IRC log for #asterisk on 20121112

`00:01.59`	`*** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net)`
`00:05.02`	`ChannelZ`	`mathi: it works for me. Did you reload logger and cli or restart asterisk?`
`00:05.45`	`ChannelZ`	`actually you'd have to restart asterisk I think so that the root process would pick up the log level in cli.conf`
`00:06.28`	`jeffspeff`	`you can't put variables in a extension pattern can you? like exten=${myvar}XXXX,1,foo()`
`00:06.42`	`ChannelZ`	`don't think so`
`00:07.02`	`jeffspeff`	`k`
`00:07.02`	`WIMPy`	`The log level in cli.conf probably doesn;t have any influence on log files (any more).`
`00:07.09`	`ChannelZ`	`It does here`
`00:07.29`	`WIMPy`	`jeffspeff: Yes, you can.`
`00:07.32`	`ChannelZ`	`actually... let me see`
`00:07.53`	`jeffspeff`	`WIMPy, really? I'm getting an auto-fallthrough when i try`
`00:08.41`	`WIMPy`	`jeffspeff: Are you missing the _ or was that a C&P error?`
`00:09.05`	`jeffspeff`	`I'm missing the _`
`00:09.06`	`jeffspeff`	`:P`
`00:09.49`	`jeffspeff`	`i wasn't sure if i should even still put it in there. i thought that * only expected that in front of XXXX`
`00:10.16`	`jeffspeff`	`or whatever your pattern is, but i didn't think it would recognize a variable as the same type of pattern`
`00:10.17`	`WIMPy`	`Yes, and you have XXXX in there.`
`00:10.43`	`WIMPy`	`Well, for any pattern, i.e. znx.! .`
`00:11.00`	`WIMPy`	`And [], off course.`
`00:11.46`	`jeffspeff`	`if i set a var in one context, that var is carried to the next context right? or not?`
`00:11.56`	`jeffspeff`	`same call, but different context`
`00:12.06`	`jeffspeff`	`which makes it a channel variable?`
`00:12.12`	`WIMPy`	`variables are global or on a channel.`
`00:12.32`	`WIMPy`	`Context doesn't matter.`
`00:13.17`	`jeffspeff`	`so, how do i set a channel variable? i'm just using the exten=XXXXX,1,Set(foo=123) method`
`00:13.27`	`jeffspeff`	`so would foo then be a channel or global?`
`00:13.31`	`WIMPy`	`That's one.`
`00:14.00`	`WIMPy`	`Unless you define it in your dialplan under [global] or use the GLOBAL() function, it's a channel variable.`
`00:14.08`	`jeffspeff`	`ok`
`00:16.22`	`mathi`	`ChannelZ, it works now thanks`
`00:16.55`	`*** join/#asterisk wonderworld (~w@dsdf-4db530da.pool.mediaWays.net)`
`00:17.10`	`mathi`	`my next problem is that HANGUPCAUSE_KEYS() returns nothing, though I have Asterisk 11.0.0`
`00:19.17`	`*** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain)`
`00:21.13`	`mathi`	`same => n,Set(HANGUPCAUSE_STRING=${HANGUPCAUSE_KEYS()})`
`00:21.38`	`mathi`	`-- Executing [s@subHangUp:3] Set("SIP/ipbx-soderwi-00000005", "HANGUPCAUSE_STRING=") in new stack`
`00:21.53`	`ChannelZ`	`Is that after a dial?`
`00:21.59`	`mathi`	`in the hang up handler`
`00:22.23`	`mathi`	`when I hang up the phone, this dialplan is executed`
`00:22.38`	`mathi`	`${HANGUPCAUSE_KEYS()} returns empty`
`00:22.41`	`WIMPy`	`Oh, mathi is in full 11 fever :-)`
`00:23.43`	`mathi`	`WIMPy, yeap.... will you help? :P`
`00:24.22`	`WIMPy`	`I haven't found a use case for those features, yet.`
`00:26.21`	`ChannelZ`	`well it's blank here too but I'm not sure what the use is quite yet`
`00:27.33`	`mathi`	`ChannelZ, you have asterisk 11?`
`00:27.35`	`mathi`	`it's a new feature`
`00:27.36`	`*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)`
`00:27.40`	`ChannelZ`	`yes`
`00:27.55`	`ChannelZ`	`It works immediately after a dial()`
`00:28.54`	`mathi`	`ChannelZ, ah, so it works only if I transfer using Dial() ?`
`00:28.56`	`*** join/#asterisk widler (47c8d864@gateway/web/freenode/ip.71.200.216.100)`
`00:28.58`	`ChannelZ`	`whether it's a bug or not that it doesn't in the hangup extension, or is getting cleared by something else you're doing prior, that I don't know.`
`00:29.13`	`widler`	`hello all`
`00:29.19`	`widler`	`I'm new to asterisk`
`00:29.33`	`widler`	`and need much help`
`00:30.31`	`carrar`	`widler, check this out then`
`00:30.32`	`carrar`	`~book`
`00:30.32`	`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`
`00:32.02`	`widler`	`thanks. i install it ok. my gui works, but I cant seem to add a dialplan`
`00:32.22`	`mathi`	`ChannelZ, it works only after a Dial() for me, in other cases I have an empty string returned`
`00:32.28`	`ChannelZ`	`Oh no, the G word`
`00:33.04`	`widler`	`iax2 reload don't work`
`00:33.06`	`slav3_kitten`	`it's a good book`
`00:33.13`	`ChannelZ`	`mathi: maybe the channels are already destroyed by the time the h exten is executed, I don't know.`
`00:33.51`	`widler`	`what's the h extension and how do i get the channels back`
`00:34.25`	`ChannelZ`	`I was talking to mathi`
`00:34.36`	`*** join/#asterisk Nemus (~Nemus@unaffiliated/nemus)`
`00:34.41`	`Nemus`	`can asterisk play midi files?`
`00:34.48`	`ChannelZ`	`oh dear god`
`00:35.06`	`ChannelZ`	`please puncture my ear holes`
`00:35.21`	`Nemus`	`okay`
`00:35.25`	`widler`	`ok.`
`00:35.29`	`Nemus`	`stand still`
`00:38.29`	`mathi`	`ChannelZ, WIMPy, ok all these questions about logging and hangupcause... is because I am trying to debug something very mysterious happening. At one point, my IVR asks the user to enter his mobile phone (plus or less 10 digits). And from time to time... the user gets disconnected while entering the numbers. And I can't explain why`
`00:39.06`	`*** join/#asterisk wonderworld (~w@dsdf-4db530da.pool.mediaWays.net)`
`00:39.30`	`p3nguin`	`jeffspeff: If you set a variable in an extension, without setting it as a GLOBAL, it is only valid on the channel that is executing that extension. When the channel goes away, the variable goes away. Also, if that channel spawns other channels, the variable is not carried to new channel "levels" unless you use underscores to activate the possibility of inheritance of the variable.`
`00:39.51`	`WIMPy`	`mathi: Onless the normal verbose 3 output tells you anything, you will have to dig at the protocoll level.`
`00:40.18`	`mathi`	`WIMPy, I have versbose 300 :-) ... How do I dig at the prtocol level ?`
`00:40.42`	`mathi`	`(because all the logs say is [Nov 12 00:52:43] VERBOSE[9023][C-00000004] app_read.c: -- User disconnected)`
`00:40.49`	`WIMPy`	`Usually with <wahteverprotocoll> set debug ...`
`00:41.06`	`mathi`	`aahhh`
`00:41.24`	`mathi`	`WIMPy, and how can I log that into my file, what level do I need to set ?`
`00:41.25`	`WIMPy`	`That surely doesn't look like it has anythig to do with your dialplan.`
`00:41.46`	`WIMPy`	`What cahanneltype are tey comming in?`
`00:41.51`	`WIMPy`	`they`
`00:41.56`	`mathi`	`WIMPy, no... I have done many Read() for many inputs, I know how to use it, and the code to ask the mobile is very simple....`
`00:42.05`	`mathi`	`WIMPy, SIP`
`00:42.51`	`WIMPy`	`Then it might be easier to use wireshark.`
`00:43.59`	`mathi`	`WIMPy, but actually there is a first server (with a Digium card) that that is like a trunk between my server (I have no cards) and the caller`
`00:45.10`	`WIMPy`	`So they come in via a phone line in to one Asterisk and then you pass them on via SIP to another Asterisk where you see them diconnect?`
`00:46.20`	`jeffspeff`	`p3nguin, in my beginning context, the dialplan grabs some info from a database then cuts that info, setting different vars. of the things it grabs and sets is a customer id and a destination. i then use those for my goto priority. which is getting concatenated correctly. i verified this through verbose output in cli. so, it's supposed to go to EXTENSIONS,customerid+destination,1 which is equaling EXTENSIONS,11234,1`
`00:46.23`	`mathi`	`yes, (but tey disconnect sometimes only when they need to enter their phone number (ptobably because they need to enter many numbers), there are no problems with other inputs asking 1 or 2 numbers)`
`00:46.55`	`jeffspeff`	`then in my EXTENSIONS context i have exten=_${CoID}XXXX,1,Verbose(blah blah) etc.`
`00:47.06`	`mathi`	`WIMPy, pstn => some server => SIP => my server`
`00:47.24`	`jeffspeff`	`${CoID} is the actual var being set in the original context where the data is pulled`
`00:47.45`	`jeffspeff`	`but i keep getting an auto-fallthrough on the channel after it does the goto`
`00:47.54`	`WIMPy`	`mathi: You should be debugging on the other server then.`
`00:48.08`	`mathi`	`WIMPy, you suspect something on the other server?`
`00:48.20`	`jeffspeff`	`it leads me to believe that ver 11 doesn't do variables in the exten patterns`
`00:48.45`	`jeffspeff`	`otherwise exten=_${CoID}XXXX,1,foo() would work`
`00:48.50`	`ChannelZ`	`do a SIP debug and see if you just get a BYE from the 'other server' which would tell you`
`00:48.56`	`WIMPy`	`mathi: Either there or even further away.`
`00:49.22`	`WIMPy`	`jeffspeff: It works for me. But maybe you should show us a failed call.`
`00:49.53`	`jeffspeff`	`ok, just a sec`
`00:49.57`	`p3nguin`	`I don't think "extensions" is a very good name for a context, considering every context is full of extensions (that's what contexts are).`
`00:51.10`	`mathi`	`ChannelZ, the command is "core set debug on" ?`
`00:51.51`	`WIMPy`	`mathi: sip set debug peer <otherserver>`
`00:51.59`	`jeffspeff`	`here's the cli output and the dialplan http://pastebin.com/NxBUv9JJ`
`00:52.39`	`jeffspeff`	`p3nguin, this is just a test box, no where near any of my production systems... i'm just toying with different ideas and databasing, etc. just trying to get better at asterisk. :)`
`00:53.11`	`p3nguin`	`My comment was for the purpose of getting better with asterisk.`
`00:53.20`	`mathi`	`WIMPy, No such command 'sip set debug <ipbx-wizdo>'`
`00:53.28`	`mathi`	`OOPS`
`00:53.33`	`jeffspeff`	`p3nguin, oh,`
`00:53.46`	`jeffspeff`	`well there's only 2 contexts. :)`
`00:53.57`	`p3nguin`	`I try to encourage best practice.`
`00:54.40`	`WIMPy`	`And what's the value of CoID?`
`00:54.41`	`jeffspeff`	`my production system has so many contexts in a huge dialplan i finally just split the dp into multiple .conf files and used file includes within extensions.conf`
`00:54.50`	`jeffspeff`	`WIMPy, 1`
`00:55.02`	`WIMPy`	`Oh never mind.`
`00:55.06`	`mathi`	`WIMPy, Unable to get IP address of peer 'ipbx-wizdo'`
`00:55.07`	`jeffspeff`	`line 40`
`00:55.35`	`WIMPy`	`Why do you wan to go to an extension that you just invented on the fly?`
`00:55.48`	`p3nguin`	`mathi: Set it manually. sip set debug ip <addr>`
`00:56.13`	`mathi`	`p3nguin, ahhh thanks, :-) I will try, just need to figure out how to find its ip, I quite forgot`
`00:56.35`	`jeffspeff`	`WIMPy, no real reason right now; just trying to get some different logic working; if it's possible`
`00:56.39`	`WIMPy`	`jeffspeff: I guess the variable will only be parsed at dialplan load time. But I don't see how that's a restriction.`
`00:57.14`	`jeffspeff`	`the var gets parsed when the call comes through, it's in a database that gets queried based on the DID of the incoming call`
`00:57.25`	`WIMPy`	`You're dtrying to match what you're going to send. That doesnt really make sense.`
`00:58.05`	`p3nguin`	`I don't think this can ever work.`
`00:58.08`	`WIMPy`	`That's like a 'if true then'.`
`00:58.27`	`jeffspeff`	`i'm not matching what i'm sending, i want to go to EXTENSION,11234,1 but it doesn't work in the EXTENSIONS context`
`00:58.28`	`p3nguin`	`If the variable does not exist when pbx_config loads the dial plan, the extensions will be invalid.`
`00:58.45`	`jeffspeff`	`pretend that those vars are anything other than the numbers they are, it should work`
`00:58.59`	`WIMPy`	`Not invalid, just without that part.`
`00:59.08`	`WIMPy`	`It doesn't matter.`
`00:59.35`	`jeffspeff`	`WIMPy, so, the channel variable of ${CoID} can't be called within an extension pattern`
`00:59.45`	`p3nguin`	`I would make an extension that matches more things, and then use the values of the variables to goto something else which also matches more things.`
`00:59.54`	`WIMPy`	`Only if it's global.`
`01:00.11`	`WIMPy`	`But as stated, I don't see that as an issue.`
`01:00.16`	`p3nguin`	`If the value of CoID is always an integer, create an extension with a pattern that matches the possible integers.`
`01:00.28`	`jeffspeff`	`p3nguin, you're missing the point here`
`01:00.57`	`p3nguin`	`No, I'm not. You're trying to do something that falls outside the realm of possibilities of dial plan.`
`01:00.58`	`jeffspeff`	`there's not some type of project i'm working on, or problem to solve. i'm simply trying to call a var in an exten pattern that i set in a previous context.`
`01:01.06`	`WIMPy`	`jeffspeff: No you making a point that doesn't exist.`
`01:01.40`	`WIMPy`	`Contexts still don't matter.`
`01:01.59`	`p3nguin`	`You can go to a place in dial plan that does reflect the value of your variable.`
`01:02.04`	`jeffspeff`	`the point is, as we've now learned, you can not create a channel var and then call that var within an extension pattern`
`01:02.07`	`p3nguin`	`But the problem is how you're trying to get there.`
`01:03.18`	`p3nguin`	`If I knew the possible range if data that your variable could contain, I'd be able to devise an appropriate dial plan to make it happen.`
`01:03.20`	`jeffspeff`	`fully aware there are 1000^99999999999999 ways to do this, but like i said, i'm just playing around, trying new things. this obviously doesn't work as expected`
`01:04.10`	`jeffspeff`	`p3nguin, the problem is easily solved with using any regular expressions of extension patterns other than custom variables.`
`01:05.23`	`WIMPy`	`You'd better construct a use case for that Goto in the first place. Otherwise there isn't much to comment on.`
`01:06.00`	`*** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net)`
`01:10.09`	`p3nguin`	`Is there any such creation as a realtime dial plan?`
`01:11.25`	`p3nguin`	`jeffspeff: dialplan show EXTENSIONS`
`01:11.40`	`p3nguin`	`What extension(s) does it show were loaded?`
`01:11.49`	`jeffspeff`	`p3nguin, we're done with this`
`01:11.55`	`p3nguin`	`s/were/was\/were/`
`01:12.09`	`p3nguin`	`Oh.`
`01:12.16`	`jeffspeff`	`over an hour ago, i asked if it worked like i was wanting to. you said no, wimpy said yes, i tried, it doesn't work. moving on`
`01:12.24`	`p3nguin`	`moves on, too`
`01:12.29`	`jeffspeff`	`thanks for your help.`
`01:12.57`	`p3nguin`	`You did make me curious if there is a way around the limitation.`
`01:13.02`	`WIMPy`	`Well I didn't imagine, you would try that with a channel variable.`
`01:14.57`	`jeffspeff`	`WIMPy, now imagine the possiblities that become available if you could do that. and it would still be logical in the dialplan too`
`01:15.16`	`WIMPy`	`I don't see any.`
`01:15.20`	`jeffspeff`	`but there are many many ways to accomplish the same task`
`01:15.45`	`WIMPy`	`As already said, constucting a wildcard for something you already know seems completely pointless to me.`
`01:16.03`	`jeffspeff`	`who said I knew the values of that?`
`01:16.18`	`jeffspeff`	`the value of CoID can change to be a large number`
`01:16.18`	`WIMPy`	`You set them just before.`
`01:16.34`	`jeffspeff`	`but the extensions context doesn't know what i just set it to`
`01:16.37`	`WIMPy`	`Where? When?`
`01:17.16`	`WIMPy`	`I'm not talking about the number one you've got the. It's about the fact that you set that variable on the same channel. The value doesn;t matter.`
`01:17.52`	`WIMPy`	`A Goto() doesn't cahnge your variables.`
`01:18.12`	`jeffspeff`	`if you have multi-tenant environment where people can be typing thing in and naming menus and extensions, etc. then you need to differentiate based on the customer name or number`
`01:18.40`	`WIMPy`	`Yes, but that doesn't change per call.`
`01:19.06`	`WIMPy`	`And I'm still unsure if that makes sense.`
`01:19.07`	`jeffspeff`	`that way the same [MAIN-INBOUND] and [USER-EXTENSIONS] contexts can be used for all clients without any problems`
`01:19.13`	`[TK]D-Fender`	`teknoprep: ...`
`01:19.24`	`[TK]D-Fender`	`jeffspeff: each CALL's cvariables are complete separate from one another`
`01:19.31`	`[TK]D-Fender`	`jeffspeff: there is NO sharing between channels`
`01:19.40`	`WIMPy`	`That would only make sense if they all had the same extensions.`
`01:19.45`	`jeffspeff`	`right, not trying to share between channels`
`01:20.15`	`[TK]D-Fender`	`jeffspeff: and context's have precisely ZERO impact on variable scope. * doesn't HAVE any sense of variable scope. DOES. NOT. EXIST.`
`01:20.30`	`[TK]D-Fender`	`jeffspeff: Your "protections" are pretty much irrelevant`
`01:21.20`	`jeffspeff`	`WIMPy, what if more than one customer wants to have extensions like 1XXX ?`
`01:21.34`	`jeffspeff`	`you need a way to dynamically differentiate them and there menus, etc`
`01:21.38`	`[TK]D-Fender`	`jeffspeff: put those in separate contexts`
`01:21.41`	`[TK]D-Fender`	`^`
`01:22.11`	`jeffspeff`	`[TK]D-Fender, and if i don't want to hop in and manually write new contexts each time i get a client?`
`01:22.33`	`WIMPy`	`Yes. Goto(cusomer,extension,1) or something. Or skip the Goto all together and just use CoID, wiuch you already have.`
`01:22.57`	`[TK]D-Fender`	`jeffspeff: then you're going to have extension overlap and in EACH of them you'll have to confirm who your caller is and what that # they dialed really does.`
`01:23.09`	`WIMPy`	`I can;t make any sense of that example.`
`01:23.34`	`mathi`	`ChannelZ, I had to call about 20 times to get the error .... I see a BYE that has been sent, what now?`
`01:24.02`	`WIMPy`	`mathi: "sent" as in which direction?`
`01:24.09`	`jeffspeff`	`[TK]D-Fender, i'm handling that in the beginning inbound context. it queries a database to get the info, there's many ways to accomplish this, but you just can't do it the first way i was trying`
`01:24.42`	`*** join/#asterisk DarthExpeditor (~IceChat9@rrcs-71-43-76-226.se.biz.rr.com)`
`01:26.01`	`[TK]D-Fender`	`jeffspeffthat way the same [MAIN-INBOUND] and [USER-EXTENSIONS] contexts can be used for all clients without any problems <-- if you're expecting customers to code in their own dialplans and have potentially identical names then that is just not going to work.`
`01:26.31`	`mathi`	`WIMPy, it says from my server, via the server with card, and to my server (exactly as a usual hang up...)`
`01:32.36`	`*** join/#asterisk dgeary2 (~david@120.21.101.81)`
`01:36.48`	`*** join/#asterisk Kyril (~Kyril@fedora/Kyril)`
`01:42.01`	`mathi`	`WIMPy, I guess I need to start looking on the first server... but do you have any ideas why that server would sometimes send me a packet BYE at that moment (when the user enters more than 3-4 numbers) ?`
`01:43.18`	`WIMPy`	`The only thing that springs to mind would be "features" being enabled and it accidentally detecting the hangup feature code.`
`01:43.39`	`*** join/#asterisk droemel (~droemel@p4FCAD195.dip.t-dialin.net)`
`01:44.51`	`mathi`	`WIMPy, and what is that hangup feature??`
`01:45.14`	`WIMPy`	`The one that's configured in features.conf.`
`01:45.30`	`*** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net)`
`01:47.43`	`mathi`	`WIMPy, do you mean this? parkedcallhangup`
`01:48.52`	`WIMPy`	`"disconnect" would be the main suspect.`
`01:51.15`	`*** join/#asterisk widler (47c8d864@gateway/web/freenode/ip.71.200.216.100)`
`01:51.41`	`widler`	`can anyone tell me how to fix this make: *** [pridump.o] Error 1`
`01:52.30`	`WIMPy`	`widler: We need some more lines above that.`
`01:52.35`	`WIMPy`	`~pb`
`01:52.35`	`infobot`	`A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude.`
`01:53.02`	`widler`	`asteriskpbx@msi ~/src/asterisk-complete/asterisk $ cd libpri/ asteriskpbx@msi ~/src/asterisk-complete/asterisk/libpri $ make gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c gcc -g -Wall -Werror -Wstrict-pr`
`01:53.13`	`p3nguin`	`Better to paste more stuff than needed to figure it out as opposed to not pasting enough.`
`01:54.45`	`WIMPy`	`widler: Do you have any ISDN interfaces at all?`
`01:56.16`	`widler`	`no`
`01:56.18`	`widler`	`i don't`
`01:56.35`	`WIMPy`	`Then you have no use for libpri anyway.`
`01:56.46`	`widler`	`ok`
`01:56.48`	`widler`	`thanks`
`02:07.09`	`widler`	`I couldn't get SQLite3 in the repository`
`02:08.08`	`widler`	`i did apt-get intall SQLite3 and it couldnt find it.`
`02:08.18`	`widler`	`is there and alternative`
`02:08.52`	`p3nguin`	`Is apt-get case sensitive?`
`02:09.06`	`p3nguin`	`apt-cache search sqlite`
`02:09.11`	`p3nguin`	`See what it finds.`
`02:09.49`	`widler`	`it is case sensitive`
`02:10.02`	`widler`	`i'm just gonna search`
`02:10.16`	`carrar`	`apt-get -i`
`02:10.26`	`slav3_kitten`	`widler, listen to p3nguin`
`02:10.48`	`slav3_kitten`	`also what distro`
`02:11.19`	`widler`	`thanks`
`02:12.04`	`drmessano`	`apt-get install libsqlite3-dev`
`02:12.08`	`slav3_kitten`	`because sqlite3 is debian/ubuntu repos for sure`
`02:12.33`	`slav3_kitten`	`and yea you need lib for development`
`02:12.48`	`slav3_kitten`	`as drmessano beat me to the punch by saying`
`02:14.19`	`widler`	`that worked just fine`
`02:15.57`	`widler`	`as you all can see, i'm new to asterisk`
`02:16.06`	`widler`	`and new to linuz`
`02:16.23`	`slav3_kitten`	`you've chosen wisely`
`02:16.40`	`widler`	`but I installed asterisk earlier and the asterisk gui`
`02:16.56`	`p3nguin`	`I would have thought that apt-get install asterisk would have installed any necessary dependencies quite elegantly.`
`02:17.20`	`widler`	`can i add all my users, dialplans and sip and iax from the gui`
`02:17.29`	`slav3_kitten`	`gui?`
`02:17.43`	`widler`	`asterisk-gui`
`02:18.04`	`p3nguin`	`Wrong channel for that.`
`02:18.06`	`slav3_kitten`	`i don't think that's officially supported anymore`
`02:18.37`	`widler`	`i know, i just thought someone might know`
`02:19.23`	`p3nguin`	`If you want to configure asterisk, we can help you here. If you want to use another piece of software to configure asterisk, it's best to ask someone else.`
`02:20.31`	`slav3_kitten`	`so i actually have a question`
`02:20.53`	`slav3_kitten`	`what would you suggest to lock outbound calling to the uk. i was thinking an ivr`
`02:21.03`	`widler`	i see. asteriskpbx@msi ~/src/asterisk-complete/asterisk/1.8 $ sudo make make[1]: Entering directory `/home/asteriskpbx/src/asterisk-complete/asterisk/1.8' make[5]: ccar: Command not found make[5]: * [../lib/libpj-x86_64-unknown-linux-gnu.a] Error 127 make[4]: * [pjlib] Error 2 make[3]: * [all] Error 1 make[2]: * [pjproject] Error 2 make[1]: *** [res] Error 2 make[1]: Leaving directory `/home/asteriskpbx/src/asterisk-complet
`02:21.09`	`widler`	`sorry`
`02:21.27`	`slav3_kitten`	`widler, so you know. current version is 11. 1.8 is quite old`
`02:21.34`	`p3nguin`	`An IVR? Like, "if you want to make a call to the UK, press 1 now."`
`02:21.47`	`p3nguin`	`1.8 is not old, it's current.`
`02:21.53`	`widler`	`it actually is 11`
`02:22.05`	`slav3_kitten`	`more like a password to call out to the uk`
`02:22.19`	`p3nguin`	`1.8.18.0 was just released a few days ago.`
`02:22.41`	`slav3_kitten`	`1.8 lts iirc`
`02:22.48`	`p3nguin`	`so is 11`
`02:23.04`	`widler`	`the folder is named 1.8`
`02:23.49`	`slav3_kitten`	`shrugs`
`02:23.53`	`p3nguin`	`I would probably use well-devised extension patterns which match either what I do want to be allowed, match what is not allowed, or both, with appropriate things happening when each gets called.`
`02:24.56`	`p3nguin`	`For the North America, it's easy: _NXXNXXXXXX matches all of what is allowed. Anything else would be disallowed.`
`02:25.02`	`slav3_kitten`	`i was thinking dial a uk number an it prompts for a password then dials if you got correct`
`02:25.22`	`p3nguin`	`If you have a similar numbering plan, patterns should be easy.`
`02:26.01`	`*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)`
`02:26.58`	`slav3_kitten`	`thing is i know the uk pricing crap but other people in the house don't, i'd hate for say my sister to call a uk number that's 25 cents a minute without making a hugely convoluted dial plan that allows or disallows all the uk area codes`
`02:26.59`	`*** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com)`
`02:27.07`	`slav3_kitten`	`for all the*`
`02:27.22`	`slav3_kitten`	`i mean i guess i could`
`02:27.55`	`p3nguin`	`You can query a db with the rates in it on each call.`
`02:28.25`	`slav3_kitten`	`and now we are above my pay grade lol`
`02:28.34`	`slav3_kitten`	`that's a great idea however`
`02:30.20`	`p3nguin`	`Some dial plan logic could query the db rates and, if the rate is above N cents, prompt for a password to override a block on expensive numbers.`
`02:33.18`	`slav3_kitten`	`that's so far beyond what i know how to do at this point...`
`02:33.29`	`slav3_kitten`	`but we all start someplace`
`02:33.49`	`p3nguin`	`I had a plan to do that for myself, but I never got around to it.`
`02:34.24`	`p3nguin`	`I was going to do rate-based call routing.`
`02:34.49`	`slav3_kitten`	`nod`
`02:48.09`	`WIMPy`	`And someone would have to keep the rates up to date all the time.`
`02:48.50`	`slav3_kitten`	`oh that would be a downside`
`02:48.56`	`*** join/#asterisk serafie (~erin@76.73.167.231)`
`02:49.02`	`slav3_kitten`	`i could hire some minority to do it though`
`02:50.12`	`p3nguin`	`I was going to just download the rate file periodically from the provider.`
`02:50.23`	`p3nguin`	`Once a month or so.`
`02:50.47`	`WIMPy`	`If you only have one to choose from...`
`02:50.47`	`slav3_kitten`	`then convert it from a csv into db?`
`02:51.53`	`p3nguin`	`When I wa going to set it up for myself, I had two providers to work with. One provider had a fixed rate for all calls, the other had various rates within the same area code.`
`02:52.50`	`slav3_kitten`	`are or country code?`
`02:52.52`	`slav3_kitten`	`area*`
`02:53.20`	`p3nguin`	`The one with various rates provided a rate file. I was going to check the rate db for each call. If the rate was lower than the fixed-rate carrier, route it through that provider. If it was higher, route it through the fixed-rate carrier.`
`02:53.36`	`slav3_kitten`	`ah`
`02:54.06`	`p3nguin`	`Very basic logic to make it happen.`
`02:54.28`	`p3nguin`	`You could use the same concept, prompting for a PIN if the rate exceeds some predetermined amount.`
`02:55.01`	`p3nguin`	`If the rate is less than the predetermined amount, call normally without a PIN.`
`02:55.18`	`p3nguin`	`Update the rate db monthly.`
`02:55.57`	`WIMPy`	`That get get extremely expensive.`
`02:56.42`	`WIMPy`	`Best to only use providers that announce the price at the beginning of the call. At least you know when you're being ripped off then.`
`02:57.51`	`p3nguin`	`I guess I don't get out much. Of the half dozen North American providers I have relationships with, zero of them announce the rate of the call.`
`02:58.40`	`*** join/#asterisk tonylyliu (~tonylyliu@183.3.153.248)`
`03:19.50`	`*** join/#asterisk LiuYan (~LiuYan@211.154.128.171)`
`03:41.43`	`*** join/#asterisk DarthExpeditor (IceChat9@173.46.183.219)`
`03:49.52`	`slav3_kitten`	`dijib, no conference call tonight? :P`
`04:05.51`	`*** part/#asterisk LiuYan (~LiuYan@211.154.128.171)`
`04:25.08`	`*** join/#asterisk deo_ (~deo@58.71.19.178)`
`04:36.07`	`SeRi`	`slav3_kitten: I am about to jump in in a few min`
`04:40.22`	`slav3_kitten`	`what's the address?`
`04:41.46`	`*** join/#asterisk mintos (mvaliyav@nat/redhat/x-mmoldehqrkzxnbof)`
`04:41.51`	`*** join/#asterisk ChrisInSydney (~Administr@202-129-83-200.perm.iinet.net.au)`
`04:43.58`	`ChrisInSydney`	`g'day all`
`04:44.17`	`slav3_kitten`	`yo chris`
`04:45.06`	`ChrisInSydney`	`Can anyone help me with the Dial() app and contexts for t & T option transfers. How do I know which context I am transferring from ??`
`04:45.17`	`ChrisInSydney`	`hey slav3_kitten`
`04:46.00`	`slav3_kitten`	`my noob answer would be read the book :\|`
`04:46.12`	`slav3_kitten`	`then again i'm on page 200 something`
`04:48.19`	`ChrisInSydney`	`I am jumping between contexts using Gotos etc, and it doesn't want to work from the current context.`
`04:55.56`	`slav3_kitten`	`yea i'm still on trying to get voicemail working right so i'm not the best one to answer i don't think`
`04:59.26`	`ChrisInSydney`	`<PROTECTED>`
`04:59.43`	`ChrisInSydney`	`yet Im in another context when the dial app is made`
`04:59.46`	`ChrisInSydney`	`bugger`
`05:00.07`	`slav3_kitten`	`you can include the context to be dialed to`
`05:00.34`	`*** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri)`
`05:00.51`	`ChrisInSydney`	`more app_dial.c hacking. Loks like I can get it to behave if I Dial(Local/something,,t) and then Dial form there but thats ugly and crap`
`05:01.51`	`ChrisInSydney`	`call comes in, extension that is dialled is an external number Dial(SIP/some-trunk/mymobile)`
`05:01.52`	`p3nguin`	`It will match the peer's context.`
`05:02.01`	`ChrisInSydney`	`Transfer keys are pressed`
`05:02.21`	`p3nguin`	`If the peer's context is A, but the call is made to the peer in B, you can't transfer to another extension in B.`
`05:02.22`	`ChrisInSydney`	`Allison says "Transfer"`
`05:02.34`	`ChrisInSydney`	`Then I punch a number`
`05:02.52`	`p3nguin`	`I fought with this earlier today.`
`05:03.26`	`p3nguin`	`Your call is going to be going to the context defined in the 'some-trunk' peer entry.`
`05:03.32`	`ChrisInSydney`	`p3nguin: Bugger so it will work from the entry in the sip.conf for that peer, not the context you are currently in`
`05:03.52`	`p3nguin`	`For me, that was the inbound context rather than the phones context.`
`05:03.59`	`ChrisInSydney`	`Can I swear in this channel ?`
`05:04.12`	`p3nguin`	`It is a problem, and I have not figured out how to work around it yet.`
`05:04.32`	`ChrisInSydney`	`swearing in this channel or ,t contexts Xp`
`05:04.41`	`ChrisInSydney`	`OK time to hack the code`
`05:04.58`	`ChrisInSydney`	`thanks for the heads up`
`05:05.14`	`ChrisInSydney`	`p3nguin; You have saved a few hours of my life`
`05:05.36`	`p3nguin`	`Odd how that is the exact configuration I was testing with.`
`05:05.55`	`ChrisInSydney`	`I'll see what I can hack together in the app_dial.c`
`05:06.35`	`ChrisInSydney`	`got custom prompts / priv-callerintro channel vars working with p&P options`
`05:07.06`	`ChrisInSydney`	`helpful to have a custom dir for the files and option 9 deletes the intro recording and connects the call`
`05:10.56`	`ChrisInSydney`	`This hack isnt going to be so straight forward`
`05:13.11`	`slav3_kitten`	`when are they ever`
`05:13.59`	`ChrisInSydney`	`some have been a couple of lines of code and eveything has worked`
`05:14.52`	`ChrisInSydney`	`app_pickup instead of a full string compare, changing it to a partial string compare and you can pick up groups easily`
`05:15.01`	`ChrisInSydney`	`thats a simple one`
`05:15.53`	`ChrisInSydney`	`p3nguin: I cant seem to see where the blind transfer thingy takes place in app_dial.c Looks like it might take place in res_features.c have you looked into that ?`
`05:16.18`	`ChrisInSydney`	`or is this a ? for #asterisk-dev ?`
`05:18.10`	`slav3_kitten`	`shrugs`
`05:19.09`	`ChrisInSydney`	`stares into the code hoping for a "neo" moment but can't help thinking there is more to life......`
`05:19.39`	`slav3_kitten`	`i'm just sitting in bed trying to get lung cancer an shoutcast music on hold`
`05:19.51`	`ChrisInSydney`	`:D`
`05:20.17`	`ChrisInSydney`	`dont like smoking tobacco`
`05:20.59`	`slav3_kitten`	`wht do you smoke?`
`05:21.14`	`ChrisInSydney`	`ahhhh`
`05:21.31`	`Maliuta`	`prepares the call the the NSW police force ;)`
`05:21.41`	`ChrisInSydney`	`that may be for another channel`
`05:21.47`	`ChrisInSydney`	`#bob-marley`
`05:21.53`	`slav3_kitten`	`oic`
`05:22.04`	`Maliuta`	`#rastaSkank ???`
`05:22.09`	`ChrisInSydney`	`looks forward to a call from the cops as eveyone knows that they always have the best gear`
`05:22.16`	`ChrisInSydney`	`:D`
`05:22.27`	`slav3_kitten`	`our cops are`
`05:22.34`	`slav3_kitten`	`well they leave a lot to be desired`
`05:22.52`	`slav3_kitten`	`they used tear gas one point locally, no one thought to bring the gas masks`
`05:22.52`	`ChrisInSydney`	`actually, I only say that to illicit an response from people`
`05:23.04`	`slav3_kitten`	`well none of the police, guy in the house had one`
`05:23.21`	`ChrisInSydney`	`slav3_kitten: Where are you again ??`
`05:23.24`	`Maliuta`	`hmm I wonder if ChrisInSydney spent time attached to the Catholic church in The Hunter Valley ;P`
`05:23.33`	`slav3_kitten`	`rural america`
`05:23.51`	`Maliuta`	`slav3_kitten: well then, there's your problem ;)`
`05:24.22`	`ChrisInSydney`	`bends a 2nd up to a maj 3 followed by a 5 8 interval`
`05:24.38`	`ChrisInSydney`	`(musical notation)`
`05:24.41`	`slav3_kitten`	`they at one point closed the local PD, so someone broke in an stole all the shotguns and a number of radios`
`05:24.50`	`ChrisInSydney`	`your kidding`
`05:24.54`	`slav3_kitten`	`not at all`
`05:24.57`	`Maliuta`	`ROFL`
`05:25.14`	`ChrisInSydney`	`Malitua: I did spend alot of time in catholic churches`
`05:25.17`	`slav3_kitten`	`24 hours after disbanding the local PD, the station was robbed blind, week later we had 3 new cops`
`05:25.40`	`ChrisInSydney`	`but fortunately I was never offered the eucharist in that "special" third form ;-)`
`05:25.47`	`Maliuta`	`although there have been cases of revolvers being stolen from a safe at a police station not far from here. That was while they were transitioning to the glocks`
`05:26.02`	`Maliuta`	`(and shooting them selves in the arse alot)`
`05:26.32`	`Maliuta`	`ChrisInSydney: may still have to refer you to the enquiry ;)`
`05:26.36`	`slav3_kitten`	`that's entirely a training shortcoming, glocks are safe if you practice basic weapon handling skills`
`05:26.56`	`ChrisInSydney`	`We had a few security companies held over for cash and guns here a few years ago`
`05:26.59`	`*** join/#asterisk dgeary2 (~david@120.21.26.46)`
`05:27.20`	`Maliuta`	`slav3_kitten: 18months after deploying them they did a review ... something like 8 cases of guys shooting their butts while putting on their belts`
`05:27.44`	`Maliuta`	`ChrisInSydney: you get that hanging around the cross though :)`
`05:28.00`	`ChrisInSydney`	`you also get king hit and killed too`
`05:28.17`	`Maliuta`	`or run over by a 16y/o while you're on the sidewalk`
`05:28.46`	`Maliuta`	`ChrisInSydney: you're more of an Oxford St guy??`
`05:29.00`	`ChrisInSydney`	`Maliuta: When I go out, I normally head Newtown way as the Cross is F*&^ed and Darlo is full of pissheads`
`05:29.06`	`Maliuta`	`it's not really CampInSydney is it? ;)`
`05:29.11`	`ChrisInSydney`	`ahh`
`05:29.52`	`Maliuta`	`ChrisInSydney: I can hook you up with one of my mates from politics who moved down there recently ... I'm sure he'd like the "company" ;P`
`05:29.54`	`ChrisInSydney`	`Three things not to say when in a bar on Oxford Street`
`05:30.09`	`Maliuta`	`"Nice shoes" ;)`
`05:30.14`	`ChrisInSydney`	`1: Say "Can I bum a fag"`
`05:30.22`	`Maliuta`	`LOL`
`05:30.22`	`ChrisInSydney`	`2: Say "I'm Positive"`
`05:30.39`	`ChrisInSydney`	`3: Say "Could you please push in my stool"`
`05:30.53`	`Maliuta`	`ROFPML`
`05:31.20`	`ChrisInSydney`	`and finally, if you drop your wallet on oxford Street, don't bend down, just keep kicking it all the way to George Street`
`05:32.56`	`*** join/#asterisk Neptu (~Neptu@static-213.88.195.204.addr.tdcsong.se)`
`05:33.15`	`slav3_kitten`	`lotta gay people i take it ChrisInSydney ?`
`05:33.17`	`ChrisInSydney`	`Maliuta: If you are trying to hit on me, thats cool, I'm flattered, but I'm straight. Anyway, if I wasn't, my hair dresser says he's first !`
`05:35.06`	`ChrisInSydney`	`slav3_kitten: Just between Darlinghurst and Paddington, commonly referred to as Paddyhurst.`
`05:35.29`	`slav3_kitten`	`nods`
`05:35.31`	`ChrisInSydney`	`Its a bit like Castro Street in San Francisco I guess`
`05:36.05`	`slav3_kitten`	`except less people with guns`
`05:36.21`	`ChrisInSydney`	`not really`
`05:36.40`	`ChrisInSydney`	`there was a few drive bys last year`
`05:36.44`	`slav3_kitten`	`thought they took all your guns away`
`05:36.53`	`ChrisInSydney`	`Only from the good people`
`05:37.30`	`ChrisInSydney`	`We haven't got a 2nd here. Actually, technically we dont have the equivilent of the first !`
`05:38.11`	`slav3_kitten`	`you did start out as a prison colony though`
`05:38.25`	`ChrisInSydney`	`Yup, prisoners and drunken Irish`
`05:38.52`	`ChrisInSydney`	`Fook This, they said`
`05:39.17`	`slav3_kitten`	`well the brits prolly figured since everything over there can kill you that you'd all just pass on`
`05:39.42`	`slav3_kitten`	`it's staggering the number of things that can kill you in that country`
`05:39.48`	`ChrisInSydney`	`yup, but the Abo's have been here for 35000 years`
`05:40.14`	`ChrisInSydney`	`only two spiders, but a shit load of snakes`
`05:40.39`	`slav3_kitten`	`35,000 years of pure badasses. natural selection at it's finest`
`05:40.58`	`ChrisInSydney`	`Then we have the "Red Centre", thousands of kms of F'All`
`05:42.35`	`slav3_kitten`	`this has not convinced me that you're not all off your rocker you know`
`05:43.20`	`ChrisInSydney`	`I certainly wasn't trying to convince anyone of that, whay would I want to mislead you`
`05:43.22`	`ChrisInSydney`	`?`
`05:43.45`	`slav3_kitten`	`lol`
`05:44.00`	`slav3_kitten`	`i almost went to midfur one year but couldn't get the capital together`
`05:45.42`	`ChrisInSydney`	`may have found where this transfer thingy is set up`
`05:46.04`	`slav3_kitten`	`yay`
`05:50.40`	`ChrisInSydney`	`Not quite`
`05:51.52`	`ChrisInSydney`	`ast_verb(3, "************** HERE!!!!!!!!!!! ******************.\n");`
`05:52.08`	`ChrisInSydney`	`using a few of these to work out whats going on`
`05:52.28`	`ChrisInSydney`	`'cause I dont know how to use debug`
`05:54.52`	`ChrisInSydney`	`it must be in features`
`05:56.58`	`slav3_kitten`	`debug is`
`05:57.08`	`slav3_kitten`	`sip set debug on iirc at the console`
`05:57.21`	`ChrisInSydney`	`talking c debugs`
`05:57.40`	`slav3_kitten`	`oh`
`05:57.48`	`slav3_kitten`	`i'm a moron`
`05:57.50`	`slav3_kitten`	`an it's late`
`05:57.59`	`ChrisInSydney`	`have another smoke`
`05:58.00`	`slav3_kitten`	`mostly the moron bit`
`05:58.17`	`ChrisInSydney`	`your choice what you want to light up`
`06:01.47`	`ChrisInSydney`	`could be a 5 line fix :-)`
`06:04.03`	`ChrisInSydney`	`its a big file 8.8K lines of code :-/`
`06:04.16`	`ChrisInSydney`	`features.c in ~/main`
`06:05.01`	`ChrisInSydney`	`typo. Recompile`
`06:08.15`	`ChrisInSydney`	`FUCKYEAHHHHH!!!!!!!! it works`
`06:08.28`	`ChrisInSydney`	`XFER_CUSTOM_CONTEXT`
`06:08.51`	`ChrisInSydney`	`Set that and features.c picks it up as the preferred context`
`06:09.08`	`ChrisInSydney`	`p3nguin: Got a fix :-)`
`06:14.12`	`ChrisInSydney`	`<PROTECTED>`
`06:14.13`	`ChrisInSydney`	`<PROTECTED>`
`06:14.13`	`ChrisInSydney`	`<PROTECTED>`
`06:14.13`	`ChrisInSydney`	`<PROTECTED>`
`06:14.28`	`ChrisInSydney`	`forgot`
`06:14.51`	`ChrisInSydney`	`<PROTECTED>`
`06:14.51`	`ChrisInSydney`	`<PROTECTED>`
`06:14.54`	`ChrisInSydney`	`that was forst`
`06:14.57`	`ChrisInSydney`	`first`
`06:17.17`	`ChrisInSydney`	`Anyone want the fix ??`
`06:18.01`	`ChrisInSydney`	`How to set the context for doing a blind transfer using the Dial(SIP/xxx,,t) option`
`06:20.59`	`ChrisInSydney`	`Maliuta: America sleeps while our euro cousins are just waking up to a Teasmade`
`06:21.14`	`ChrisInSydney`	`just you and me and maybe a stra Kiwi ;-)`
`06:22.30`	`ChrisInSydney`	`s/just you and me and maybe a stra Kiwi ;-)/just you and me and maybe a stray Kiwi ;-)`
`06:23.57`	`Maliuta`	`I'm not 100% compus mentus right now ... and I'm distracted by the cricket :)`
`06:27.47`	`ChrisInSydney`	`how are we going ??`
`06:28.44`	`ChrisInSydney`	`BTW, there is a TRANSFER_CONTEXT channel var that already exists. Just found it in the code in ~/main/features.c`
`06:28.56`	`ChrisInSydney`	`what a bloody waste of time :-/`
`06:29.00`	`ChrisInSydney`	`not completely`
`06:29.37`	`ChrisInSydney`	`I can now split my contexts for blind / attended / park ext. I cant see why I would need such a feature, but I now have it`
`06:29.40`	`ChrisInSydney`	`time to go home`
`06:29.45`	`ChrisInSydney`	`5:30pm here`
`06:30.04`	`ChrisInSydney`	`Type later, when I get home`
`06:30.52`	`ChrisInSydney`	`c yaz`
`06:35.35`	`*** join/#asterisk elico (~Thunderbi@bzq-109-64-221-76.red.bezeqint.net)`
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`07:19.01`	`ChrisInSydney`	`back`
`07:19.14`	`v0lZy`	`hi ChrisInSydney`
`07:19.17`	`ChrisInSydney`	`hey`
`07:19.22`	`v0lZy`	`been a while, whats cooking?`
`07:19.38`	`ChrisInSydney`	`pizza hopefully`
`07:19.40`	`ChrisInSydney`	`;-)`
`07:20.01`	`ChrisInSydney`	`just got home from the office, so I have to do a quick shop`
`07:20.20`	`ChrisInSydney`	`get some stuff for the pizza bases`
`07:20.34`	`ChrisInSydney`	`The young guy is 3 now !`
`07:21.01`	`ChrisInSydney`	`so most of my evenings are family stuff`
`07:21.20`	`ChrisInSydney`	`yourself?? whats new ??`
`07:22.37`	`v0lZy`	`here, well... working half time now`
`07:22.48`	`v0lZy`	`not my choice so sucks.`
`07:23.08`	`ChrisInSydney`	`bugger :-/`
`07:25.30`	`v0lZy`	`it stinks cause its a personal matter really.`
`07:26.09`	`v0lZy`	`owner is the defacto boss, but his partner whos a step lower than the owner's a backstabber.`
`07:27.58`	`v0lZy`	`I went to Japan in April and this other guy started nitpicking. Looking for a reason. Since he couldnt find one, he made some shit up and I told him to go bugger off.`
`07:28.28`	`v0lZy`	`i come back from vacation and they want me to handover IT to an external company, citing that inhouse is too expensive`
`07:28.43`	`v0lZy`	`turns out the external IT wants twice my pay check to do my stuff.`
`07:28.58`	`ChrisInSydney`	`pricks`
`07:29.11`	`v0lZy`	`Obviously, they cant terminate on technical redundancy grounds`
`07:29.42`	`v0lZy`	`so the big men himself acts on behalf of this second in command prick.`
`07:30.08`	`ChrisInSydney`	`we have worker protection laws. They cant always stop things, but they can be a pain in the arse, exspecally for larger companies`
`07:30.21`	`v0lZy`	`We have that too, on paper anyway.`
`07:30.29`	`ChrisInSydney`	`are they getting a kick back ?`
`07:31.01`	`v0lZy`	`who?`
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`07:31.07`	`v0lZy`	`The owner and the guy?`
`07:31.18`	`ChrisInSydney`	`the bosses ? are they owners, or just managers ?`
`07:31.24`	`v0lZy`	`owner = boss, the other guy's the acting boss.`
`07:31.54`	`v0lZy`	`basically the owner is the boss and signs everthing, but thats about it. everything else is done by this other prick... which mostly comes down to yelling and not doing anything profitable.`
`07:32.13`	`v0lZy`	`Its not a merit based employment system really.`
`07:32.32`	`ChrisInSydney`	`never is`
`07:32.48`	`v0lZy`	`anyway, this middle guy was close to my dad.`
`07:33.20`	`deo_`	`hi all. .may I interrupt... just want to ask how to check dahdi channel on asterisk ???`
`07:33.34`	`ChrisInSydney`	`thats right. I remember all that crap :-/`
`07:33.39`	`v0lZy`	`my dad passed away in 2009. my parents were divorced early on, so he had this other woman... not married though`
`07:33.55`	`v0lZy`	`Anyway, this guy was banging her behind his back apparently.`
`07:34.21`	`ChrisInSydney`	`Jerry!!!! Jerry!!!!! Jerry!!!!!`
`07:34.25`	`v0lZy`	`So come all the inheritance stuff and the inheritance court thing, obviously he's siding with the woman.`
`07:34.34`	`v0lZy`	`Yeah, Springer stuff.`
`07:34.35`	`ChrisInSydney`	`was referring to a Jerry Springer Show`
`07:34.52`	`v0lZy`	`I know :D`
`07:35.27`	`v0lZy`	`Anyway... we're still not done with the whole thing.. i got a new contract here on the table, not signing it,`
`07:35.32`	`ChrisInSydney`	`anyways, dahdi is dahdi show .... same as sip show ...`
`07:36.08`	`v0lZy`	`I mean i dont mind my working hours being half of what they were.. but my pay check should be there abouts too.`
`07:36.45`	`ChrisInSydney`	`so long as they pay you the same money, you're sweet`
`07:37.06`	`ChrisInSydney`	`Otherwise, start looking and see of you can get some moonlight work`
`07:39.20`	`v0lZy`	`i dont feel like being in this company anymore anyway.`
`07:39.25`	`v0lZy`	`jobs are hard to find here though.`
`07:40.14`	`deo_`	`hi all.. ive checked one of my extension dial plan,, theres part which says Dial Zap/6 > is this extension directly connected to my TDM Card???`
`07:40.57`	`deo_`	`i cant find the cable though because its on the rooftop lol`
`07:41.23`	`v0lZy`	`Thing's are getting bad here. No real jobs anymore... everything's either sales or marketing and no fixed pay on that.`
`07:42.06`	`v0lZy`	`Basically it means there's no more market and they're calling in the hords to try and sell the crap they import.`
`07:42.32`	`kaldemar`	`deo_: that depends on the dahdi configuration (/etc/dahdi/system.conf) and the card itself.`
`07:42.46`	`deo_`	`hmmmn i wann check it..`
`07:42.55`	`deo_`	`btw, what is mean by this > DAHDI/4-1`
`07:43.03`	`deo_`	`4 > the card???`
`07:43.16`	`deo_`	`i find that on the call logs`
`07:43.17`	`kaldemar`	`deo_: no. 4 is a channel.`
`07:43.25`	`kaldemar`	`deo_: what are you trying to do?`
`07:43.27`	`deo_`	`and what is 1?`
`07:43.41`	`deo_`	`im gonna check one of the extension...`
`07:43.53`	`deo_`	`i found in the dialplan > Dial Zap/6`
`07:44.00`	`kaldemar`	`it's an identifier`
`07:44.36`	`deo_`	`identifier for what?`
`07:47.45`	`kaldemar`	`deo_: a call on the channel.`
`07:49.29`	`kaldemar`	`what do you mean by "check"?`
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`08:16.57`	`deo_`	`hi kaldemar sorry for the late reply...`
`08:17.14`	`deo_`	`i troubleshoot the problem`
`08:17.37`	`deo_`	`i reloaded the machine :D`
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`08:26.31`	`loggiew`	`I keep getting the error wss and ws are not a valid transport when reloading sip.conf, any ideas on whats up? Trying to use webrtc`
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`08:28.04`	`kaldemar`	`loggiew: are you using version 11.0.0?`
`08:28.59`	`loggiew`	`ah ha im a dumbass apparently not. sorry for dumb question`
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`09:21.09`	`loggiew`	`for the life of me I cannot seem to figure out why asterisk 11 refuses to configure srtp support before I run make. Always complains that it cant create a shared object.`
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`09:25.15`	`kaldemar`	`loggiew: have you installed the development package for libsrtp and re-run the configure script?`
`09:26.49`	`loggiew`	`i downloaded srtp source and installed then went to asterisk for recompile and it perpetually says it cannot be linked as a shared object`
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`09:28.44`	`kaldemar`	`loggiew: did you run the configure script?`
`09:28.55`	`loggiew`	`yep`
`09:29.08`	`kaldemar`	`using --with-srtp with a correct path?`
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`09:30.23`	`loggiew`	`i actually have a question about that, path to what exactly? Ive seen a number of pages referencing using /usr but Im not certain which files it is looking for in /usr so I can make sure thats the correct path`
`09:32.30`	`kaldemar`	`headers`
`09:32.51`	`loggiew`	`checking srtp/srtp.h usability... yes`
`09:32.51`	`loggiew`	`checking srtp/srtp.h presence... yes`
`09:32.51`	`loggiew`	`checking for srtp/srtp.h... yes`
`09:32.54`	`x1user`	`I am not seeing sip in the CLI in asterisk 11 ?`
`09:32.58`	`loggiew`	`sorry, hope a short paste was ok`
`09:33.25`	`loggiew`	`it appears to find everything and then says it cant link it`
`09:33.29`	`loggiew`	`sigh`
`09:33.42`	`kaldemar`	`x1user: then you don't have chan_sip.so loaded.`
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`09:36.50`	`x1user`	`thanks kaldemar`
`09:40.38`	`loggiew`	`i may need to bang my head against the wall more tomorrow instead`
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`09:50.48`	`loggiew`	`interesting`
`09:50.54`	`loggiew`	`the way configure tests that`
`09:51.18`	`loggiew`	`is compiling a short .c which includes the header and runs srtp_init()`
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`09:51.40`	`loggiew`	`when performing the same action manually, I get an undefined reference to srtp_init()`
`09:51.55`	`loggiew`	`to get it to compile without error i have to use -lsrtp`
`09:52.04`	`loggiew`	`which it appears the script is doing`
`09:52.37`	`loggiew`	`but otherwise why is it returning failure on compiling that .c`
`09:53.23`	`loggiew`	`i dunno, still reading`
`09:56.13`	`BorjaGVO`	`Hi everyone, a question here...Does anyone know how can I establish a minimum time for a caller to stay in a queue? I mean, I want that at least, the caller stays holding on in the queue for 20 seconds. The way I got around it is playing MoH for 20 seconds before putting the caller into the queue...that way I know for sure that he/she will wait at least for 20 seconds before entering the queue. However, this is fine when "max wai`
`09:57.36`	`BorjaGVO`	`....but if "max wait time" has to be around 30 seconds, if I do the "trick" of playin MoH before entering the queue there will be very little time for picking up the phone...how can this be done so that queue members phones ring?`
`10:08.53`	`*** join/#asterisk brian98 (~brian98@79.97.2.26)`
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`10:20.06`	`loggiew`	`no matter what it appears compiling srtp doesn't appear to properly be passing the -fPIC parameter`
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`10:33.14`	`bombev`	`Hi guys, I have problem with my skype asterisk calls.`
`10:33.35`	`bombev`	`the inbound skype call goes to the wrong route....`
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`10:35.48`	`ruied`	`Hi, have anyone used a USB ISDN adapter, like Gerdes Primux USB ? I would like to know if it works ok. I'm thinking of make some tests for a small pbx like: Raspberry_Board+USB-ISDN with asterisk.`
`10:38.31`	`ruied`	`this could be a good and cheap way for small business with less than 6 concurrent calls...`
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`10:50.31`	`ruied`	`is the skype for asterisk free or is it payed?`
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`10:53.20`	`Chainsaw`	`ruied: "PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013."`
`10:57.42`	`bombev`	`here is my asterisk log http://pastebin.ca/2250922`
`10:59.07`	`Chainsaw`	`bombev: It can't reach any of the 6 3-digit SIP extensions you have set up.`
`10:59.20`	`Chainsaw`	`bombev: So it reports back to Skype saying "busy" and terminates the call.`
`11:02.52`	`bombev`	`ChanServ the strange thing here is`
`11:03.07`	`bombev`	`I have setup the skype call to go to ring group 6007`
`11:03.32`	`bombev`	`but in the asterisk log it is shown it goes to 6002 ring group`
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`11:05.21`	`ruied`	`Chainsaw, I did know that, I was in the hope that they have changed... :(`
`11:05.49`	`Chainsaw`	`ruied: Do you need to queue reload?`
`11:06.00`	`weinerk`	`Hi. Please help - if I make an outbound Dial from AGI - how do I know which side did a hangup?`
`11:06.03`	`Chainsaw`	`ruied: Are there other aspects of Asterisk that you have reconfigured on disk but not reloaded?`
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`11:22.46`	`fredericve`	`Hi, is there any function to escape special characters in the configuration? e.g. the colon (:) needs to be escaped when you use it in the IF function.`
`11:30.30`	`bombev`	`Chainsaw any idea?`
`11:30.46`	`Chainsaw`	`bombev: With what you've given me, this is all I can say.`
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`11:37.45`	`Greenlight`	`Howdy folks. I'm having an odd issue with Asterisk 10.6.1. It's seg faulted twice in the mornings over the past few days. Looking at the logs I see a number of these "[2012-11-12 09:43:04] WARNING[19506] pbx.c: Failed to create new channel thread" before it finally dies. Any ideas?`
`11:44.12`	`x1user`	`I have strange problem, i got iptables service stopped and asterisk 11 max core debug and verbose, but i cant see attempt for registering sip accounts?`
`11:45.26`	`Greenlight`	`Try "sip set debug on"`
`11:45.56`	`Greenlight`	`And see if you see anything that way. IS it a public IP?`
`11:55.12`	`Greenlight`	`Anyone any ideas what might cause "WARNING[19506] pbx.c: Failed to create new channel thread" ?`
`12:00.09`	`kaldemar`	`x1user: iptables is not a service. what did you really do to it?`
`12:00.55`	`x1user`	`service iptables stop, i am sniffing now my network for the sip packets, i got sip response 401 which is unathorized =/`
`12:01.01`	`Greenlight`	`I'm guessing he did "service iptables stop". It normally operates as a pseudo service and the firewall can be stopped like that`
`12:02.59`	`*** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone)`
`12:03.49`	`*** join/#asterisk surferboy (~surferboy@41.177.76.231)`
`12:03.57`	`surferboy`	`total asterisk newbie here`
`12:04.06`	`surferboy`	`can someone assist with an issue I have?`
`12:04.32`	`surferboy`	`how to you redirect incoming calls if they are unanswered by the sip phone?`
`12:05.40`	`kaldemar`	`x1user: use asterisk to see what goes on, "sip set debug on". 401 is normal behavior, telling the other end to authenticate with a new message.`
`12:06.26`	`kaldemar`	`surferboy: if you want to do it in asterisk, set a timeout in app Dial and do what you want in the following priority.`
`12:07.58`	`surferboy`	`kaldemar, k can you help me with that?`
`12:08.11`	`surferboy`	`there is an elasix setup if that helps`
`12:09.01`	`kaldemar`	`that certainly does not help.`
`12:09.19`	`surferboy`	`lol`
`12:09.32`	`surferboy`	`so can you help me with what you said?`
`12:09.42`	`surferboy`	`how do I set a timeout in app Dial?`
`12:09.50`	`kaldemar`	`probably just makes it harder. if you're using a GUI to configure asterisk, you should ask help in the appropriate place, which in this case is #elastix.`
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`12:10.08`	`x1user`	`omg it was wrong ip :D`
`12:10.49`	`surferboy`	`I'm not using a GUI to configure anything`
`12:11.01`	`surferboy`	`kaldemar, which config file do I need to have a look at`
`12:11.11`	`*** part/#asterisk tyrion (~tyrion@skynet.skynet.ie)`
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`12:16.23`	`fredericve`	`surferboy: /etc/asterisk/extensions.conf`
`12:16.41`	`kaldemar`	`surferboy: extensions.conf or some other file that is included from it with "#include ..."`
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`12:21.13`	`Greenlight`	`Are there any know issues with 10.6.1 such as file handle leaks or something?`
`12:22.17`	`ChrisInSydney`	`hi all, has anyone hacked asterisk to make it work with SIP NOTIFY - Record On/Off messages from Snom handsets ?? Anyone know of any patches ??`
`12:23.42`	`ChrisInSydney`	`ahh. looks like something might be happening`
`12:23.58`	`ChrisInSydney`	`[Nov 12 23:22:58] WARNING[7701]: chan_sip.c:18988 handle_request_info: Recording requested, but no One Touch Monitor registered. (See features.conf)`
`12:25.49`	`x1user`	`http://pastebin.com/24gJHutB Why I still cant register my sip phone ?`
`12:27.49`	`coreyf1513`	`Greenlight: I suggest update to 10.10.0 see if the issue still exists, that would contain bug fixes compared to 10.6.1`
`12:29.32`	`Greenlight`	`I couldn't see anything that jumped out when looking through changelog, and we had issues using 10.8 with the AMI interface, so I was reluctant to upgrade. Is there a specific bug fix your thinking of?`
`12:31.31`	`Greenlight`	`From googling people seem to relate the message to a memory issue`
`12:31.47`	`kaldemar`	`x1user: "403 Forbidden (Bad auth)"`
`12:32.02`	`x1user`	`password and username are ok`
`12:32.11`	`x1user`	`but pass is plain text`
`12:32.14`	`*** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain)`
`12:32.16`	`coreyf1513`	`Greenlight: sorry nothing specific, just a generic suggestion`
`12:32.25`	`kaldemar`	`x1user: doesn't look like they are ok.`
`12:33.13`	`Greenlight`	`coreyf1513: No probs, thanks, I'll see if we can get upgraded`
`12:33.14`	`kaldemar`	`x1user: there's also "No matching peer for '666' from '192.168.9.194:5060'" in your pastebin. that's something you need to fix.`
`12:33.16`	`*** join/#asterisk jacekowski (jacekowski@jacekowski.org)`
`12:34.10`	`x1user`	`http://pastebin.com/x8bhX8Vs`
`12:34.37`	`Greenlight`	`x1user: Did you reload after editing sip.conf ?`
`12:34.51`	`kaldemar`	`x1user: nothing matches "666", which your zoiper is sending.`
`12:34.56`	`Greenlight`	`Do a "core reload" to be sure..`
`12:35.16`	`kaldemar`	`"sip reload" is enough when modifying sip.conf.`
`12:35.35`	`x1user`	`yes i did it already`
`12:36.30`	`Greenlight`	`How is the peer defined in sip.conf ?`
`12:37.06`	`x1user`	`http://pastebin.com/eHV1RkPz`
`12:38.17`	`Greenlight`	`I cant see a "666" peer`
`12:38.45`	`Greenlight`	`Your zoiper is trying to register as "666" but it looks like it should be "zoiper1"`
`12:40.09`	`x1user`	`http://pastebin.com/p7WXYDyg take a look at this`
`12:40.22`	`x1user`	`this 666 is from other voip phone ie tried`
`12:41.38`	`kaldemar`	`[zoipper] vs zoipper1`
`12:41.48`	`kaldemar`	`still no match.`
`12:43.11`	`x1user`	`omg`
`12:43.45`	`x1user`	`the truth is always obvious`
`12:44.29`	`x1user`	`thanks anyway`
`12:55.33`	`*** join/#asterisk chris_n (~Chris@184.7.21.42)`
`12:58.17`	`ChrisInSydney`	`Looks like the Snom Record button does work out of the box :-)`
`12:58.47`	`Greenlight`	`Don't you need to configure it via features.conf`
`13:00.57`	`ChrisInSydney`	`yep and put a w in the Dial / Queue() app`
`13:01.00`	`ketas`	`i wonder why my previous "what are my other sessions" question confused everybody`
`13:01.22`	`ketas`	`(talking about sip client)`
`13:01.53`	`ChrisInSydney`	`now to work out how to set the file name and recording file location`
`13:02.06`	`ketas`	`strange x problems, if i get it up i'll research it by myself... maybe google understands me better`
`13:14.21`	`Greenlight`	`I'm wondering if my problems with Asterisk crashing are related to the kernel doing too much cacheing of disk stuff, and maybe even forcing Asterisk to use swap at its expense. Anyone had any issues like this? The kernel seems to chew up lots of memory for cache when I'm moving around a boat load of call recordings.`
`13:18.36`	`*** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain)`
`13:21.19`	`p3nguin`	`chrisinsydney: Problem solved.`
`13:21.44`	`p3nguin`	`chrisinsydney: That channel variable made it work a WHOLE LOT better.`
`13:21.51`	`ChrisInSydney`	`:-)`
`13:21.56`	`ChrisInSydney`	`found it in the code`
`13:22.09`	`p3nguin`	`I'm glad you did that for me while I slept.`
`13:22.16`	`ChrisInSydney`	`Googled it and found a few mentions`
`13:22.56`	`ChrisInSydney`	`I've also setup BLIND_TRANSFER_CONETXT and ATTENDED_TRANSFER_CONTEXT`
`13:23.01`	`*** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com)`
`13:23.18`	`ChrisInSydney`	`useful to force attended transfers for transfers to external numbers`
`13:23.26`	`ChrisInSydney`	`Thats working`
`13:23.58`	`p3nguin`	`TRANSFER_CONTEXT covers both, and those two allow you to set each individually?`
`13:24.07`	`ChrisInSydney`	`Yup`
`13:24.16`	`ChrisInSydney`	`but you need a patch for them to work`
`13:24.50`	`ChrisInSydney`	`const char *custom_dial_context = NULL;`
`13:25.06`	`ChrisInSydney`	`ast_channel_lock(chan);`
`13:25.07`	`ChrisInSydney`	`custom_dial_context = pbx_builtin_getvar_helper(chan, "BLIND_TRANSFER_CONTEXT");`
`13:25.07`	`ChrisInSydney`	`transferer_real_context = !ast_strlen_zero(custom_dial_context) ? ast_strdupa(custom_dial_context) : transferer_real_context;`
`13:25.07`	`ChrisInSydney`	`ast_channel_unlock(chan);`
`13:25.07`	`ChrisInSydney`	`ast_verb(3, "Context to transfer from is %s.\n", transferer_real_context);`
`13:25.25`	`*** join/#asterisk wonderworld (~w@dsdf-4db530da.pool.mediaWays.net)`
`13:25.46`	`ChrisInSydney`	`in ~main/features.c`
`13:26.48`	`ChrisInSydney`	`In fact thats pretty much the code I am using to create other chanel var hacks`
`13:27.12`	`*** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com)`
`13:27.34`	`coreyf1513`	`Greenlight: kernel cache using nearly all 'available' memory is normal. http://www.linuxatemyram.com/`
`13:27.50`	`*** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net)`
`13:28.18`	`ChrisInSydney`	`my challenge now it so get a call to a mobile from a queue, where the mobile rejects the incomming call, for the mobile (cell phone) agent to pause until the next call comes in or for a predefined time`
`13:29.05`	`p3nguin`	`Oh. You added those other variables... I thought you were saying they were built in already.`
`13:29.17`	`ChrisInSydney`	`nope`
`13:29.50`	`p3nguin`	`That's okay, though. Setting the transfer context takes care of the issue I encountered.`
`13:30.08`	`ChrisInSydney`	`had to add. Infact i added before I found the ${TRANSFER_CONTEXT} thingy`
`13:30.46`	`ChrisInSydney`	`When I get these all working, I'll post them up somewhere as .patches`
`13:31.19`	`ChrisInSydney`	`stick a post in the Ast support forums`
`13:31.34`	`*** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri)`
`13:31.38`	`ChrisInSydney`	`until then, back to queues`
`13:32.30`	`ChrisInSydney`	`I would love to be able to set the time out and retry on an agent by agent basis`
`13:33.09`	`ChrisInSydney`	`Not too sure whether to cobble a dial plan macro thingy or hack some more C`
`13:33.16`	`[TK]D-Fender`	`chris_n, Then set a timeout mased on your max and cause an inner dial to limit it internally`
`13:33.27`	`[TK]D-Fender`	`ChrisInSydney, ^`
`13:33.45`	`*** join/#asterisk elico (~Thunderbi@bzq-109-67-228-219.red.bezeqint.net)`
`13:34.18`	`ChrisInSydney`	`[TK]D-Fender: Thats what I am trying`
`13:34.34`	`*** join/#asterisk mathi (Matthew@ip-62-235-244-174.dsl.scarlet.be)`
`13:34.52`	`ChrisInSydney`	`but you've made me rethink`
`13:34.53`	`ChrisInSydney`	`:)`
`13:34.55`	`ChrisInSydney`	`cheers`
`13:35.01`	`ChrisInSydney`	`more coffee`
`13:35.07`	`ChrisInSydney`	`12:35am here`
`13:35.08`	`*** join/#asterisk BorjaGVO (d51beb92@gateway/web/freenode/ip.213.27.235.146)`
`13:36.02`	`p3nguin`	`I'm 17 hours behind you.`
`13:37.43`	`p3nguin`	`And at 0.0006944444 RPM, I'm never going to catch up!`
`13:38.04`	`ChrisInSydney`	`p3nguin :)`
`13:38.38`	`BorjaGVO`	`Hi! Anyone can help? I would like to set two different periodic-announcements in queues.conf so that one says "please hold the line, all our agents are busy" and the other one, after 20 secs for example, says something like "sorry, our agents are still busy, please hold on"..for example. The thing is that it seems not to be possible..is there any way of doing this?`
`13:38.47`	`ChrisInSydney`	`Got the Mondy night lotto results, I can send them to you and we can split it down the middle ;-)`
`13:38.55`	`p3nguin`	`Sweet!`
`13:39.34`	`*** join/#asterisk wonderworld (~w@dsdf-4db530da.pool.mediaWays.net)`
`13:39.52`	`ChrisInSydney`	`BorjaGVO: Play annoucement 1 before you enter the queue`
`13:41.55`	`*** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net)`
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`13:44.00`	`[TK]D-Fender`	`BorjaGVO, There is no queue option for this`
`13:45.17`	`p3nguin`	`If you only need to announce it that one time, doing so immediately before starting the queue should solve it.`
`13:45.39`	`ChrisInSydney`	`or hack a custom music on hold class`
`13:46.07`	`ChrisInSydney`	`music, announce 1, more music, announce 2, etc`
`13:46.11`	`ChrisInSydney`	`crude`
`13:46.42`	`p3nguin`	`Playback(agents-busy); Queue(main)`
`13:46.45`	`p3nguin`	`That's what I do.`
`13:47.10`	`p3nguin`	`Then the announce setting plays the still-busy file.`
`13:47.48`	`[TK]D-Fender`	`Yup, hacking music is probably the closest you'll get.Only downside is that agents might answer during the message and the cut-off would sound awkward`
`13:48.46`	`ChrisInSydney`	`Isnt there a pause when an agent picks up and something to force a ring tone ??`
`13:49.04`	`p3nguin`	`You could also have multiple queues, where the first queue announces once and runs for only 20s. Then the next queue can play the other announcement.`
`13:49.37`	`p3nguin`	`I don't like that idea, but it would achieve the same results.`
`13:49.59`	`ChrisInSydney`	`I'm looking at the code for app_queue.c, you could hack an alternating announcement if you were keen`
`13:50.16`	`p3nguin`	`Sounds like a lot of work.`
`13:50.27`	`p3nguin`	`I'd use what options are already available.`
`13:51.02`	`*** join/#asterisk wonderworld (~w@dsdf-4db530da.pool.mediaWays.net)`
`13:51.59`	`ChrisInSydney`	`If you had 0.5 a clue, you could probably fix up something in a couple of hours.`
`13:52.05`	`*** join/#asterisk bchia (~Adium@nat/digium/x-tucdbwcauujlspnb)`
`13:53.06`	`p3nguin`	`Lovely. Feels like -4C today.`
`13:53.26`	`jaytee`	`do Digium T1 cards support fractional T1?`
`13:55.06`	`ChrisInSydney`	`got back from Darwin last week. was 34C most of the time and close to 80%-90% humidity. Just before the wet`
`13:55.36`	`p3nguin`	`How was the dew point?`
`13:55.45`	`ChrisInSydney`	`jaytee: Just enable the channels that are active. At least thats what I do on the E1 cards`
`13:56.04`	`ChrisInSydney`	`beer point`
`13:56.49`	`ChrisInSydney`	`we got one supercell. dropped right down to 22 for around an hour or so`
`13:56.52`	`*** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4)`
`13:57.19`	`ChrisInSydney`	`storm, lightning, sideways rain`
`13:57.28`	`ChrisInSydney`	`then hot and muggy`
`13:57.36`	`ChrisInSydney`	`posted a few photos on +`
`13:59.47`	`*** join/#asterisk widler (47c8d864@gateway/web/freenode/ip.71.200.216.100)`
`14:00.29`	`*** join/#asterisk serafie (~erin@nat/digium/x-yeyygrsnpfmcamfa)`
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`14:07.13`	`*** mode/#asterisk [+o pabelanger] by ChanServ`
`14:08.16`	`BorjaGVO`	`ChrisInSydney: the thing is that this first announcement has to be played after 20 seconds once the caller is in the queue`
`14:09.00`	`BorjaGVO`	`[TK]D-Fender: and do you think of any work around for playing one announcement first (after 20 secs once the caller got into queue) and the other later on?...`
`14:09.33`	`[TK]D-Fender`	`BorjaGVO, Considering you're using FreePBX ... NO`
`14:10.07`	`BorjaGVO`	`well...I can modify dialplan manually...`
`14:10.49`	`BorjaGVO`	`but I don't imagine how to do it since once you've got into queue you just can use queues.conf options`
`14:11.15`	`[TK]D-Fender`	`BorjaGVO, If you go code your own queue and dialplan before entry you could launch a process that on a timed interval would originate a local channel to chanspy+whisper to the original channel (which you'd have to disable on agent answer), but might work.`
`14:11.26`	`[TK]D-Fender`	`BorjaGVO, MoH is the best answer for this though`
`14:11.45`	`Greenlight`	`I'm noticing massive load average spikes on my Asterisk box. Goes from a steady 0.5-1.0 way up to like 40. The asterisk process remains at a steady 150%. I've 12 cores so this is nothing. I've not got any disk IO issues.`
`14:12.46`	`ChrisInSydney`	`[TK]D-Fender: BorjaGVO, MoH is the best answer for this though (+1)`
`14:13.20`	`BorjaGVO`	`The thing is that music on hold wouldn't let the agents to pick up the phone...that is what I was doing until now...but is useless...`
`14:13.35`	`ChrisInSydney`	`'splain`
`14:14.44`	`ChrisInSydney`	`you set your moh class as sequestial, set the first file to be 20 seconds long, file 2 = announcement 1, file 3.... keeps playing and use your periodic announcements from the queue`
`14:14.51`	`ChrisInSydney`	`unless you want to alternate`
`14:15.02`	`[TK]D-Fender`	`BorjaGVO, that does not make sense. You hear MoH while waiting for the agent ALREADY. How does it STOP agents from answering?`
`14:15.02`	`BorjaGVO`	`[TK]D-Fender, coding my own queue? How could I launch a process on a timed interval?`
`14:15.15`	`*** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net)`
`14:15.17`	`ChrisInSydney`	`the moh class has nothing to do with agents picking up the call`
`14:15.55`	`ChrisInSydney`	`BorjaGVO: are you talking hacking up app_queue.c ? or just coding a dialplan thingy`
`14:16.24`	`BorjaGVO`	`If I enter the queue...music on hold is set..of course...but if I play music on hold for 20 seconds before entering the queue no agent would be able to pick up the phone`
`14:16.32`	`BorjaGVO`	`since the call didn't get into the queue yet`
`14:17.00`	`ChrisInSydney`	`you set your moh class as sequestial, set the first file to be 20 seconds long, file 2 = announcement 1, file 3.... keeps playing and use your periodic announcements from the queue`
`14:17.18`	`ChrisInSydney`	`s/sequestial/sequential`
`14:17.34`	`[TK]D-Fender`	`BorjaGVO, not BEFORE entering the queue.... CHANGE THE MUSIC TO A FIXED RECORDING WITH BOTH MESSAGES PRE-INTEGRATED`
`14:17.49`	`BorjaGVO`	`ChrisInSydney: I'm asking [TK]D-Fender what he was thinking of...I thought same as you..`
`14:18.12`	`BorjaGVO`	`[TK]D-Fender: ok, i got you`
`14:18.20`	`BorjaGVO`	`that's a good idea...`
`14:18.22`	`ChrisInSydney`	`all good`
`14:18.40`	`[TK]D-Fender`	`ChrisInSydney, nice idea but you can't control the timing aspect so well with mutliple files`
`14:18.45`	`BorjaGVO`	`I think it'll be the easiest one...`
`14:19.44`	`BorjaGVO`	`ChrisInSydney: sequential...hmmm...how can I do that?`
`14:19.45`	`ChrisInSydney`	`[TK]D-Fender: True.`
`14:20.13`	`ChrisInSydney`	`musiconhold.conf`
`14:20.18`	`BorjaGVO`	`yeah`
`14:20.22`	`BorjaGVO`	`but what about the timing?`
`14:20.32`	`*** join/#asterisk GeoGeek (~steve-o_@69.26.219.66)`
`14:21.09`	`ChrisInSydney`	`random=no`
`14:21.22`	`BorjaGVO`	`wouldn't that be the same thing that Fender said?...well..almost the "same" thing..`
`14:21.26`	`ChrisInSydney`	`use audacity to chop up the music and create your files`
`14:21.32`	`ChrisInSydney`	`or have one big file`
`14:21.35`	`[TK]D-Fender`	`^`
`14:21.41`	`ChrisInSydney`	`or a few big files`
`14:21.56`	`ChrisInSydney`	`and then you can choose random`
`14:22.11`	`ChrisInSydney`	`just make sure you pick up the call before you run out of moh file`
`14:22.40`	`ChrisInSydney`	`[TK]D-Fender: Does a large moh file chew RAM resources ??`
`14:22.52`	`[TK]D-Fender`	`Dunno....`
`14:22.54`	`ChrisInSydney`	`or is it clever and just streams from fole`
`14:23.00`	`*** join/#asterisk wonderworld (~w@dsdf-4db530da.pool.mediaWays.net)`
`14:23.01`	`BorjaGVO`	`I think I'll create one file with music and the announcements integrated already in it..and then set it as MoH...`
`14:23.34`	`ChrisInSydney`	`probably resource hungry, as a guess as AFAIK you need to restart to get new moh files to work`
`14:24.00`	`BorjaGVO`	`Alright..I'll give it a try..thanks guys for the ideas`
`14:24.04`	`ChrisInSydney`	`BorjaGVO: There are more than once way to skin these cats,`
`14:24.17`	`ChrisInSydney`	`main thing is that you do at least one`
`14:24.21`	`ChrisInSydney`	`you can always go back`
`14:24.27`	`ChrisInSydney`	`good luck`
`14:24.51`	`BorjaGVO`	`ok, thanks`
`14:25.14`	`[TK]D-Fender`	`ChrisInSydney, Nooooo.. don't skin them ... that's where all the nutrients are!`
`14:25.38`	`*** join/#asterisk WolWid (WolWid@pD9ED50DD.dip.t-dialin.net)`
`14:25.43`	`ChrisInSydney`	`but I've worn out my bunny slippers`
`14:25.47`	`ChrisInSydney`	`:D`
`14:27.11`	`ChrisInSydney`	`So back to my queuing issue, with the cell phone / mobile being part of the queue as a Local/ agent`
`14:28.12`	`ChrisInSydney`	`use the Dial app with a shorter time out and use the customised privact options to set the ${DIALSTATUS } and chose what to do from there`
`14:28.26`	`ChrisInSydney`	`I can 1 accept the call`
`14:28.43`	`ChrisInSydney`	`2 reject the call and leave it in the queue`
`14:29.04`	`*** join/#asterisk blee (~blee@72.188.117.219)`
`14:29.09`	`ChrisInSydney`	`I still need to pause myself otherwise it will ring me back, whcih I dont want to do`
`14:29.42`	`ChrisInSydney`	`So I guess I could have the next incoming call unpause me`
`14:31.11`	`*** join/#asterisk mjordan (~mjordan@nat/digium/x-gdjpflyhzbycrleg)`
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`14:35.27`	`edve`	`Anyone knows where is the record all calls in Freepbx ?`
`14:36.01`	`[TK]D-Fender`	`~freepbx`
`14:36.01`	`infobot`	`[~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`
`14:36.02`	`[TK]D-Fender`	`^^`
`14:36.19`	`[TK]D-Fender`	`edve, And there is no "all calls" option. There is a checkbox per extension.`
`14:38.10`	`*** join/#asterisk serafie1 (~erin@76.73.167.231)`
`14:38.36`	`ChrisInSydney`	`[TK]D-Fender: I guess I could have a Local/ agent that doesn't actually dial or connect, and use that dialplan fork to reset paused agents. Use astdb to store times that the external agents were paused and unpause when past the expiry`
`14:39.10`	`ChrisInSydney`	`not too much of a kludge :-/`
`14:39.20`	`Greenlight`	`Load average now hitting 65 and effecting call quality, yet the Asterisk process is only 150%. What else could be causing it?`
`14:39.24`	`[TK]D-Fender`	`ChrisInSydney, On reject I'd launch a background script that'll sleep and then unpause on timeout`
`14:39.52`	`[TK]D-Fender`	`ChrisInSydney, Better than hacking your dialplan to death and more reliable as to the schedule`
`14:39.54`	`ChrisInSydney`	`did that, but when the call hangs up, that script also hangs up`
`14:40.03`	`[TK]D-Fender`	`chris_n, BACKGROUND <-`
`14:41.15`	`*** join/#asterisk serafie (~erin@76.73.167.231)`
`14:41.20`	`ChrisInSydney`	`[TK]D-Fender; Cheers`
`14:41.52`	`ChrisInSydney`	`back to school ;-)`
`14:44.42`	`ChrisInSydney`	`[TK]D-Fender: I'm confused`
`14:44.52`	`ChrisInSydney`	`not too hard to do...but`
`14:45.29`	`*** join/#asterisk fisted (~fisted@unaffiliated/fisted)`
`14:46.23`	`ChrisInSydney`	`Are you talking about using a system call and a .sh script to control via CLI / AGI or similar ??`
`14:47.10`	`[TK]D-Fender`	`yes`
`14:47.19`	`ChrisInSydney`	`ahh`
`14:47.39`	`[TK]D-Fender`	`System(doyouseethefuckingampersandafterme.sh &)`
`14:47.40`	`[TK]D-Fender`	`:)`
`14:48.34`	`ChrisInSydney`	`Rather than System(forgotthatfuckingampersand.sh)`
`14:48.38`	`ChrisInSydney`	`:D`
`14:49.36`	`ChrisInSydney`	`makes sense`
`14:53.17`	`ChrisInSydney`	`asterisk -rx or agi ??`
`14:53.32`	`ChrisInSydney`	`asterisk -rx might be easier`
`14:55.15`	`ChrisInSydney`	`[TK]D-Fender: whatcha think ?`
`14:55.37`	`[TK]D-Fender`	`ChrisInSydney, "rx" would be easier`
`14:55.43`	`ChrisInSydney`	`cheers`
`14:57.41`	`ChrisInSydney`	`1 accept the call`
`14:58.14`	`ChrisInSydney`	`2 leave the call in the queue and pause for <time yet to be determined probably 2 minutes>`
`14:58.16`	`*** join/#asterisk fisted (~fisted@unaffiliated/fisted)`
`14:58.35`	`ChrisInSydney`	`3 leave the call in the queue and pause for <time yet to be determined probably 10 minutes>`
`14:58.47`	`ChrisInSydney`	`4 leave the call in the queue and log out of the queue`
`15:00.08`	`p3nguin`	`5 kill dash nine`
`15:00.45`	`ChrisInSydney`	`Its basically for a support queue that needs to flick over to a mobile if the guys in the office dont pick up in time`
`15:01.08`	`*** join/#asterisk skrusty (~ksrusty@77-44-25-149.xdsl.murphx.net)`
`15:01.20`	`ChrisInSydney`	`p3nguin: 5 system(/sbin/shutdown -g0 -y)`
`15:01.27`	`jaytee`	`anyone used a Digium T122B with an Intel D525 Dual Core Atom mini-ITX board?`
`15:01.56`	`ChrisInSydney`	`jaytee. I have on a VIA board`
`15:02.14`	`ChrisInSydney`	`whats the challenge ?`
`15:02.15`	`*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)`
`15:03.29`	`*** join/#asterisk dvossel (~dvossel@24.42.230.136)`
`15:03.43`	`*** part/#asterisk dvossel (~dvossel@24.42.230.136)`
`15:03.57`	`jaytee`	`I've used the Intel D525 as an Asterisk server pure SIP before with no problems. Wondered if anyone tried it with a T1 card and whether the D525 board can handle the T1 PCI card without issues.`
`15:04.12`	`ChrisInSydney`	`cant see why not`
`15:04.37`	`ChrisInSydney`	`Actually I think I have a customer with one in a Supermicro server too`
`15:04.43`	`ChrisInSydney`	`no issues`
`15:05.06`	`ChrisInSydney`	`15 active channels on the E1 and 10 on the system with a VIA board`
`15:05.17`	`ChrisInSydney`	`E1 too`
`15:06.40`	`jaytee`	`ChrisInSydney, thanks`
`15:06.51`	`ChrisInSydney`	`no stress`
`15:11.37`	`ChrisInSydney`	`2am here`
`15:11.39`	`ChrisInSydney`	`night all`
`15:11.53`	`ChrisInSydney`	`cheers for the help`
`15:13.03`	`*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)`
`15:18.12`	`jaytee`	`so are all the Digium T1/E1 cards now low-profile only?`
`15:20.27`	`p3nguin`	`I think you can get a regular bracket or a low-pro bracket.`
`15:21.33`	`*** join/#asterisk Rokfan (~Rokfan@D522448D.static.ziggozakelijk.nl)`
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`15:26.22`	`jeffspeff`	`anybody have experience using openmeetings?`
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`15:35.05`	`*** join/#asterisk Shane-S (~chatzilla@c-68-46-130-94.hsd1.nj.comcast.net)`
`15:36.30`	`Shane-S`	`?: Would asterisk handle having say 5 phone lines, but customers only dial a main number, and they are somehow swapped to an open line, so customer B can dial the main number and not get a busy signal. To mean that seem like carrier end.`
`15:36.57`	`p3nguin`	`shane-s: What kind of phone lines?`
`15:37.23`	`Shane-S`	`No clue yet, just a thought, say it were POTS/Analog`
`15:37.48`	`p3nguin`	`If the telco rolls calls over to other lines, there won't be a problem.`
`15:38.14`	`p3nguin`	`Or you can go with SIP and just use a single DID which has multiple channels.`
`15:39.40`	`Shane-S`	`okay, what about using Google Voice to play with it? Say I get 2-3 voice accounts to play with.`
`15:39.41`	`*** join/#asterisk unicron (~unicron@the.wrong.domain.name)`
`15:40.12`	`p3nguin`	`You'll have to share all of the phone numbers with the people who will call you.`
`15:40.24`	`p3nguin`	`They'll call one, get a busy signal, then call another.`
`15:40.27`	`p3nguin`	`Not ideal.`
`15:40.54`	`p3nguin`	`If you were going to pay for five POTS lines, you can surely afford to buy a single DID and use VoIP.`
`15:41.13`	`Shane-S`	`p3nguin: so where I work they have analog/pots and we just give out our main number, I am guessing Verizon is doing that callrolling?`
`15:41.19`	`p3nguin`	`Yes.`
`15:42.12`	`*** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-mnwcgnxralfgjxyj)`
`15:42.32`	`Shane-S`	`p3nguin: alright thanks, that is a huge help for me, I was not sure how/where that is done, and it didn't make sense that a PBX could freely switch a call to another line without me knowing...cause heck I could program 1-900 in there :P`
`15:43.12`	`Shane-S`	`I also had not idea what it was called or what to look for on the SIP/DID side`
`15:43.28`	`p3nguin`	`You need an ITSP.`
`15:43.34`	`p3nguin`	`~itsp`
`15:43.34`	`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.`
`15:43.48`	`p3nguin`	`You'll want a DID.`
`15:43.50`	`p3nguin`	`~did`
`15:43.50`	`infobot`	`well, did is Direct Inward Dialing, or just a phone number`
`15:44.03`	`*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)`
`15:44.18`	`p3nguin`	`If you have a phone number that you want to keep, you can port your number into the ITSP that you choose.`
`15:44.24`	`Shane-S`	`p3nguin: would Comcast's VOIP offering with a modem be a ITSP?`
`15:44.59`	`p3nguin`	`Technically, they are acting as your ITSP.`
`15:45.11`	`*** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net)`
`15:45.32`	`drmessano`	`Comcast doesn't sell VoIP.. they sell POTS lines`
`15:45.42`	`drmessano`	`The transport is irrelevant`
`15:46.14`	`p3nguin`	`They aren't going to give you five channels on your single DID, though, I'm sure.`
`15:46.37`	`Shane-S`	`drmessano: gotcha, so if I went with them, I would have to get say 5 modems, and then a FXS card, even though the are VOIP because they use those modems?`
`15:46.53`	`p3nguin`	`What a mess.`
`15:47.19`	`p3nguin`	`With a traditional ITSP, you buy your DID, they send it to you over SIP (or IAX2), and you network it directly into your IP PBX.`
`15:47.20`	`drmessano`	`YOUR access to them is not VOIP. They provide you with a FXS POTS INTERFACE. Forget that they are using IP for transport`
`15:48.08`	`drmessano`	`Comcast voice is a huge waste of time, IMO`
`15:48.23`	`p3nguin`	`Most Comcast customers using the phone service have no idea that it isn't a regular phone line. They plug their regular phone into the FXS jack on the cable modem and it works like it did when they had AT&T.`
`15:48.44`	`p3nguin`	`Or at least in principle it works the same.`
`15:48.58`	`*** join/#asterisk Vince-0 (~Vincent@41-132-156-117.dsl.mweb.co.za)`
`15:49.00`	`*** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher)`
`15:49.07`	`p3nguin`	`"Work" is a relative term. :(`
`15:49.35`	`Shane-S`	`I only mentioned it because we have then for internet, so I could "bundle" for saving,s but it does sound a mess`
`15:49.52`	`p3nguin`	`You probably won't save that much anyway.`
`15:50.12`	`p3nguin`	`Are you wanting to set up a business phone or a home phone with those five lines?`
`15:50.28`	`Shane-S`	`if I went with an ITSP, do they provide how much bandwidth a line needs?`
`15:50.39`	`p3nguin`	`Like the requirements?`
`15:50.59`	`*** join/#asterisk elico (~Thunderbi@bzq-109-67-228-219.red.bezeqint.net)`
`15:51.05`	`Shane-S`	`for now I will do it at home to "play". I have google voice setup now, but I am trying to educate myself against the companies I am asked to call in for a phone update`
`15:52.23`	`p3nguin`	`If that was what you meant, figure on ~80k per second per call using the ulaw codec.`
`15:54.14`	`Shane-S`	`p3nguin: okay thanks, we have a 10Mb connection with a FiOS failover at 25Mb (I know should use the 25Mb as primary...but Verizon "stole" our fiber twice now and took 2 weeks each time to track down that a tech stole our working pair)`
`15:54.42`	`SeRi`	`ouch thats not good`
`15:54.51`	`jacekowski`	`i'm using digium phones with DPMA module and i can't get presence to work`
`15:54.53`	`p3nguin`	`seri: Back to work, you!`
`15:55.07`	`jacekowski`	`whatever i do, phone does not show up in subscriptions`
`15:55.10`	`Shane-S`	`SeRi: Poor you!`
`15:55.29`	`coppice`	`25M on fibre is like a snail on an autobahn :-\`
`15:56.01`	`Shane-S`	`coppice: I can't complain both lines were free :P`
`15:56.17`	`coppice`	`oh, well, at least its good value`
`15:57.38`	`Shane-S`	`Verizon came into our town, and the deal was the had to give schools and municipalities 1 free internet connection. They contract states non-networked, but the best part was, they came out with a their wi-fi router...and I looked at the tech and was like "uhhh?"...`
`15:58.35`	`p3nguin`	`They wanted to add you to the national wifi network?`
`15:58.50`	`p3nguin`	`Free webz for everyone!`
`15:59.02`	`coppice`	`"Verizon came into our town" sounds deeply sinister`
`15:59.05`	`Shane-S`	`p3nguin: no its home router, home wi-fi`
`15:59.18`	`*** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net)`
`15:59.34`	`Shane-S`	`I called and they said they know it will be networking, just contract wise they can't state that as it is a "free" connection`
`16:00.05`	`SeRi`	`p3nguin: LOL.`
`16:00.24`	`p3nguin`	`I don't get it. What did you expect to give you, and what are they giving you that is different?`
`16:00.35`	`p3nguin`	`s/give you/get/`
`16:00.36`	`*** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net)`
`16:00.56`	`Shane-S`	`coppice: Verizon is sinister with their fiber, my home town, 7 miles from here, doesn't have it...they are very picky where this fiber is going`
`16:01.00`	`coppice`	`I get an image of a Verizon van driving into town, and all the foliage shrivelling as it passes`
`16:01.12`	`*** join/#asterisk Widler (47c8d864@gateway/web/freenode/ip.71.200.216.100)`
`16:01.53`	`Shane-S`	`coppice: lol, I can't say Comcast is much better these days`
`16:03.08`	`Shane-S`	`I do agree with the 25Mb on fiber though...the best part I found, was the line running INTO my FiOS cabinet from outside...sure as heck looks like coaxial.`
`16:03.19`	`*** join/#asterisk fisted (~fisted@unaffiliated/fisted)`
`16:03.37`	`Shane-S`	`its the same thickness and the same termination end...and it it threaded into the box.`
`16:05.04`	`Shane-S`	`meanwhile my dead T1 box...has the tell tale 2 yellow colored wires coming from a box with 1/4" thick black wire with a yellow stripe going in. Yet that could only do 1.44Mbs according to the installer.`
`16:06.13`	`Shane-S`	`begins to wonder if I maybe do drugs, because the fiber IN my building does 1Gbs MINIMUM, its actually my SFP modules I think that limit it.`
`16:06.20`	`*** join/#asterisk vlad_starkov (~vlad_star@83.149.8.120)`
`16:06.52`	`*** join/#asterisk spditner (~simon@206-248-134-56.dsl.teksavvy.com)`
`16:07.40`	`spditner`	`Has anyone seen an issue where an unreplied conntrack entry becomes stuck in their router, where it does not have the opporunity to expire due to asterisk banging at it with further register and option packets?`
`16:09.03`	`Chainsaw`	`spditner: Can't say I have. It's worth trying SIP over TCP if your devices support it.`
`16:09.21`	`p3nguin`	`shane-s: Do they have an ONT on the outside of the building and then run a coax into the building into another box?`
`16:09.27`	`Chainsaw`	`spditner: Those don't often get in that limbo state stateless UDP connections do.`
`16:09.43`	`Chainsaw`	`spditner: (And playing devil's advocate for a moment, your NAT table is too small and you need a bigger router)`
`16:09.54`	`*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)`
`16:11.26`	`spditner`	`Chainsaw: if I weren't seeing it across multiple devices at multiple sites, I'd go with changing the router and service settings about, but sadly, not isolated for me.`
`16:11.41`	`Shane-S`	`p3nguin: there could be an ONT on the poll...but nothing on the outside of the building. I will have to climb up in the cabinet area it is mounted and confirm that coaxial, but I am 90% certain it is the FiOS from outside as they stapled it to the board, and it screws into the box under the panel door where the Cat5e connects to the router`
`16:11.50`	`jacekowski`	`spditner: it happens apparently`
`16:12.25`	`jacekowski`	`spditner: i've seen a lot of info about it on voip-info.org wiki`
`16:12.40`	`p3nguin`	`shane-s: It sounds like your ONT is in that cabinet, but I can't imagine then calling it FiOS if they are HFC.`
`16:12.41`	`jacekowski`	`http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html`
`16:13.03`	`Shane-S`	`p3nguin: aside from a DID, if I want to make calls out with the ITSP, is that something else?`
`16:13.54`	`[TK]D-Fender`	`Shane-S, Depends on the product you pay for`
`16:14.20`	`Shane-S`	`[TK]D-Fender: K, thanks, I will look at the vendor sites`
`16:14.23`	`p3nguin`	`I've only seen the single strand of fiber running from the manholes under the ground to the gateway (ONT) on the outside of the building. But we also don't have Verizon FiOS here, it's a much smaller company.`
`16:14.23`	`[TK]D-Fender`	`Shane-S, A DID is jsut a phone number itself. What they include as a bundled "product" depends on the vendor\`
`16:14.51`	`p3nguin`	`shane-s: The DID provides a way for people to call you from the PSTN.`
`16:14.56`	`Chainsaw`	`spditner: It's because UDP retransmissions. It's a stateless protocol, so Asterisk can't tell the difference between "minor packetloss" and "host gone away".`
`16:15.22`	`p3nguin`	`For termination, the ITSP puts your VoIP call onto the PSTN much like the way it takes your call from the PSTN and puts it onto VoIP to send to you.`
`16:15.29`	`Chainsaw`	`spditner: Prevent the host going away, go for a lower qualify timer, or opt for SIP over TCP so you can tell what's really going on.`
`16:15.55`	`Chainsaw`	`spditner: Or in fact, higher qualify timer so that the transmissions don't happen as often. Downside is the host spends longer in limbo.`
`16:16.49`	`spditner`	`Chainsaw: Right, that was part of my solution, but some higher-end solutions have really high timeouts, on the order of 10+ minutes`
`16:17.38`	`spditner`	`Chainsaw: What I am guessing at this time is that asterisk doesn't see the WAN go down and IP address change so that it does something like transmit a response to a REGISTER or something, to which the remote system doesn't reply, and gets stuck in this state.`
`16:24.04`	`*** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net)`
`16:25.40`	`Chainsaw`	`spditner: Yes, UDP going away means retransmits. Aggressive, on the default settings.`
`16:25.52`	`Chainsaw`	`spditner: If you're sure you never lose any UDP because you don't run your ports hot... you could tone them down a bit?`
`16:27.58`	`*** part/#asterisk WolWid (WolWid@pD9ED50DD.dip.t-dialin.net)`
`16:30.57`	`*** join/#asterisk fisted_ (~fisted@unaffiliated/fisted)`
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`16:43.42`	`*** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net)`
`16:43.52`	`Greenlight`	`Any recommendations on someone/somewhere UK based to give paid for consultancy/support on a few specific issues I'm getting with Asterisk?`
`16:44.12`	`Chainsaw`	`Greenlight: Outside working hours I'd be happy to assist you.`
`16:44.24`	`Chainsaw`	`Greenlight: During working hours I have a day job (which involves Asterisk, among other things).`
`16:45.19`	`Greenlight`	`Okay - my servers are live and in use from 10AM - 9PM when the issues can be observed.`
`16:45.28`	`_Corey_`	`Greenlight: If you submit a request on Digium's website, I'm sure they can refer you to a partner in your area.`
`16:45.42`	`Chainsaw`	`Unless you want someone independent of course.`
`16:45.54`	`Chainsaw`	`(6PM-9PM would work well for me)`
`16:46.08`	`Greenlight`	`I really wanted someone who knows their stuff, not just an Asterisk reseller, or else they'll be of no help`
`16:46.18`	`*** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com)`
`16:47.53`	`Greenlight`	`It's some rather specific issues I'm getting, and if they're fixed I've got almost a blank cheque, but on the flipside I don't want to pay out hundreds of pounds to someone for just looking at it.`
`16:48.53`	`Greenlight`	`How do these things generally work`
`16:50.56`	`[TK]D-Fender`	`Greenlight, First you become a whole lot less "general"`
`16:51.47`	`Greenlight`	`I mean in terms of getting a consultant in to help - whats the "done thing" in terms of financials etc?`
`16:52.21`	`*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)`
`16:52.21`	`*** mode/#asterisk [+o sruffell] by ChanServ`
`16:52.35`	`[TK]D-Fender`	`Greenlight, Depends who you deal with. You're asking out loud in a channel full of people from throughout the world.`
`16:52.59`	`Greenlight`	`Fair enough`
`16:53.04`	`[TK]D-Fender`	`Greenlight, Maybe you could jsut tell us what your problems are so that those how aren't capable of specific aspects know this up front beofre proposing service/solutions.`
`16:53.10`	`slav3_kitten`	`[TK]D-Fender, what is it with a lot of these voip services recording calls in their terms of use`
`16:53.20`	`[TK]D-Fender`	`Greenlight, Don't try to define the terms before the problem.`
`16:53.32`	`[TK]D-Fender`	`slav3_kitten, Never heard of it....`
`16:53.44`	`SeRi`	`slav3_kitten: who has that on their TOS?`
`16:54.22`	`slav3_kitten`	`let me try to remember, it was late last night an maybe i misread`
`16:54.53`	`SeRi`	`slav3_kitten: You probably did. I have never heard of that. And even on a TOS. I think that's inlegal.`
`16:54.54`	`[TK]D-Fender`	`slav3_kitten, You should really be actually sure of this stuff before asking it out loud...`
`16:55.05`	`SeRi`	`^^+1^^`
`16:55.39`	`slav3_kitten`	`^^+2^^`
`16:56.01`	`*** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain)`
`16:56.03`	`slav3_kitten`	`flowroute http://www.flowroute.com/legal/termsofuse/ item 7`
`16:56.09`	`slav3_kitten`	`consent to monitoring`
`16:57.11`	`SeRi`	`is reading`
`16:57.41`	`*** join/#asterisk anonymouz666 (~anonymouz@189-25-53-120.user.veloxzone.com.br)`
`16:58.13`	`*** join/#asterisk vinhdizzo (~vinh@128.195.52.49)`
`16:58.45`	`Qwell`	`slav3_kitten: Where's the part about recording calls?`
`16:58.59`	`*** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu)`
`16:59.06`	`slav3_kitten`	`they say they may monitor calls, which to monitor means it's recorded someplace`
`16:59.08`	`SeRi`	`Qwell: I was about to ask the same question`
`16:59.16`	`SeRi`	`They are talking about phisical server monitoring`
`17:00.23`	`[TK]D-Fender`	`<slav3_kitten> they say they may monitor calls, which to monitor means it's recorded someplace <- Get a new dictionary`
`17:00.24`	`slav3_kitten`	`well what does that entail then?`
`17:00.47`	`*** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4)`
`17:01.00`	`[TK]D-Fender`	`slav3_kitten, I can monitor traffic in a GENERAL way like BW stats. THAT is "monitoring".`
`17:01.06`	`[TK]D-Fender`	`Listening LIVE is montoring.`
`17:01.39`	`slav3_kitten`	`[TK]D-Fender, no need to be mean about it. i obviously misunderstood it, you act as if you've never misunderstood something in your life.`
`17:01.45`	`[TK]D-Fender`	`er. However, anyone using this server agrees that Flowroute may monitor the server contents periodically to (1) comply with any necessary laws, regulations or other governmental requests, or (2) to operate the server properly or to protect itself and its users. <--- #1`
`17:02.40`	`drmessano`	`I also don't recall "`
`17:02.43`	`[TK]D-Fender`	`slav3_kitten, You jumped to a heavy conclusion without thinking more than half a second. Spend that half a second. And a few more. Coffee is AWESOME. Only thing better than coffee is MORE coffee.`
`17:02.52`	`drmessano`	`I also don't recall "A lot of these services" that record calls`
`17:03.14`	`[TK]D-Fender`	`slav3_kitten, Got more samples for the "a lot"?`
`17:03.20`	`SeRi`	`lol`
`17:03.30`	`drmessano`	`One TOS from one provider with a misunderstood pointer to call monitoring hardly paints a picture of call recording across multiple ITSPs`
`17:04.20`	`slav3_kitten`	`i've looked at a bunch of services and saw similar things on them. they are just the example i had handy.`
`17:04.30`	`[TK]D-Fender`	`drmessano, You know from what this one guy heard, the female body has a way of shutting that whole thing down anyway...`
`17:04.38`	`drmessano`	`LOL`
`17:04.40`	`drmessano`	`Yep`
`17:04.47`	`drmessano`	`Is this JUSTIFIABLE call recording?`
`17:04.58`	`drmessano`	`because that's allowed`
`17:05.05`	`[TK]D-Fender`	`slav3_kitten, Well go read them again and let us know if you still have concerns over them...`
`17:05.17`	`slav3_kitten`	`i can understand recording with a warrant an such`
`17:05.42`	`slav3_kitten`	`i don't do anything illegal i just don't like the idea of someone listening in on my conversations willy nilly.`
`17:05.42`	`drmessano`	`I wonder if they even have the facilities to do such`
`17:06.32`	`drmessano`	`Flowroute actually doesn't handle any call media`
`17:06.55`	`slav3_kitten`	`how's that work then drmessano?`
`17:07.03`	`drmessano`	`It's all passed on to their upstream provider..`
`17:07.06`	`drmessano`	`They don't touch the media`
`17:07.17`	`slav3_kitten`	`nod`
`17:07.45`	`*** join/#asterisk sickgrinder (~sickgrind@rrcs-97-76-33-146.se.biz.rr.com)`
`17:10.33`	`SeRi`	`paranoya at best....`
`17:10.35`	`slav3_kitten`	`[TK]D-Fender, please forgive my lack of complete and total understanding of VoIP technology but you berating me for such really precludes any actual learning... i happened to think for more then 5 seconds, i read it, read it again, and got the impression that they will at will monitor phone calls which really bothered me since i have an expectation of privacy when calling someone. i made a mistake, i fully admit that, i'm in`
`17:10.35`	`slav3_kitten`	`fact not perfect.`
`17:11.27`	`[TK]D-Fender`	`slav3_kitten, I ended my statement with a joke. We're all over it. Come join us.`
`17:11.32`	`SeRi`	`slav3_kitten: dont be sensitive. [TK]D-Fender is just that way. rough up. with the beating you just got you learn something from him.`
`17:12.39`	`slav3_kitten`	`SeRi, the majority of what i learned is you don't ask noob questions in here or make statements based on a misunderstanding of a number of ToS agreements because it will result in a proper brow beating by those with more knowledge.`
`17:13.03`	`lantizia`	`slav3_kitten, you're like new to IRC then?`
`17:13.07`	`lantizia`	`:D`
`17:13.25`	`SeRi`	`slav3_kitten: LO. Totally not what just happen`
`17:13.32`	`SeRi`	`s/LO/LOL/`
`17:13.41`	`jaytee`	`beware the infamous ClueBat(tm)!`
`17:13.43`	`slav3_kitten`	`i figured that's what he meant`
`17:13.55`	`SeRi`	`You got "politically corrected"`
`17:14.06`	`slav3_kitten`	`uh huh`
`17:15.04`	`jaytee`	`"You need to follow the chain of command." "Well, what's the chain of command?" "It's the chain I go and beat you with until you learn who is in command!"`
`17:15.28`	`slav3_kitten`	`lol`
`17:16.40`	`*** join/#asterisk fritz09 (~Adium@pop1-3530.catv.wtnet.de)`
`17:16.47`	`fritz09`	`hi`
`17:19.54`	`slav3_kitten`	`jaytee, that is a funny saying :D`
`17:19.55`	`WIMPy`	`wonders if Greenlight solved his issue himself while trying to describe it.`
`17:20.47`	`Greenlight`	`wishes`
`17:21.38`	`[TK]D-Fender`	`Greenlight, Are you going to actually jsut TELL us what your needs are?`
`17:24.17`	`Greenlight`	`SOrry, I've taken the issue to a /w and and discussing it further there.`
`17:24.37`	`[TK]D-Fender`	`Greenlight, Best of luck then...`
`17:24.49`	`Greenlight`	`Thanks :)`
`17:32.41`	`*** join/#asterisk JohnnyAsterisk (~cianmaher@154.50.194.130)`
`17:36.21`	`drmessano`	`What makes the difference in the response to your question is purely in how it is asked. There is a big difference between "Why does Flowroute record all my calls?" and "Does this mean that flowroute can record my calls?"`
`17:36.47`	`slav3_kitten`	`what's a /w?`
`17:37.10`	`drmessano`	`The first one is a question based on an ill-fed CONCLUSION, which will only irritate those with more sense.`
`17:37.23`	`drmessano`	`CONCLUSIONS are the easiest way to get trolled on IRC`
`17:38.16`	`drmessano`	`Ask someone for HELP, and they will provide it. Tell someone about your belief system, and they attempt to DESTROY it.`
`17:39.12`	`*** join/#asterisk Widler (47c8d864@gateway/web/freenode/ip.71.200.216.100)`
`17:39.44`	`Widler`	`I cant get SQLite3 what's an alternative`
`17:39.52`	`Qwell`	`for what?`
`17:40.20`	`Widler`	`configure: WARNING: * Asterisk now uses SQLite3 for the internal Asterisk database. configure: WARNING: * Please install the SQLite3 development package.`
`17:40.27`	`Qwell`	`There isn't one.`
`17:40.37`	`Qwell`	`Why "can't" you get it?`
`17:41.06`	`drmessano`	`Widler: Did you install the package I told you to install?`
`17:41.15`	`Widler`	`asteriskpbx@msi /usr/src/asterisk/asterisk-11 $ sudo apt-get install SQLite3 Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package SQLite3`
`17:41.27`	`drmessano`	`Thats NOT what I told you to install`
`17:41.37`	`drmessano`	`libsqlite3-dev (I believe it is)`
`17:42.01`	`slav3_kitten`	`drmessano, i thought we dropped this issue a while ago. i got the point an understand what you're saying`
`17:42.07`	`Greenlight`	`sqlite3-devel on CentOS`
`17:42.37`	`Widler`	`got it`
`17:42.50`	`Widler`	`I'm sorry. new to linux`
`17:43.13`	`drmessano`	`New to reading, as well.`
`17:44.11`	`slav3_kitten`	`drmessano, reading can be hard`
`17:44.15`	`p3nguin`	`deja vu`
`17:44.21`	`*** join/#asterisk bchia (~Adium@nat/digium/x-iuvwuezhghyphtnr)`
`17:44.30`	`p3nguin`	`I seem to remember going over this yesterday.`
`17:45.01`	`Widler`	`haha guys`
`17:46.45`	`drmessano`	`I know I supplied that info last night`
`17:47.00`	`p3nguin`	`And I advised on how to search for packages.`
`17:47.10`	`drmessano`	`and I ate a sandwich`
`17:47.26`	`p3nguin`	`I knew I made a sandwich, and I didn't remember eating it.`
`17:47.45`	`drmessano`	`Meatball with duck sauce and capers?`
`17:48.02`	`p3nguin`	`Nah, capers are weird.`
`17:48.13`	`drmessano`	`HA`
`17:48.26`	`drmessano`	`My wife says the same thing.`
`17:48.31`	`SeRi`	`lol`
`17:48.34`	`Widler`	`the package search command again please`
`17:48.40`	`drmessano`	`She says capers are just weird. Thats all she can tell me`
`17:48.45`	`p3nguin`	`apt-cache search sqlite`
`17:49.15`	`drmessano`	`Also, google`
`17:49.57`	`drmessano`	`If you're talking about dependencies, you're generally looking for a development package.. So "sqlite debian development package" would get you there`
`17:51.02`	`p3nguin`	`I'm still inclined to believe that installing the asterisk package would cause the necessary sqlite package to get installed, thanks to apt's ability to solve deps.`
`17:51.26`	`p3nguin`	`I could be mistaken, since I don't typically touch things that look like Debian, but I doubt it.`
`17:51.42`	`drmessano`	`I dont believe this is a package install`
`17:52.10`	`Widler`	`i'm following along asteriskdocs.org it doesnt's say to make samples. should i do it anyways since I don't know what i'm doing`
`17:53.04`	`p3nguin`	`If you want sample files, make samples. If you already have configs in place, do not make samples because the samples will overwrite your configs.`
`17:53.31`	`SeRi`	`Thinks that "look like" Debian make me sick... ie: ubuntu`
`17:53.43`	`SeRi`	`s/Think/Things/`
`17:53.50`	`*** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com)`
`17:55.01`	`drmessano`	`I refuse to even build my kernel unless I am using the kernel I built using a kernel that was built with a kernel from Linus' dev workstation`
`17:55.31`	`drmessano`	`All my machines are based on that purebred kernel`
`17:55.36`	`drmessano`	`I call it "Obelisk"`
`17:56.26`	`*** join/#asterisk fisted (~fisted@unaffiliated/fisted)`
`17:58.03`	`drmessano`	`If you check the source of Obelisk, you will see where Linus broke one of the PCMCIA kernel mods when he dropped ketchup on his keyboard. Lines 4958475847 thru 4958475850 are just a bunch of random letters and numbers from him dabbing a napkin on the keyboard`
`17:58.10`	`drmessano`	`So.. Yeah. I am elite`
`17:58.42`	`p3nguin`	`Backspace key was not working that day?`
`17:59.15`	`drmessano`	`He apparently didn't realize it, and who the hell would notice something broken with PCMCIA?`
`17:59.28`	`WIMPy`	`Real men don;t need a backspace key.`
`17:59.40`	`*** join/#asterisk navaismo (~navaismo@189.191.12.116)`
`17:59.48`	`WIMPy`	`cat\|gcc`
`18:04.19`	`SeRi`	`WIMPy: lol`
`18:04.40`	`unicron`	`drmessano: where can i download Obelisk`
`18:08.13`	`SeRi`	`waz up unicron`
`18:13.56`	`drmessano`	`unicron: You need to get it from Linus.`
`18:14.15`	`drmessano`	`He'll deny it exists, and also the PCMCIA thing.`
`18:14.21`	`drmessano`	`:(`
`18:16.08`	`unicron`	`sup seri`
`18:16.20`	`unicron`	`trying to debug thunderbird issue for a user atm`
`18:16.21`	`seanbright`	`anyone know if there is anything weird about kernel timing on a dell poweredge 2850?`
`18:16.37`	`seanbright`	`like low resolution for example`
`18:16.50`	`unicron`	`drmessano: may i have your copy?`
`18:18.00`	`drmessano`	`unicron: That wouldn't be right`
`18:19.27`	`unicron`	`you tease, bragging about this awesome code you can't show anyone :(`
`18:19.46`	`SeRi`	`lol`
`18:20.52`	`drmessano`	`I'm sorry. I suck, I know :(`
`18:21.22`	`*** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com)`
`18:21.56`	`drmessano`	`~happyclownPBX`
`18:21.56`	`infobot`	`[HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone`
`18:22.21`	`drmessano`	`^_^`
`18:23.35`	`SeRi`	`sounds promissing.`
`18:24.08`	`SeRi`	`/s/promissing/promising/`
`18:27.22`	`edve`	`Anyone knows where is the record all calls in Freepbx ?`
`18:28.05`	`WIMPy`	`edve: Someone in #freepbx perhaps`
`18:28.09`	`slav3_kitten`	`wow 12gb...`
`18:32.00`	`drmessano`	`edve: This was answered for you, and you were redirected`
`18:34.24`	`*** join/#asterisk wonderworld (~w@dsdf-4db530da.pool.mediaWays.net)`
`18:37.27`	`*** join/#asterisk TimeRider (~steve@188.220.34.144)`
`18:39.49`	`SeRi`	`drmessano: Good memory`
`18:40.09`	`SeRi`	`He has been around here before looking for support regarding his freepbx setup`
`18:40.12`	`p3nguin`	`It was like an hour ago.`
`18:40.24`	`SeRi`	`and previous days`
`18:40.24`	`Katty`	`infobot: crittercam`
`18:40.24`	`infobot`	`hmm... crittercam is The Birdie Breakfast Buffet!!! http://www.ustream.tv/channel-popup/birdie-breakfast-buffet`
`18:40.26`	`[TK]D-Fender`	`I have answered this repeatedly. In BOTH channels`
`18:40.37`	`Katty`	`it's thanksgiving on crittercam tday!`
`18:40.45`	`Katty`	`sadly i've only had 1 cardinal :<`
`18:41.30`	`Katty`	`and the light is reflecting off the white blinds....must do something about that`
`18:41.49`	`[TK]D-Fender`	`Katty, Evidently you were not born a Catholic schoolboy ;)`
`18:42.12`	`Katty`	`no, no i was not.`
`18:42.25`	`Katty`	`but i detest the idea that anyone is born of any particular faith`
`18:42.27`	`*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)`
`18:42.27`	`*** mode/#asterisk [+o pabelanger] by ChanServ`
`18:42.34`	`Katty`	`regardless of your cutesy little comment. which was very cutesty btw`
`18:44.26`	`Nivex`	`Everyone is born an atheist.`
`18:45.23`	`p3nguin`	`Tell that to the ones who aren't.`
`18:45.56`	`_Corey_`	`Nivex: I'd disagree... I had an ominous looking mobile in the crib that I worshiped as a deity until age 3.`
`18:46.53`	`Katty`	`i think everyone is born not really caring`
`18:47.00`	`Katty`	`all they care about is being warm, and happy, and not hungry`
`18:47.01`	`_Corey_`	`:-)`
`18:47.11`	`Katty`	`beyond that, they don't really grasp what's going on`
`18:47.23`	`Katty`	`hell, they probably don't even grasp that much. just everything is cool, or cry`
`18:47.42`	`Katty`	`they might be able to recognize a parent`
`18:47.44`	`Nivex`	`amazing what how much a hug can fix`
`18:48.37`	`Katty`	`indeed.`
`18:48.42`	`Katty`	`even now. a hug fixes much.`
`18:50.12`	`n3hxs`	`They grasp the nipple for lunch.`
`18:55.08`	`UForgotten`	`This just in, n00bs in #asterisk cited for nipple grasping. film at 11?`
`18:57.49`	`navaismo`	`I'm calling to my gtalk peer configured in asterisk but asterisk do not answer, I can see the incoming xmpp request in the CLI but asterisk never respond. Hints to debug it?`
`18:59.58`	`SeRi`	`unicron: you around?`
`19:03.22`	`*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)`
`19:03.22`	`*** mode/#asterisk [+o malcolmd] by ChanServ`
`19:04.38`	`drmessano`	`navaismo: Asterisk 11?`
`19:05.28`	`navaismo`	`yep`
`19:06.00`	`drmessano`	`Make sure you're sending to an existing context`
`19:06.19`	`drmessano`	`and that the context has an '`
`19:06.24`	`drmessano`	`and that the context has an 's' extension`
`19:06.51`	`drmessano`	`I am not aware of any additional debug. I couldn't find any.. but those are the things I had broken`
`19:07.00`	`navaismo`	`yes, the context exist, and has the s extension`
`19:07.17`	`drmessano`	`Drop a 'noop' in there`
`19:07.57`	`p3nguin`	`I don't have an extension s, but I do have my email addresses of the gv users.`
`19:08.23`	`p3nguin`	`s/./ as extensions./`
`19:08.46`	`drmessano`	`p3nguin: Asterisk 11?`
`19:08.55`	`p3nguin`	`1.8`
`19:09.08`	`drmessano`	`Completely different`
`19:09.21`	`p3nguin`	`The calls go first to the email address, then, if those don't exist, they fall back on s.`
`19:09.26`	`file`	`provided you are using chan_motif, that is`
`19:09.42`	`drmessano`	`I guess I assumed..`
`19:09.57`	`drmessano`	`navaismo: You ARE using chan_motif, yes?`
`19:10.52`	`navaismo`	`yes`
`19:11.24`	`navaismo`	`hmm outbound call can reach the other end but never take the answer, issues with ports maybe?`
`19:11.36`	`file`	`if it's not doing anything that means your chan_motif is not configured properly`
`19:11.59`	`file`	`pastebin the motif.conf file`
`19:14.02`	`navaismo`	`it is doing, When I call from asterisk to gmail users i get the incoming call in the web client but after answer my asterisk keep ringing. On incoming call asterisk cant take the call i see the xmpp request but asterisk do not send the respond, 1 sec for the pb`
`19:14.34`	`drmessano`	`Incorrect protocol`
`19:14.55`	`drmessano`	`Sounds like 2 different issues`
`19:15.17`	`p3nguin`	`Sign out of the web.`
`19:15.35`	`drmessano`	`incorrect protocol on the outbound, misconfiguration on the inbound`
`19:15.41`	`loggiew`	`getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known`
`19:15.48`	`loggiew`	`anyone know why its passing that value`
`19:15.54`	`loggiew`	`or what I may have done wrong`
`19:16.18`	`p3nguin`	`How did you arrive at this?`
`19:16.35`	`p3nguin`	`No evidence of what is going on, just the conclusion that is is broken.`
`19:17.22`	`drmessano`	`navaismo: transport=google ; Since this is a Google Talk endpoint we want to offer Google Jingle for outgoing sessions <--- Is that what you have for the endpoint definition in motif.conf?`
`19:17.48`	`tm1000`	`navaismo: dont you use freepbx`
`19:18.56`	`navaismo`	`tm1000, no, but i have one freepbx test server`
`19:19.06`	`tm1000`	`just saying. we have a motif module`
`19:19.16`	`tm1000`	`didnt know if you were doing this by hand inside freepbx`
`19:19.33`	`tm1000`	`navaismo: you did add icesupport to rtp.conf yes?`
`19:19.59`	`file`	`tm1000, next version of Asterisk 11 will have that on by default :D`
`19:20.13`	`navaismo`	`tm1000, yep`
`19:20.23`	`tm1000`	`file: thanks josh. does it matter if we still add it to rtp.conf`
`19:20.25`	`drmessano`	`navaismo, transport=google?`
`19:20.29`	`file`	`tm1000, won't matter`
`19:20.29`	`navaismo`	`drmessano, http://pastebin.com/BT2isHyZ let me try that`
`19:21.01`	`drmessano`	`navaismo: You dont have a transport defined`
`19:21.02`	`tm1000`	`navaismo: drmessano this is pointless we can go through every single option. Just post your xmpp.conf and motif.conf and your exten context`
`19:21.12`	`drmessano`	`He just did`
`19:21.16`	`tm1000`	`oh`
`19:21.18`	`tm1000`	`haha`
`19:21.28`	`drmessano`	`and like I suspected, no transport`
`19:21.32`	`drmessano`	`so whatevs`
`19:21.39`	`tm1000`	`drmessano: yup`
`19:21.58`	`tm1000`	`drmessano: do people follow walkthroughs.....`
`19:22.19`	`navaismo`	`and now this let me increase the cpu thing`
`19:22.20`	`navaismo`	`[Nov 12 13:21:43] WARNING[9443]: chan_motif.c:2255 jingle_action_session_initiate: Failed to start PBX (call limit reached)`
`19:22.56`	`tm1000`	`navaismo: out of CPU`
`19:23.24`	`file`	`sits back and lets tm1000 and drmessano work`
`19:23.30`	`drmessano`	`service quakeserver stop`
`19:23.32`	`drmessano`	`or`
`19:23.51`	`drmessano`	`killall -9 chrome`
`19:23.58`	`drmessano`	`Then try again`
`19:24.01`	`tm1000`	`drmessano: lol`
`19:24.04`	`navaismo`	`Opera`
`19:24.14`	`drmessano`	`Wow`
`19:24.32`	`navaismo`	`ok working now outbound and inbound but no audio`
`19:24.48`	`drmessano`	`Did you add the transport line`
`19:25.13`	`navaismo`	`yes`
`19:25.17`	`drmessano`	`Show us`
`19:25.18`	`tm1000`	`navaismo: no audio or no ring`
`19:25.26`	`navaismo`	`that seems to fixed it drmessano`
`19:25.39`	`drmessano`	`So its working now`
`19:25.53`	`navaismo`	`tm1000, no audio, i can hear the ring but when answer only a nosie`
`19:26.14`	`navaismo`	`aaand the cli says its the codec`
`19:26.20`	`tm1000`	`navaismo: how many times did you try it now`
`19:26.26`	`tm1000`	`ok can you show use the message...`
`19:26.32`	`tm1000`	`us*`
`19:26.36`	`navaismo`	`<PROTECTED>`
`19:27.31`	`tm1000`	`why is it doing that`
`19:28.05`	`tm1000`	`navaismo: your server is changing the codec which is why you cant hear anything`
`19:28.45`	`navaismo`	`hmm`
`19:29.02`	`Katty`	`danny`
`19:29.10`	`p3nguin`	`devito`
`19:29.17`	`Katty`	`clever boy.`
`19:29.24`	`navaismo`	`and another problem wit my db peer [Nov 12 13:27:35] WARNING[10121][C-00000005]: frame.c:821 ast_parse_allow_disallow: Cannot disallow unknown format ''`
`19:29.25`	`navaismo`	`[Nov 12 13:27:35] WARNING[10121][C-00000005]: chan_sip.c:30141 build_peer: Codec configuration errors found in line 0 : disallow =`
`19:29.48`	`p3nguin`	`You can't disallow NOTHING.`
`19:29.50`	`navaismo`	`I upgrade the sip_buddies table to sipfriends table and messed`
`19:30.04`	`navaismo`	`changing to NULL`
`19:30.04`	`Katty`	`p3nguin: you can in alabama.`
`19:30.07`	`Katty`	`p3nguin: <3`
`19:30.12`	`p3nguin`	`haha`
`19:30.27`	`Katty`	`mister Qwell`
`19:30.33`	`Katty`	`pesters Qwell`
`19:31.04`	`tm1000`	`navaismo: changing what to null`
`19:32.21`	`navaismo`	`disallow on sip table`
`19:33.47`	`navaismo`	`aaah great, thanks tm1000, drmessano & file working now`
`19:35.00`	`tm1000`	`ohhh`
`19:35.07`	`tm1000`	`navaismo: this is realtime?`
`19:35.27`	`Qwell`	`Katty: WAT`
`19:35.28`	`file`	`I hope realtime never gets added to chan_motif because I will cry ;(`
`20:45.31`	`*** join/#asterisk infobot (~infobot@rikers.org)`
`20:45.31`	`*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.1 (2012/11/05), 10.10.0 (2012/11/06), 1.8.18.0 (2012/11/06), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org`
`20:46.14`	`SeRi`	`for what ever reason when I dial to my brother's office in PR they see my cid as 212345. He is using a local ITSP in PR. though calling cell lines and pstn lines in PR show my CID correct`
`20:50.42`	`*** join/#asterisk ruied (~ruied@po-217-129-155-146.netvisao.pt)`
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`21:48.30`	`ibercom`	`How to use Call Completion between servers ? Anybody know ?`
`21:49.31`	`p3nguin`	`What is this "Call Completion" phenomenon that you speak of?`
`21:50.33`	`WIMPy`	`Hasn't that been around since 1.8 already?`
`21:51.07`	`ibercom`	`Call Completion Supplementary Services (CCSS) or ring again ...`
`21:52.02`	`WIMPy`	`Or to the non Asterisk user better known as the two parts of it, CCBS and CCNR.`
`21:53.17`	`ibercom`	`Yes, I want use it between two servers ? Is it possible ?`
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`21:54.16`	`WIMPy`	`iax or sip or what?`
`21:55.48`	`ibercom`	`Now, IAX trunk. But if it is necessary SIP trunk.`
`21:56.05`	`p3nguin`	`SIP doesn't trunk.`
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`21:57.11`	`WIMPy`	`didn't care to invest in to that. Missing a real phone line it would have very limited use to me.`
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`22:00.32`	`ibercom`	`Where can I find info about this ?`
`22:00.45`	`WIMPy`	`On the wiki.`
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`22:03.36`	`ibercom`	`It doesn't talk about CCSS between servers. I need find something, a guide.`
`22:04.19`	`WIMPy`	`I'm not even sure it can be done on the channels tehemseves.`
`22:04.35`	`WIMPy`	`Maybe you could fake it via distributed device states.`
`22:05.37`	`artyx`	`I have this asterisk msg that extensions (changes depending on which cosmic ray is bouncing off of the hardware at that moment) become unreachable. At the same time this error occurs, sip updates to that phone for things like "Turn on ringing, turn off ringing, turn off blf, turn on blf" etc becoem "stuck"`
`22:06.00`	`artyx`	`I have done some ping tests, and some saturation tests on the ethernet, and it is not the ethernet adapter... it isn't the phone either`
`22:06.37`	`p3nguin`	`Extensions don't become unreachable, devices become unreachable.`
`22:06.45`	`ibercom`	`I need investigate more ...`
`22:07.03`	`artyx`	`Is that the extent of yoru contribution to the conversation p3 ?`
`22:07.15`	`AkkerKid`	`has anyone implemented the penalty timeouts in queues?`
`22:07.45`	`p3nguin`	`You say something silly, I educate you about the matter... yeah, that's pretty much the extent of it.`
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`22:08.00`	`artyx`	`Thanks for that, the problem disssappeared, it wasn't the extention after all.`
`22:08.07`	`artyx`	`Oh wait... they are still randomly goign unreachable.`
`22:08.16`	`WIMPy`	`artyx: If you want help in here, you need to learn the special terminology first :-(`
`22:08.20`	`p3nguin`	`Again with the silliness.`
`22:08.51`	`artyx`	`The Terminology isn't device, it's peer`
`22:09.07`	`artyx`	`So if you want to preach from a soapbox, get it right i would assume`
`22:09.39`	`artyx`	`(notice) chan_sip.c Peer XXXX is now UNREACHABLE!`
`22:09.59`	`p3nguin`	`I don't see anything about extensions.`
`22:10.11`	`artyx`	`And i dont see anything about device`
`22:10.55`	`navaismo`	`hmm someone need to read the book`
`22:10.59`	`navaismo`	`~book`
`22:10.59`	`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`
`22:11.14`	`artyx`	`I have this book, printed and bound But what i read is [2012-11-12 09:34:27] NOTICE[12203] chan_sip.c: Peer '2005' is now UNREACHABLE! Last qualify: 61`
`22:11.45`	`WIMPy`	`If we could read, we wouldn't be here.`
`22:11.49`	`artyx`	`That does not say the word extension (you are correct) or device (you are wrong) afa the log go's. Now yes, it (the peer) may be a device in this instance, but that could change`
`22:12.30`	`navaismo`	`check your peer/device`
`22:12.35`	`artyx`	`The fact is, it goes out, and comes back... WHEN this occurs the sip awk packets aren't making it back`
`22:12.46`	`artyx`	`navaismo: With a big black sharpy>? be more specific pls`
`22:13.22`	`artyx`	`It randomly breaks, getting rid of qualify fixes the constant nagging of peer is lagged, but does not fix the fact the host is not connected at that moment`
`22:14.42`	`navaismo`	`with the sip debug, with some network traces(ping tcpdump) & check the plug on the wall and in your device, i.e. old aastra phones looses the plug and you need to resolder it`
`22:15.39`	`artyx`	`These are brand new t-26p's. at the saem time the device is "unreachable" its responding to a ping, i've also done a ping -f and held a call. both locally through extension, through ip, and sip -> outside through pbx`
`22:16.23`	`navaismo`	`well yealink are not the best brand I used two news and back to warranty`
`22:16.28`	`p3nguin`	`Are you using iptables or some other method if connection tracking between the device and asterisk?`
`22:16.29`	`navaismo`	`try updating the firmware`
`22:16.33`	`artyx`	`The asterisk service is multi-nic atm, the internal nic has not had any packet loss or timeouts on packets on the`
`22:17.25`	`artyx`	`The firmware on the phone is newest gen. the ethernet ports go from asterisk server to switch (which ive replaced) to pulls to users, Ive also put a device onto the switch locally (without going through ceiling and jacks etc)`
`22:18.15`	`navaismo`	`did you tried with other phone brand or softphone? just to discard the device`
`22:18.30`	`artyx`	`softphone yes, ive tried, (part of my debug to my extension)`
`22:19.05`	`artyx`	`I am wondering if this version of linux has an issue im not aware of yet`
`22:19.19`	`artyx`	`but the version is required courtesy of asix drivers`
`22:19.48`	`navaismo`	`with the sofphone you see the same problem?`
`22:20.14`	`artyx`	`I cannot replicate the blf error with softphone`
`22:20.24`	`p3nguin`	`The qualify packets are just SIP OPTIONS packets (which are application level), where pings are just checking the networking and don't care about SIP at all.`
`22:20.33`	`navaismo`	`what OS do you use?`
`22:20.41`	`artyx`	`for PBX or softphone client`
`22:20.46`	`navaismo`	`PBX`
`22:21.23`	`WIMPy`	`points at te good old wireshark`
`22:21.26`	`artyx`	`Linux SESHAT 2.6.32-279.11.1.el6.i686 #1 SMP Tue Oct 16 14:40:53 UTC 2012 i686 i686 i386 GNU/Linux`
`22:22.00`	`WIMPy`	`Not a nice version number.`
`22:22.09`	`Chainsaw`	`Linux phoenix 3.7.0-rc4-00170-gb251f0f #1 SMP PREEMPT Sat Nov 10 22:29:01 GMT 2012 x86_64 Six-Core AMD Opteron(tm) Processor 2435 AuthenticAMD GNU/Linux`
`22:22.10`	`artyx`	`My 2.4 base system did not experience this issue, nor did it work with asix`
`22:22.16`	`Chainsaw`	`Outdated doesn't quite describe it.`
`22:22.29`	`artyx`	`This is the newest supported kernel for this distribution`
`22:22.47`	`Chainsaw`	`RedHat Expensive Linux or something?`
`22:22.48`	`p3nguin`	`Linux cpe-e650 2.6.32-lts #1 SMP Mon Mar 28 17:08:32 UTC 2011 i686 VIA Nehemiah CentaurHauls GNU/Linux`
`22:22.54`	`artyx`	`Yeh. Cent actually Chainsaw`
`22:22.59`	`p3nguin`	`I, too, run 2.6.32.`
`22:23.01`	`Chainsaw`	`shakes head in disbelief`
`22:23.05`	`artyx`	`and lts is ubuntu`
`22:23.08`	`artyx`	`i would assume`
`22:23.38`	`artyx`	`I can also disregard the possibility its a firewall config, as ive stopped all iptables`
`22:24.07`	`WIMPy`	`I thought it was a thing of the 90s, but as I had to learn (in here actually) you still don't want to use distribution kernels.`
`22:24.26`	`artyx`	`You mean use custom compiled kernels?`
`22:24.43`	`Qwell`	`WIMPy: Says who?`
`22:24.47`	`WIMPy`	`No the ones with unofficial patches.`
`22:25.03`	`artyx`	`I can... I have, i did, i will ... And i use the centos patches as well, since its a centos system`
`22:25.21`	`p3nguin`	`I see absolutely no problem using distribution kernels, as long as they are vanilla.`
`22:25.38`	`artyx`	`Mostly i just turn off module support for things i dont need. and i embed drivers i do need`
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`22:26.25`	`artyx`	`so im 90% confident the switching infrastructure is good`
`22:26.45`	`artyx`	`The "devices" did not fail me in the previous installation, using a different major rls on the os`
`22:27.36`	`artyx`	`I imported the etension config, and the tftp/provision config from the old system, but rebuilt the sip trunk, dahdi, DIDs, etc`
`22:28.13`	`wwalker`	`OT: looking for recommendations for a client. The client is an emergency notification group and want some more VOIP providers (currently have 2). They are looking for a reliable provider that can handle them spinning up 500 ports at a moment's notice.`
`22:28.39`	`wwalker`	`please PM me with any providers you have dealt with and would recommend.`
`22:31.06`	`Chainsaw`	`wwalker: Knowing what part of the world you're in would help narrow things down.`
`22:32.08`	`Chainsaw`	`artyx: Anyhow, is this a Polycom with UC firmware against Asterisk 10?`
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`22:33.05`	`artyx`	`Chainsaw: Yealink T-26P against asterisk 10.9`
`22:33.53`	`Chainsaw`	`artyx: Hrm, not familiar with those. If it can do SIP over TCP, that always helps. Asterisk will notice earlier.`
`22:34.03`	`wwalker`	`DOH. Servers in US and Canada, prefer to make calls over the internet rather than having to colo at the provider's location, but that's not a show stopper. Calls to US48, Canada, HA, and AL`
`22:34.11`	`wwalker`	`Thanks Chainsaw`
`22:34.30`	`Chainsaw`	`wwalker: I think there's a US ITSP list in the bot. p3nguin might be able to unlock it.`
`22:34.43`	`artyx`	`Yes they support tcp`
`22:35.16`	`WIMPy`	`~itsplist-us`
`22:35.16`	`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`
`22:35.21`	`WIMPy`	`~itsplist-ca`
`22:35.21`	`infobot`	`somebody said itsplist-ca was Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca`
`22:35.49`	`p3nguin`	`Which providers are you already using?`
`22:37.02`	`wwalker`	`Excel and Airespring.`
`22:37.16`	`wwalker`	`Thanks for those lists.`
`22:37.46`	`wwalker`	`both pricey, but when we made the calls, the ports were there.`
`22:38.16`	`wwalker`	`and with Sandy's little tour of the northeast, we got to test that a lot just now`
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`22:38.59`	`wwalker`	`fair disclaimer, I worked at Excel years ago`
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`23:03.05`	`artyx`	`2003 or 2005`
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`23:12.28`	`JustinAiken`	`On Asterisk 1.8, we use SET CDR to set a custom field (filename)… it works great with dial, but if a Bridge is made (after the SET), the field is not written`
`23:13.14`	`JustinAiken`	`Is there a way to make it just fill in the field?`
`23:13.38`	`loggiew`	`if hostport = df7jal23ls0d.invalid then what value should it default to when responding? STANDARD_SIP_PORT?`
`23:14.05`	`loggiew`	`but then its expecting hostport to be an ip or dns`
`23:14.18`	`loggiew`	`so STANDARD_SIP_PORT isn't right`
`23:15.08`	`EmleyMoor`	`has solved his Asterisk crash :-)`
`23:15.28`	`loggiew`	`nice what version EmleyMoor`
`23:17.04`	`EmleyMoor`	`1.8.13-1~dfsg-1 - added a patch I found on the matter at issues.asterisk.org`
`23:17.18`	`loggiew`	`awesome`
`23:17.43`	`Sanon`	`simplest way to add google voice please`
`23:17.45`	`EmleyMoor`	`Have let the Debian bugtracker know too`
`23:21.38`	`mjordan`	`EmleyMoor: thanks for looking up the issue on the bug tracker. It should be fixed in versions 1.8.18.0 and 10.10.0.`
`23:23.09`	`mjordan`	`JustinAiken: Set(CDR) will set the CDR field on whatever channel you put it on, regardless of how it gets bridged (either implicitly via Dial or explicitly via Bridge). What is your dialplan where it doesn't show up, and on what channel(s)?`
`23:23.50`	`EmleyMoor`	`mjordan: My partner actually tried to look for the window in his Jabber client earlier when he got a call (before I'd fixed it)... but from now on it will work`
`23:26.54`	`*** join/#asterisk tapout (~tapout@unaffiliated/tapout)`
`23:27.01`	`EmleyMoor`	`also discovered he hadn't whitelisted his own mobile phone`
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`23:32.02`	`loggiew`	`file: dude you're right, chan_sip.c is nuts`
`23:32.15`	`loggiew`	`gonna take me a while to keep digging`
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`23:38.20`	`Sanon`	`simplest way to add google voice please`
`23:39.51`	`mjordan`	`Sanon: have you looked at the wiki?`
`23:40.10`	`Sanon`	`I did...`
`23:40.13`	`mjordan`	`loggiew: we feel you`
`23:40.50`	`mjordan`	`Sanon: which wiki page did you look at?`
`23:40.57`	`[TK]D-Fender`	`I hope not ... there are minors in here!`
`23:41.09`	`mjordan`	`[TK]D-Fender: feel your pain :-P`
`23:41.43`	`Sanon`	`https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google`
`23:42.10`	`loggiew`	`haha`
`23:42.17`	`mjordan`	`Sanon: great! So what do you have a question on?`
`23:43.50`	`Sanon`	`I know this is simple. i just don't want to mess this up. when i edit the files, do i replace the text already in there or just add the text and in wiki and modify`
`23:44.12`	`mjordan`	`Sanon: by files, do you mean your configuration files?`
`23:45.05`	`mjordan`	`Sanon: if you're using Asterisk 11 (which you should be, since that page is for Asterisk 11 - an older page exists for versions prior to that), you may need to create some of the configuration files, since the Motif channel driver is new`
`23:45.54`	`p3nguin`	`If there are instructions telling you what needs to be in the config files, that's what you need to put in the config files. If other things are already in the files, remove or comment out those things.`
`23:46.02`	`mjordan`	`p3nguin: depends`
`23:46.15`	`mjordan`	`p3nguin: the first step is to modify rtp.conf`
`23:46.22`	`mjordan`	`p3nguin: I wouldn't delete everything out of there :-)`
`23:46.31`	`loggiew`	`Sanon: make a copy of everything first, no worries`
`23:46.52`	`p3nguin`	`If the instructions aren't clear, someone needs to fix them.`
`23:46.59`	`Sanon`	`thanks again`
`23:47.24`	`mjordan`	`p3nguin: it says to change a particular setting, so I think its clear if you're familiar at all with Asterisk configuration`
`23:48.05`	`mjordan`	`p3nguin: but I agree, if there's a particular section that is unclear, it should be updated`
`23:51.57`	`loggiew`	`as one learning this, I think the documentation is good but there is a certain level of additional information that would make things more clear`
`23:52.03`	`loggiew`	`after I figure it all out Ill add`
`23:56.17`	`mjordan`	`loggiew: that would be much appreciated`
`23:58.07`	`*** part/#asterisk JustinAiken (~JustinAik@justinaiken.com)`

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