 | edge | [TK]D-Fender: Thanks again. I have all the knowledge i need to make a informed choice. |
| 19:39.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.40.134) |
| 19:40.04 | [TK]D-Fender | edge, When you want to break down what you think your final picture will look like we'll be here |
| 19:40.30 | [TK]D-Fender | edge, To help make sure the overall picture suits your needs & expectations |
| 19:40.45 | mcf3782 | My brain hurts. It's so frustrating that I'm dense as a post when it comes to programming. |
| 19:41.06 | drmessano | When choosing an ITSP you should start off by using Flowroute, and then umm.. Thats pretty much it |
| 19:41.20 | gusto | now i know what is different |
| 19:41.31 | drmessano | Good |
| 19:41.32 | gusto | on the working example, i have one name for both peers |
| 19:41.33 | drmessano | We dont |
| 19:42.13 | p3nguin | This is why I share a good example for Asterisk to Asterisk configuration. |
| 19:42.20 | gusto | so it's username=virtualboxbridge on both sides |
| 19:42.39 | drmessano | .... |
| 19:42.59 | p3nguin | I'm going to make a drop-in IAX2 config to supplement the SIP to SIP config I already gave you for reference. |
| 19:44.32 | gusto | well and then try to make a call |
| 19:45.08 | gusto | because authentification, peers, registry, all works, the problem occurs when you want to call someone through that IAX-IAX |
| 19:46.44 | [TK]D-Fender | p3nguin, ... not worth it. |
| 19:47.03 | p3nguin | I'll do it for others with the same questions. |
| 19:47.51 | [TK]D-Fender | p3nguin, Better odds... |
| 19:49.51 | drmessano | "I want help, but really I want moral support for the stupid shit I am going to do, because I don't plan on actually listening" |
| 19:54.17 | carrar | heh |
| 19:59.46 | *** join/#asterisk rolandow (~rolandow@546BB29B.cm-12-4c.dynamic.ziggo.nl) |
| 20:00.12 | rolandow | good evening! :) |
| 20:00.45 | rolandow | i am considering running asterisk on a dedicated server, so i can connect several offices to that asterisk .. so i'd run my own hosted asterisk. |
| 20:00.53 | rolandow | what would be good enough ping times? |
| 20:01.04 | rolandow | like.. would 20ms be sufficient? |
| 20:01.08 | [TK]D-Fender | more than |
| 20:01.27 | rolandow | ok .. and other than that i think asterisk runs mainly from memory right? |
| 20:01.30 | p3nguin | Here's the supplement for IAX2: http://pastebin.com/UUxJtERA |
| 20:01.56 | rolandow | so i/o and cpu isn't all that important.. i'm talking about around 100 sip clients total |
| 20:02.11 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 20:03.47 | [TK]D-Fender | rolandow, No, they can be very important as transcoding, conferencing, etc all comes into play. |
| 20:03.50 | [TK]D-Fender | recording as well |
| 20:04.20 | rolandow | hm.. but transcoding is probably temporary space right? |
| 20:04.25 | rolandow | i could run it from tmpfs maybe? |
| 20:04.50 | Osage_ | No, transcoding occurs realtime in the cpu or in a dsp |
| 20:05.30 | justdave | is there a way with the built-in voicemail in asterisk to preview your existing greeting messages on your voicemail? |
| 20:05.43 | justdave | if I just want to know what my greeting already is without re-recording it? |
| 20:06.03 | justdave | the menu options appear to only deal with recording it over again |
| 20:06.12 | rolandow | but are 7k2 sata's sufficient? |
| 20:06.37 | rolandow | server4you has nice deals for us since they host in europe |
| 20:07.27 | rolandow | justdave: why not just dial your own voicemail then? |
| 20:07.43 | [TK]D-Fender | rolandow, transcode = CPU |
| 20:07.49 | rolandow | would be nicer to have in the menu though |
| 20:07.51 | [TK]D-Fender | and recording is IO to HD |
| 20:08.29 | [TK]D-Fender | justdave, "core show application playback" <- |
| 20:08.32 | justdave | rolandow: because that only gives me the unavailable message and I want to hear the busy one? |
| 20:08.36 | mcf3782 | Can someone spot what I'm doing wrong in lines 11-16 in this paste, based on the log messages below it? http://pastebin.com/wGNkf9xj |
| 20:08.41 | [TK]D-Fender | And also usually transcoding as well |
| 20:08.50 | gusto | ok |
| 20:09.06 | gusto | i got it working with type=peer, but i had to get rid of the username= on both sides |
| 20:10.17 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 20:10.28 | justdave | [TK]D-Fender: you're suggesting I roll a custom dialplan extension for it I'm guessing. :) (Which is certainly an option) |
| 20:10.55 | rolandow | [tk]d-fender: ok .. i see .. but i suppose with such less users normal sata disks will do .. we don't have voicemail either.. |
| 20:11.08 | [TK]D-Fender | justdave, OR fall into your own VM by whatever otehr means you normally would |
| 20:11.28 | *** join/#asterisk Alex_Bkash (b4ea7e56@gateway/web/freenode/ip.180.234.126.86) |
| 20:11.36 | [TK]D-Fender | mcf3782, What's the problem? |
| 20:11.37 | Alex_Bkash | hello |
| 20:11.50 | Alex_Bkash | anyone using peppermint? |
| 20:12.04 | rolandow | hm.. gotta get me bitchx or epic.. brb |
| 20:12.11 | gusto | p3nguin: have you tried your http://pastebin.com/UUxJtERA ? you have username= in it |
| 20:12.23 | Alex_Bkash | dpkg error code (2) |
| 20:13.36 | Qwell | Alex_Bkash: How is that an Asterisk question? |
| 20:13.56 | Alex_Bkash | im triying to install asterisk thr |
| 20:14.44 | mcf3782 | [TK]D-Fender: The goal is to check to see if the number entered at line 9, is a conference that's already running, by trying to count the number of users in it with MeetmeCount on line 11. If there are more than 0 users in the conference, that means that conference number is already used. Play the 'unavailable' message in line 14, then send the user back to (passed) to try again. Otherwise continue with (free) at line 16. |
| 20:16.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v003-127.mobile.uci.edu) |
| 20:16.37 | [TK]D-Fender | mcf3782, same => n,GotoIf($[ ${count} >= 0 ]?inuse:free) |
| 20:16.53 | [TK]D-Fender | mcf3782, If you have ***0*** users .... how is that "inuse"? |
| 20:17.00 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 20:17.03 | p3nguin | :) |
| 20:17.22 | [TK]D-Fender | mcf3782, ">=" != ">" |
| 20:17.23 | *** join/#asterisk rolandow (~rolandow@546BB29B.cm-12-4c.dynamic.ziggo.nl) |
| 20:17.34 | [TK]D-Fender | MATH FAIL |
| 20:18.02 | [TK]D-Fender | -- Executing [2@gc-conf-bridge:8] SayNumber("SIP/503-0000007e", "0") in new stack <------ |
| 20:18.11 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:18.27 | mcf3782 | I've tried ">0" ">=0" and ">1" |
| 20:18.43 | [TK]D-Fender | mcf3782, What you're showing me now is a logic failure. |
| 20:18.54 | [TK]D-Fender | mcf3782, It is doing exactly what it shoul |
| 20:19.06 | [TK]D-Fender | mcf3782, Show me something else |
| 20:20.08 | mcf3782 | *sigh* I hate being stupid. Can I have a brain transplant from some smart person who doesn't need his? |
| 20:20.50 | [TK]D-Fender | mcf3782, May suggest .. electro-shock .... |
| 20:20.57 | [TK]D-Fender | DON'T TASE ME BRO!!! |
| 20:21.02 | mcf3782 | Yea. It's a logic failure. I'll keep trying. |
| 20:21.04 | [TK]D-Fender | bzzzzZZZZTTT!!!!! |
| 20:22.32 | edge | Alex_Bkash: you should be installing from source. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#asterisk-Install |
| 20:23.04 | rolandow | damnit.. why doesn't ircii autocomplete nicknames |
| 20:23.20 | rolandow | or in other words.. would anybody happen to know the key binding? :) |
| 20:23.36 | mcf3782 | I've been shot with a tazer. Long story, but I had to carry one for a while. In order to be allowed to carry and possibly use it; it was required that you know what it feels like to be on the receiving end of it. |
| 20:23.42 | mcf3782 | It was not pleasant. |
| 20:24.09 | p3nguin | rolandow: Consider irssi. |
| 20:24.47 | rolandow | ok .. i think i saw that in the packages .. :) brb |
| 20:27.49 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 20:29.01 | [TK]D-Fender | Checkout time, BBIAB |
| 20:29.23 | *** join/#asterisk rolandow (~rolandow@546BB29B.cm-12-4c.dynamic.ziggo.nl) |
| 20:29.37 | rolandow | nice.. this almost looks like bitchx! |
| 20:30.09 | rolandow | p3nguin: thx (this was also a undercover test to see if tab works) |
| 20:33.46 | *** join/#asterisk Alex_Bkash (b4ea72ac@gateway/web/freenode/ip.180.234.114.172) |
| 20:37.42 | *** join/#asterisk GregH105 (42557a06@gateway/web/freenode/ip.66.85.122.6) |
| 20:38.32 | GregH105 | hi all. I have just discovered a problem with our server in that some calls get placed in our "Sales" queue but never ring any lines |
| 20:39.23 | *** join/#asterisk twanny796 (~twanny@46.11.2.155) |
| 20:40.53 | gusto | p3nguin: somehow it makes sense, when there is no user on either side, then there should be no username= argument |
| 20:42.55 | gusto | p3nguin: also it makes sense why that problem came up as soon as i wanted to call, because then he looks it up first |
| 20:44.06 | gusto | p3nguin: so i mean, when i call him (even though he is a peer to me) he expects that i am a user to him when there is username= in |
| 20:44.26 | gusto | what is in most cases also correct, but not in this one |
| 20:50.02 | gusto | however, that example might work as long as you have auth=plaintext in it, but with md5 digest it does not matter if you use a username or not, becase it does trasfer it, it's just a piece of bullshit to combine it with random data and make md5's of it |
| 20:54.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:55.30 | *** join/#asterisk jero (~boo@mtl.savoirfairelinux.net) |
| 20:55.38 | wdoekes | gusto: username= is a deprecated option that does not do what you might expect |
| 20:56.33 | gusto | wdoekes: well, in iax.conf it's still being used |
| 20:57.01 | gusto | wdoekes: http://pastebin.com/UUxJtERA |
| 20:57.15 | wdoekes | if you're talking iax, then I have no idea.. I assumed you're using sip |
| 20:57.24 | gusto | hahahahahahaha |
| 20:57.31 | drmessano | Deprecated is not the same as removed |
| 20:57.35 | gusto | well, with SIP it works like you might expect |
| 20:57.42 | gusto | i know |
| 20:57.58 | gusto | i do not say that it's been removed or whatever, that was wdoekes |
| 20:58.01 | p3nguin | username still works, but with warnings. |
| 20:58.04 | p3nguin | (in sip.conf) |
| 20:58.10 | drmessano | Well, "Its still being used" is irrelevant. Doesn't make it any less buggy and deprecated |
| 20:58.16 | gusto | well, dont mix this up now |
| 20:58.37 | gusto | how can a parameter be buggy? |
| 20:59.05 | drmessano | Parameters are tied to functions |
| 20:59.06 | [TK]D-Fender | gusto: Imagine as though you wronte the code..... |
| 20:59.42 | drmessano | Nobody said the line in the config file is buggy, there's code that does stuff when you set those parameters |
| 21:00.03 | drmessano | Raw-Cat Science |
| 21:00.04 | p3nguin | or code that doesn't do stuff... |
| 21:00.22 | p3nguin | whatever the case may be. |
| 21:00.24 | gusto | well |
| 21:00.31 | gusto | what are you talking now about? |
| 21:00.42 | gusto | the username= in sip.conf or iax.conf as well? |
| 21:00.49 | drmessano | Either, or |
| 21:00.56 | gusto | not both! |
| 21:01.23 | gusto | how can that have the same code? it's two different modules. |
| 21:01.24 | p3nguin | Both! Together! It's a new file called "iax2.sip" |
| 21:02.17 | drmessano | Jesus |
| 21:02.26 | gusto | ??? |
| 21:03.21 | gusto | now no one should wonder that i do ignore stuff from time to time |
| 21:03.35 | p3nguin | You ignore the wrong bits. |
| 21:03.53 | gusto | well, but i found the right one |
| 21:04.06 | drmessano | You should be using complimentary peer names, not trying to force shit to work with username= and wondering why its failing all over the place |
| 21:04.31 | gusto | hahahahahahaha |
| 21:04.39 | gusto | that's what i was saying all the time |
| 21:04.51 | gusto | that usernames= stuff was p3nguin's idea |
| 21:05.03 | p3nguin | because it works. |
| 21:05.18 | gusto | heh, not to me |
| 21:09.41 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.2 (2012/07/05), 1.8.13.1 (2012/07/05), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 21:16.55 | mcf3782 | I have tried >0, >=0, =0, >=1, >1 with ?inuse:free. I have tried >0, >=0, =0, >=1, >1, =1 with ?free:inuse. None of these combinations works. |
| 21:18.54 | p3nguin | Show me some type of failure. |
| 21:19.26 | mcf3782 | http://pastebin.com/wGNkf9xj |
| 21:21.02 | p3nguin | Where is the failure in that? |
| 21:21.23 | PaybackTony | I know this isn't exactly asterisk related but maybe somewhere here has run into this issue before. I'm running CentOS 6 on a godaddy virtual machine and whenever I try to wget or an rpm -Uvh a package from the web, it seems to take a few minutes before it resolves and starts downloading anything. Once it starts it's very quick (10mb/s). Would this be a config issue on my side or? I haven't had this issue before on a CentO |
| 21:22.23 | PaybackTony | the files I'm trying to get are asterisk related (from the asterisk.org domain) |
| 21:25.33 | mcf3782 | Change line 13 to any combination of what I listed….. I either get "exited non-zero 12" or "unavailable" and start over at (passed). |
| 21:26.32 | mcf3782 | I have been working on this same ^$#@#@^%$%$#^ 6 lines of code for over 10 hours. |
| 21:27.45 | dijib | p3nguin: did you get my message earlier |
| 21:27.45 | dijib | ? |
| 21:28.15 | p3nguin | Where did you put it? |
| 21:30.37 | vastina | i performed an lspci and need to find the version of my digium card |
| 21:30.45 | vastina | doesn't seem to be coming up |
| 21:34.03 | p3nguin | Executing [2@gc-conf-bridge:8] SayNumber("SIP/503-0000007e", "0") |
| 21:34.06 | p3nguin | so count = 0 |
| 21:34.07 | WIMPy | clicked the do not click this button button |
| 21:34.19 | p3nguin | Executing [2@gc-conf-bridge:9] GotoIf("SIP/503-0000007e", "1?inuse:free") |
| 21:34.31 | p3nguin | The condition is TRUE, therefore going to the inuse label. |
| 21:34.44 | p3nguin | Executing [2@gc-conf-bridge:10] Playback("SIP/503-0000007e", "unavailable") |
| 21:35.03 | p3nguin | I don't see a failure here. |
| 21:36.03 | p3nguin | Nope. It's working as you told it to work. |
| 21:36.38 | p3nguin | Perhaps you wanted it to do something else. Maybe you should tell me what it is you want it to do and I can determine what it needs to be told to do. |
| 21:36.43 | mcf3782 | I'll try and create a better pastebin post. |
| 21:37.04 | p3nguin | I'm satisfied with that post. |
| 21:37.15 | p3nguin | The line of code is working correctly. |
| 21:37.23 | p3nguin | s/code/dialplan/ |
| 21:40.43 | mcf3782 | The goal is to try to use MeetMeCount, to check and see how many users are in the conference number that the user entered on line 9. If the number of users is non-zero (>=0), then there must be a conference going on in that room already. The user needs to pick a different 5 digit conference number. If the number they picked is "unavailable", tell them that at line 14, then send them back to (passed) at line 8; so that they c |
| 21:41.04 | mcf3782 | Once they pick one that's not already in use, continue on with line 16. |
| 21:42.57 | mcf3782 | Sorry, p3nguin. I'd said that in here earlier. I was trying not to flood other's screens with the same stuff over and over. I guess I should have just put all that at the top of my original pastebin post. |
| 21:44.14 | mcf3782 | To my brain, this shouldn't be any different than the comparison that's being done (and working) in line 7. |
| 21:44.45 | p3nguin | >=0 is not "non-zero" |
| 21:44.51 | p3nguin | It's 0 or more. |
| 21:45.24 | mcf3782 | Isn't ">=0" "greater than or equal to zero"? |
| 21:45.39 | drmessano | yes |
| 21:45.42 | p3nguin | equal to zero is not non-zero; it's zero. |
| 21:46.10 | p3nguin | non-zero would be >0 or <0 |
| 21:46.12 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 21:46.19 | p3nguin | But since you won't have less than zero, <0 would be bad. |
| 21:47.15 | p3nguin | If you want to check if it is not zero, you could use !=0 |
| 21:47.45 | p3nguin | (but >0 would be just as good in this case) |
| 21:48.34 | mcf3782 | >0 gets me "exited non-zero 12" |
| 21:48.50 | [TK]D-Fender | that message has nothing to do with the math |
| 21:48.58 | [TK]D-Fender | NEW PASTEBIN PLEASE |
| 21:49.09 | mcf3782 | ok. |
| 21:50.04 | drmessano | same => n(free)Playback(custom/conf-admin-pin) <--- No comma |
| 21:50.24 | mcf3782 | Let me clear out my log file and start over |
| 21:50.28 | p3nguin | That's certainly an issue. |
| 21:50.31 | drmessano | Yep |
| 21:50.37 | drmessano | free is gonna fail |
| 21:50.46 | drmessano | You need to fix that and try again |
| 21:51.00 | mcf3782 | OK. |
| 21:51.04 | mcf3782 | goes to edit |
| 21:51.07 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 21:51.19 | [TK]D-Fender | aND GET "HANG" OUT OF THERE |
| 21:51.26 | drmessano | Do I get to go on to the showcase showdown? |
| 21:51.52 | drmessano | lol |
| 21:52.16 | p3nguin | Hang? |
| 21:52.24 | drmessano | I missed that too.. at the top |
| 21:52.27 | p3nguin | lines 3 and 4, I guess. |
| 21:52.48 | mcf3782 | What's wrong with my 'hang'? |
| 21:52.48 | p3nguin | I never looked at anything other than his GotoIf() and the execution of it. |
| 21:52.53 | [TK]D-Fender | That should not be sitting in the middle of another exten you're defining |
| 21:52.56 | p3nguin | What's right iwth it? |
| 21:53.41 | mcf3782 | ok. But I don't think it's causing the problem with my 'math error |
| 21:53.42 | p3nguin | "same =>" means the same extension. It's a shortcut for "exten => 2," in your case. |
| 21:53.43 | mcf3782 | <PROTECTED> |
| 21:53.55 | p3nguin | Your math error wasn't a math error at all. |
| 21:54.16 | p3nguin | The condition compared 0 to >=0, and found it to be true. |
| 21:54.17 | [TK]D-Fender | big syntax error there |
| 21:55.13 | [TK]D-Fender | p3nguin: I only mentioned that 2 hours ago |
| 21:55.39 | mcf3782 | ok. let me clear the log, run it again, and copy and paste a new patebin….. |
| 21:57.42 | citywok | I'm working on somebody's system, and CDR(dst) is getting set to h for some reason. There is nothing after the dial command, the hangup extension is a macro that doesn't have anything that would change the dst. |
| 21:58.01 | citywok | (set to h after dialing a number that is) |
| 21:58.48 | citywok | hmm. i wonder if it's the way they are dialing, i need to find out if they use callfiles or AMI for dialing. |
| 22:00.01 | mcf3782 | [TK]D-Fender: I truly mean this in the kindest, non-threating, non-rude, non-confrontational way possible…. Yes, you did say that 2 hours ago. But I already *knew* there was a problem. I was asking for someone to point out what the problem was, and how to correct it. Just saying "it's wrong", didn't really help. The results I'm getting have told me it's wrong. :-) |
| 22:01.20 | drmessano | So where the paste of the updated dialplan and the failure? |
| 22:01.47 | mcf3782 | working on it.. :) |
| 22:04.10 | mcf3782 | I was in the middle of trying p3nguin's "!=" suggestion. |
| 22:05.13 | mcf3782 | Aaannd. That works! |
| 22:05.19 | p3nguin | It would give the same results as >0 since you will never have negatives. |
| 22:06.37 | mcf3782 | So how does "!= 0" work, but "> 0" not work. Don't they evaluate to the same thing? "not equal to zero" is the same as "greater than zero" in this particular test…….isn't it???? |
| 22:06.57 | p3nguin | >0 will be any number above zero, excluding zero itself. !=0 would be any number below zero or above zero, excluding zero itself. |
| 22:07.35 | p3nguin | I haven't seen the pastebin showing a FAILURE yet. |
| 22:07.53 | p3nguin | There's no failure, so there is nothing to troubleshoot. |
| 22:07.58 | [TK]D-Fender | Because .... now it works |
| 22:08.05 | p3nguin | What you showed me showed it working. |
| 22:08.12 | p3nguin | Even though you said it wasn't, it was. |
| 22:08.31 | drmessano | It wasn't hitting the "free" because of the comma.. the math was right.. the rest wasnt |
| 22:08.40 | drmessano | So 10 hours for the comma |
| 22:08.56 | drmessano | and a wild goose chase on a math error that wasnt an error |
| 22:09.05 | p3nguin | His test had count = 0, so it was true. The free label was never used in that test. |
| 22:09.09 | *** join/#asterisk gtTuna (~TunaWork@38.101.60.42) |
| 22:09.21 | drmessano | We're past that now, my web footed friend |
| 22:09.42 | mcf3782 | drmessano - you could be right.. I changed the missing comma at the same time as I changed the >0 to != 0. |
| 22:10.01 | drmessano | The story shall be told from here on out that the comma was the problem.. all 10 hours of it |
| 22:10.05 | drmessano | and it is so |
| 22:10.09 | p3nguin | Now THAT makes sense. Had you shown me that in pastebin, I would have told you where the problem was. |
| 22:10.23 | mcf3782 | So I sort of invalidated my own test, by changing more than one parameter at a time. |
| 22:10.30 | p3nguin | Not really. |
| 22:11.19 | p3nguin | You had an error in the dial plan which was not encountered until you fixed the non-zero matching condition. |
| 22:11.28 | gtTuna | anyone have any experience with providing SIP trunks from asterisk to a IPECS PBX? |
| 22:11.35 | p3nguin | You were matching 0. You intended to match non-zero. |
| 22:12.29 | drmessano | He was trying to match a working dialplan too.. and we see how that went |
| 22:12.30 | mcf3782 | It had to be in the pastebin…. drmessano saw it, and I only posted one pastebin. I can't believe I stared at that for 10 hours and didn't see it. |
| 22:13.44 | *** part/#asterisk gtTuna (~TunaWork@38.101.60.42) |
| 22:15.24 | mcf3782 | So the entire section looks like this now. http://pastebin.com/uud5XdtE |
| 22:20.28 | mcf3782 | I'm almost afraid to ask….but how bad does my code look? :) |
| 22:21.46 | p3nguin | You didn't correct lines 6 and 7. |
| 22:22.36 | mcf3782 | Not yet. I thought I had to have those to make….line 50 work. |
| 22:22.56 | [TK]D-Fender | Wow... gtTuna waited a whole TWO minutes..... |
| 22:23.15 | mcf3782 | I will comment them out and see what happens. :) |
| 22:23.22 | p3nguin | You might need them, but you also need to have correct syntax. |
| 22:23.39 | p3nguin | same => hang,1,Playback() is invalid |
| 22:23.59 | p3nguin | exten => hang,1,Playback() is valid |
| 22:24.23 | p3nguin | (1653.41) <p3nguin> "same =>" means the same extension. It's a shortcut for "exten => 2," in your case. |
| 22:24.52 | p3nguin | So exten => 2,hang,1,Playback() is a failure. |
| 22:25.33 | [TK]D-Fender | mcf3782: http://pastebin.com/841zM4Eh |
| 22:26.13 | [TK]D-Fender | :) |
| 22:26.21 | p3nguin | *shrug* moving on |
| 22:27.20 | mcf3782 | I thought I'd copied those lines out of one of the examples in the book. |
| 22:27.52 | p3nguin | The book does have some errors. |
| 22:29.44 | mcf3782 | I guess I still don't really completely understand the proper flow of writing dialplan logic. |
| 22:30.12 | p3nguin | What part are you having trouble with? |
| 22:30.22 | mcf3782 | I don't remember now, why those lines are where they are. |
| 22:30.54 | [TK]D-Fender | One of the reasons I don't like "same". it's a hack to feel a little less like renumbering for the lazy.... and makes tracing up harder.... Like "I'm still afraid of AEL but want to feel 'cool'" |
| 22:32.32 | mcf3782 | So you suggest not using same and continuing with the "old" way of 'exten => x,y,...'? |
| 22:32.46 | dijib | oi sorry |
| 22:32.52 | dijib | so p3nguin you got that msg? |
| 22:33.19 | [TK]D-Fender | I'm saying its ripe for people to pay less attention to what they're doing and well... karma. |
| 22:33.52 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 22:34.43 | mcf3782 | I'd started using it, because I thought that's what would be supported moving forward with newer (>1.4) versions of the code. |
| 22:35.11 | mcf3782 | It certainly seems easier to me to use the way you're talking about. |
| 22:35.48 | *** join/#asterisk d-tech (~d-dtech@72.245.233.107) |
| 22:36.50 | mcf3782 | But I'm not a developer. I figured that if "the smart people" coded the new "same =>" syntax, and quite a few examples/samples seemed to be doing it that way; that I should follow suit and use the "new method". |
| 22:38.07 | p3nguin | If you use it with the correct syntax, it will work correctly. |
| 22:39.48 | *** join/#asterisk GameGamer43 (users.5533@id-5533.hampstead.irccloud.com) |
| 22:40.09 | dijib | do you get paid for supporting asterisk p3nguin ? |
| 22:40.30 | p3nguin | Yes. |
| 22:40.37 | p3nguin | But rarely in IRC. |
| 22:40.48 | dijib | damn |
| 22:41.06 | dijib | i just thought you might have had a contract with digium to support them on freenode |
| 22:41.17 | dijib | well ive sent you that sangoma a-200 |
| 22:41.25 | p3nguin | If I'm offering help here, I'm usually volunteering the help. |
| 22:41.41 | mcf3782 | Are you anywhere near the metro Atlanta, GA area? I'll gladly buy you a beer (or beverage of your choice). :-) |
| 22:41.43 | dijib | i dont know how you can find the time and keep your marriage alive |
| 22:41.58 | dijib | send him gift cards, or better yet CA$H |
| 22:42.24 | p3nguin | I'm not near Atlanta this week. |
| 22:42.33 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 22:42.35 | dijib | when you going to come up to canada? |
| 22:42.49 | p3nguin | I wouldn't have any idea. |
| 22:43.06 | dijib | ok so i would expect that in 1-2 weeks |
| 22:43.14 | dijib | its just taking plain jane ground mail |
| 22:43.17 | p3nguin | I'm not sure I have any significant reason to go there. |
| 22:43.24 | dijib | drink beer! |
| 22:43.31 | p3nguin | We have beers here. |
| 22:43.46 | dijib | pfft not canadian beers |
| 22:43.59 | p3nguin | What kind of beers? |
| 22:44.02 | dijib | btw i now produce my own alchohol |
| 22:44.17 | p3nguin | I produce my own methane. |
| 22:44.23 | dijib | methane how? |
| 22:44.25 | dijib | corn oil? |
| 22:44.27 | p3nguin | hahaha |
| 22:44.38 | p3nguin | methane, not methanol |
| 22:44.47 | mcf3782 | heh |
| 22:44.51 | dijib | you mean you eat beans too much>? |
| 22:44.56 | p3nguin | HAHAHA |
| 22:44.58 | dijib | trying to be mexican |
| 22:45.07 | p3nguin | Speaking of that... |
| 22:45.12 | dijib | sorry SeRi no offence meant by that |
| 22:45.17 | p3nguin | seri has been AWOL for a long time. |
| 22:45.23 | dijib | god damn guy |
| 22:45.36 | dijib | i saw him a few weeks back and said yo SeRi but no answer |
| 22:45.45 | p3nguin | I think I read that. |
| 22:45.50 | dijib | speaking of mexican SeRi ... lol were bad |
| 22:46.20 | dijib | yo so i found 4 dewalt 14.4 drills with 5 batts and 4 chargers today at the dump |
| 22:46.26 | dijib | oh and my suburban died last week |
| 22:46.33 | dijib | i think its the oil pan seal that went |
| 22:46.38 | dijib | need to fix that |
| 22:47.02 | p3nguin | pan gasket? |
| 22:52.53 | *** join/#asterisk Cubber (~ronny@cpe-24-58-133-224.twcny.res.rr.com) |
| 22:56.46 | gusto | what happend? |
| 22:56.55 | gusto | what is a dewalt and batt and charger, ha? |
| 22:57.07 | mcf3782 | bloody fscking hell. fix one problem, and create another! |
| 22:57.47 | mcf3782 | Someone please shoot me. |
| 22:57.57 | gusto | rejected |
| 22:58.04 | [TK]D-Fender | mcf3782: You're lucky. Every bug you fix in FORTRAN reveals 2 more |
| 22:58.27 | mcf3782 | People still use that? ;) |
| 23:03.08 | mcf3782 | COBOL - Compiles Only Because Of Luck |
| 23:03.46 | mcf3782 | goes back to the corner to sort his punch-card deck |
| 23:05.28 | [TK]D-Fender | goes back to hand-carving player-piano rolls. |
| 23:05.44 | mcf3782 | ok you win |
| 23:09.27 | [TK]D-Fender | Alrighty.... off to play actual piano (and probably guitar & bass, etc)... back later-ish ... maybe |
| 23:09.41 | mcf3782 | have fun |
| 23:09.58 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:11.17 | mcf3782 | I should go away and do something else too….before I break this dial plan more than I already have. |
| 23:11.39 | gusto | i still do not understand "yo so i found 4 dewalt 14.4 drills with 5 batts and 4 chargers today at the dump" |
| 23:12.12 | mcf3782 | gusto - random comment between two people who know each other…has nothing to do with asterisk |
| 23:15.39 | [TK]D-Fender | gusto: 4 DeWalt brand 14.4 Volt battery powered drills with extra batteries and battery chargers |
| 23:15.55 | phix | hai [TK]D-Fender ! |
| 23:16.10 | phix | hiw was your 4th july? |
| 23:16.33 | [TK]D-Fender | Just every year..... 3 days after Canada Day :p |
| 23:16.49 | [TK]D-Fender | Just like * |
| 23:16.52 | gusto | well, however |
| 23:16.52 | phix | oh canada! |
| 23:17.02 | [TK]D-Fender | OUR HOME AND NATIVE LAND! |
| 23:17.02 | gusto | interesting names |
| 23:17.12 | phix | <3 |
| 23:17.19 | gusto | and i do not understand how can someone be so astonished about a bettery powered drill |
| 23:17.21 | p3nguin | I'm having some Canada mints right now. |
| 23:17.28 | phix | i thougjt ou were american |
| 23:17.30 | [TK]D-Fender | phix: I see the conversion is proceeding well.... niiiiiiiccceee |
| 23:17.51 | phix | :) |
| 23:17.53 | mcf3782 | It's a very expensive battery powered drill. :) |
| 23:18.09 | p3nguin | gusto: When you find them in the bin, you get surprised. |
| 23:18.16 | phix | the conversation is always nice when i am in it :p |
| 23:19.04 | p3nguin | I think I got my 14.4V DeWalt back in 1996 or so. |
| 23:19.18 | phix | i just came back from a vegetarian restraunt, my stomach is going nuts |
| 23:19.18 | p3nguin | It was about $180 for the kit back then. |
| 23:19.49 | phix | no pun intendesd |
| 23:20.45 | [TK]D-Fender | checkout time, later all |
| 23:20.48 | mcf3782 | I wanted an extra battery for the Craftsman one my dad gave me for Christmas a few years ago. Sadly, it's less expensive to buy a new drill, charger and battery; than it is to buy one battery for a model that's a couple of years old. |
| 23:20.59 | phix | nn tk |
| 23:21.35 | phix | mcf3782: goto a battery place and get the cells replaced |
| 23:22.01 | phix | might be cheaper |
| 23:22.35 | mcf3782 | There's nothing wrong with the battery I have. I just wanted an extra battery. |
| 23:23.32 | mcf3782 | Their current model 'kit' (that comes with 2 batteries, I think I only said 1 battery earlier), is less expensive than buying one of the older batteries. |
| 23:25.24 | phix | mcf3782: replace it with better cells |
| 23:26.23 | p3nguin | There's nothing to replace if he needs an extra battery pack. |
| 23:27.15 | p3nguin | The last time I looked at getting new battery packs for my old 14.4V DeWalt, they were around $50 each. |
| 23:34.09 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ssqvmkcebopbjtlm) |
| 23:42.27 | slav3_kitten | p3nguin, better cells will improve runtime |
| 23:42.27 | slav3_kitten | maybe eliminate the need for a second battery |
| 23:42.39 | slav3_kitten | i did that with all my makita stuff |
| 23:42.51 | WIMPy | You still cannot charge while working. |
| 23:43.03 | gusto | p3nguin: in what bin? |
| 23:43.30 | p3nguin | trash bin, trash can, trash dumpster, garbage depository |
| 23:45.04 | slav3_kitten | WIMPy, if your doing around the house work an you increase your runtime 25 to 50 percent you may not need to charge while working |
| 23:45.11 | slav3_kitten | then again i really prefer wired power tools |
| 23:45.12 | gusto | p3nguin: should i call you then? |
| 23:45.44 | p3nguin | What are you talking about? |
| 23:45.55 | slav3_kitten | do up your exension cord in a chain snippet then just plug in an walk off |
| 23:46.51 | slav3_kitten | its a type of "knot" iirc that's the proper name for it but you like tie your rappel rope so your weighted bag allows it to just pay out flawlessly whe you toss the bag |
| 23:47.19 | slav3_kitten | you can tie power cords so that you plug them in an walk off with the other end and it pays out properly as you go |
| 23:47.40 | slav3_kitten | really shortens the length you stow as well |
| 23:49.08 | slav3_kitten | i'm going to shut up an drink gatorade to recover now |
| 23:50.31 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-78-180.bois.qwest.net) |
| 23:51.08 | slav3_kitten | sinnet*** |
| 23:52.07 | slav3_kitten | little out of it sorry |
| 23:54.10 | dijib | ya sorry the oil pan gasket went |