IRC log for #asterisk on 20090423

`00:00.45`	`aaroneous`	`I too am faced with the same depressing array of options, tho..`
`00:00.50`	`KyleK`	`I need a single FXO so I can replace my USR voice modem as an answering machine, its been corrupting data which causes messages to get cut off`
`00:00.58`	`aaroneous`	`Linkydink and Grand Suck`
`00:01.14`	`drmessano`	`Linksys ATA's work fine`
`00:01.22`	`joobie`	`btw TK.. i've been playing with AEL for all this queuing.. is the extensions.ael basically a new sort of dialplan formatting used for macros?`
`00:01.42`	`aaroneous`	`drmessano: must be one of those few anomalous products..`
`00:01.43`	`[TK]D-Fender`	`joobie: No`
`00:01.45`	`joobie`	`so we should still use extensions.conf and sorta integrate it with AEL macros in extensions.ael? Is that the way * is heading with it?`
`00:02.45`	`joobie`	`TK, how should it be used? I just got my head around extensions.conf .. then extensions.ael was a whole new can of worms - not really clear now of the boundry between the two`
`00:02.55`	`joobie`	`or why * has introduces ael`
`00:02.56`	`[TK]D-Fender`	`joobie: AEL gets parsed back to the same extensions.conf logic as we already use. It doesn't do anything "more"`
`00:03.24`	`aaroneous`	`which Linkydink FXS is my best bet for single-port T.38 support? (not that I really need T.38 in a QoS-enabled LAN like this, but it would be nice to have anyway)`
`00:03.41`	`KyleK`	`I wonder if the linksys hardware would work as answering machine, just needs to be able to pick phone up after X rings (instead of immediately) and not react stupidly if someone picks up the line before X rings`
`00:03.52`	`joobie`	`TK, but why use AEL's to extend the logic used in extensions.conf, when you can use extensions.conf macros ?`
`00:03.52`	`aaroneous`	`googling gave me a lot of conflicting answers regarding which linksys/sippura boxes could do T.38`
`00:03.58`	`[TK]D-Fender`	`joobie: AEL was created so people who want something that looks like another language they use instead of just complaining about it`
`00:04.10`	`[TK]D-Fender`	`joobie: it does not "exten" anything`
`00:04.13`	`[TK]D-Fender`	`extend*`
`00:04.27`	`joobie`	`lol`
`00:04.32`	`joobie`	`ahh`
`00:05.12`	`joobie`	`so are asterisk planning to keep both the extensions.conf and extensions.ael to run side by side? because right now im using extensions.conf for basic stuff and sorta putting more larger macros / functions in extensions.ael.. so im sorta using both`
`00:05.49`	`joobie`	`it's not bad tho I reacon.. AEL that is`
`00:05.59`	`joobie`	`there are more functions you can use in there, making the dialplan more flexible`
`00:08.41`	`[TK]D-Fender`	`joobie: No. AEL gets parsed back to extensions.conf logic. it does precisely NOTHING more than standard dialplan logic as its parsed. Its attempt to "structure" the logic in fact RESTRICTS what you can do.`
`00:09.40`	`pfn`	`it'd be nice if dialplan steps could be all 'n' instead of needing to start with 1`
`00:09.55`	`joobie`	`oh`
`00:10.23`	`joobie`	`i didnt think extensions.conf could do switch()`
`00:11.14`	`joobie`	`hmm`
`00:11.24`	`joobie`	`can the standard dialplan use switch?`
`00:11.32`	`joobie`	`google didn't bring up anything for me.....`
`00:12.33`	`[TK]D-Fender`	`joobie: You're thinking too hard, or not at all.`
`00:13.02`	`*** join/#asterisk stoked (n=df@S01060016b62857a6.vc.shawcable.net)`
`00:13.05`	`joobie`	`TK, with AEL.. i can take a variable and run it through the switch() function..`
`00:13.12`	`joobie`	`can that be done with standard dialplans?`
`00:13.45`	`joobie`	`I just havent come across that sorta stuff in standard extensions.conf.. that's why i thought AEL is a bit more flexible in that it has more support for additional functions`
`00:13.50`	`stoked`	`just got my # ported over to my voip provider, but I didn't realize that for some reason the # in the CID doesn't seem to come through properly when going through my asterisk server`
`00:15.19`	`[TK]D-Fender`	`joobie: joobie nothing you can't do in stadard dialplan.`
`00:15.42`	`joobie`	`TK, what's the function to do switch in standard dialplan?`
`00:15.52`	`joobie`	`i mean i know it can be done with if and ifelse.. but that's gay`
`00:15.58`	`[TK]D-Fender`	`joobie: go lok what it gets parsed into.`
`00:16.19`	`joobie`	`i dont follow....`
`00:16.29`	`joobie`	`maybe i am thinking too hard.... or too little :P`
`00:16.39`	`[TK]D-Fender`	`joobie: "dialplan show" <- Wake up. Really.`
`00:16.48`	`joobie`	`ahh`
`00:17.12`	`joobie`	`i didnt know that cmd:P`
`00:17.13`	`joobie`	`thanks`
`00:17.23`	`joobie`	`smaks [TK]D-Fender around with a wet trout`
`00:18.01`	`[TK]D-Fender`	`~cluebat joobie`
`00:18.01`	`infobot`	`ACTION pulls out a ClueBat (tm) and thwaps joobie.`
`00:18.50`	`joobie`	`wow`
`00:19.05`	`joobie`	`i suddenly feel an overflow of enlightenment`
`00:20.18`	`columbo`	`?`
`00:20.32`	`joobie`	`ahh`
`00:20.54`	`joobie`	`i see what you mean TK`
`00:21.07`	`joobie`	`the switch() is much simpler to understand though coming from a perl background`
`00:21.13`	`[TK]D-Fender`	`joobie: All it does is get baked backwards.`
`00:21.14`	`stoked`	`can anyone point me in the right direction for callerid?`
`00:21.16`	`joobie`	`yeaa`
`00:21.23`	`joobie`	`i guess it depends on your background`
`00:21.57`	`[TK]D-Fender`	`joobie: Yeah, mine is "OMFG it doesn't do more and definitely does less and fewer people to help me with it and more bugs to deal with? FUCK IT"`
`00:22.07`	`joobie`	`lol`
`00:22.10`	`joobie`	`yeaa`
`00:22.17`	`joobie`	`but are * pushing down the AEL path more?`
`00:22.26`	`[TK]D-Fender`	`joobie: No, and you CAN'T`
`00:22.28`	`*** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)`
`00:22.38`	`[TK]D-Fender`	`joobie: It just gets compiled back. It'll always have more problems.`
`00:22.46`	`joobie`	`yea`
`00:23.18`	`[TK]D-Fender`	`joobie: it isn't an "alternative". All you're doing is putting the pet in a stupid skirt and asking it to dance.`
`00:23.19`	`joobie`	`they could eventually write calls to these lower level functions that extensions.conf is using, direct from AEL`
`00:23.26`	`[TK]D-Fender`	`joobie: Nope.`
`00:23.38`	`joobie`	`they can`
`00:23.47`	`[TK]D-Fender`	`joobie: because all apps are tied to the concept of context, extension, priority`
`00:23.51`	`[TK]D-Fender`	`^^^^^^`
`00:23.55`	`[TK]D-Fender`	`joobie: TIED TO IT`
`00:24.13`	`[TK]D-Fender`	`joobie: Gotoif <- WTF are you going to "goto"?`
`00:24.31`	`[TK]D-Fender`	`joobie: Its just a wrapper. Get over it`
`00:24.42`	`joobie`	`TK, think of it how i mentioned above when i was off target`
`00:24.50`	`joobie`	`extensions.conf being "basic" manipulation`
`00:24.57`	`joobie`	`and extensions.ael being more complex`
`00:25.10`	`joobie`	`you can have placeholders for functions in extensions.conf`
`00:25.13`	`joobie`	`like for example...`
`00:25.25`	`joobie`	`answer(); queue(); hangup();`
`00:25.29`	`joobie`	`queue can reference login in AEL`
`00:25.41`	`[TK]D-Fender`	`joobie: Actually the reverse. Extensions.conf is capable of more, making AEL a "simple" or "dumbed down for the masses"`
`00:25.45`	`joobie`	`so the basic functionality exists in extensions.conf.. and more complex manipulation is offset to the AEL`
`00:25.58`	`joobie`	`it still returns a result.. but all lower level queue manipulation can be done within AEL`
`00:26.05`	`joobie`	`yea`
`00:26.09`	`joobie`	`that's how it is as of today`
`00:26.15`	`[TK]D-Fender`	`joobie: What part of "AEL doesn't add 1 ^%#$ING FUNCTION to *" don't you get?`
`00:26.16`	`joobie`	`duno`
`00:26.26`	`joobie`	`TK`
`00:26.31`	`joobie`	`my point is, it is possible to do that`
`00:26.57`	`joobie`	`say the AddQueueMember function is made defunt in newer versions`
`00:27.05`	`joobie`	`and the way to add a queue member is to use functions in AEL`
`00:27.20`	`[TK]D-Fender`	`joobie: by "possible" you must mean "with massively rewriting an * execution engine" yes. But thats like saying you can turn a car into a plan by completely rebuilding it with airplane parts`
`00:27.23`	`joobie`	`the queue() function can still exist in extensions.conf.. but all the lower logic is offset to AEL`
`00:27.36`	`[TK]D-Fender`	`[20:27]<joobie>and the way to add a queue member is to use functions in AEL <- no.`
`00:27.52`	`[TK]D-Fender`	`joobie: Holy shit, its just call's *&#^$ING dialplan apps! AEL doesn't do shit!`
`00:28.08`	`joobie`	`relax TK`
`00:28.15`	`[TK]D-Fender`	`joobie: Are there any active brain-cells up there at this point?`
`00:28.21`	`joobie`	`they are moving around`
`00:28.23`	`joobie`	`i can feel them`
`00:28.28`	`joobie`	`yours are bouncing around like crazy`
`00:28.32`	`joobie`	`relax`
`00:28.35`	`[TK]D-Fender`	`joobie: It gets ocompiled back. You use AQM to add a device, you have to use RQM to remove it.`
`00:28.39`	`[TK]D-Fender`	`joobie: Its that simple`
`00:28.59`	`[TK]D-Fender`	`joobie: They gave you the tools, they make it so the name itself is blatantly obvious.`
`00:29.00`	`joobie`	`i hear ya`
`00:29.15`	`[TK]D-Fender`	`joobie: Yet you seem to have problems understanding this.`
`00:29.22`	`joobie`	`no`
`00:29.25`	`joobie`	`i completely understand how it works`
`00:29.40`	`joobie`	`what im saying is just ideas of how AEL can be used with extensions.conf standard diaplan in the future`
`00:29.53`	`[TK]D-Fender`	`joobie: you seem to think "AEL" has some miracle function, or this is any part of how you're "intended" to do to.`
`00:29.58`	`[TK]D-Fender`	`it*`
`00:29.59`	`joobie`	`because the problem with extensions.conf standard dialplan today, is that the formatting of the syntax is WAY different to any other coding / scripting language`
`00:30.15`	`joobie`	`mainstream languages are very similar.. switch() for exmaple`
`00:30.28`	`denon`	`joobie: that's because you're scripting a pbx, not a website`
`00:30.35`	`joobie`	`* goes out on a limb.. which is good to keep it niche, but bad as it makes it harder to learn`
`00:30.45`	`joobie`	`denon, that's rubbish`
`00:30.58`	`joobie`	`i can write a .net applicatino and use functions i use in php for a website`
`00:31.03`	`denon`	`extensions.conf syntax is very simple to learn and understand`
`00:31.10`	`denon`	`AEL tends to be more complex, needlessly`
`00:31.13`	`joobie`	`just because it's a different app doesnt mean it doesn't need to be tottally different to drive`
`00:31.13`	`[TK]D-Fender`	`joobie: Yes, well I'm glad AEL makes you feel better about your lack of adaptability, but don't think for a second that it has anything to do with "functionality".`
`00:31.16`	`denon`	`you can use either`
`00:31.24`	`joobie`	`denon, i disagree`
`00:31.49`	`joobie`	`i did a switch with AEL.. and to me, it looks alot simpler to the code that standard extensions.conf set out for the switch`
`00:31.54`	`denon`	`joobie: the management interfaces lets you do whatever you want .. you could do everything from a .net app based on events if you want`
`00:32.08`	`denon`	`joobie: good for you, that's why we offer AEL`
`00:32.48`	`denon`	`in the end, though, AEL is simply giving a different face to the same underlying functions`
`00:32.59`	`denon`	`most of us prefer extensions.conf for it's simplicity and directness`
`00:33.02`	`[TK]D-Fender`	`denon: I'm gray on that... not sure AGI (and varients) give you all the dialplan app functionality as several are really geared towards being used directly within, etc. Also hanguphandling, etc....`
`00:33.20`	`[TK]D-Fender`	`denon: As opposed to being executed by calls....`
`00:33.34`	`[TK]D-Fender`	`denon: Or is the another kind of interface whose value I've overlooked?`
`00:33.35`	`denon`	`[TK]D-Fender: well .. you could .. it'd be ugly, but I'm saying any more complex logic, if that's the beef`
`00:33.51`	`[TK]D-Fender`	`denon: Oh yeah.. for complex stuf... you don't want to BE in dialplan :)`
`00:34.05`	`denon`	`[TK]D-Fender: well .. I do .. dialplan's fast :)`
`00:34.15`	`[TK]D-Fender`	`denon: The biggest problem with AEL is.... extensions.conf <-`
`00:34.27`	`joobie`	`agreed`
`00:34.32`	`joobie`	`as it has to feed back, which sucks`
`00:34.37`	`[TK]D-Fender`	`denon: Being fast & going nowhere... is going nowhere mighty fast :)`
`00:34.56`	`denon`	`joobie: everything has to "feed" somewhere`
`00:35.01`	`joobie`	`but in a magical land of gumdrops and sugarcains, there could be a transition to support AEL direct to the code, without having to feed back to extensions.conf in the way it does today`
`00:35.02`	`[TK]D-Fender`	`* needs typed variables, proper escaping, and RegEx`
`00:35.21`	`joobie`	`nod, but standard extensions.conf feeds back to the src`
`00:35.26`	`joobie`	`AEL could do the same`
`00:35.32`	`denon`	`joobie: compile AEL direct to machine code? it's a configuration, not a program`
`00:35.33`	`joobie`	`rather than feeding back to extensions.conf standard dialplan`
`00:35.41`	`[TK]D-Fender`	`joobie: * would need a completely separate kind of parser for this, and debugging to match.`
`00:35.47`	`joobie`	`not to machine code`
`00:35.56`	`[TK]D-Fender`	`joobie: Once again you'll have to rewrite a ton of stuff.`
`00:36.00`	`joobie`	`to the C code that the standard extensions.conf ties to`
`00:36.03`	`joobie`	`TK, agreed`
`00:36.09`	`denon`	`extensions.conf/ael/whatever .. is a configuration file..`
`00:36.22`	`joobie`	`but the upshot of it all is the configuration of dialplans are more standardized with a hell of a lot of other languages`
`00:36.27`	`joobie`	`which makes asterisk easier to configure for the masses`
`00:36.35`	`[TK]D-Fender`	`joobie: Including the dialplan apps themselves. You can still see how AEL can't look quite like another language because of the concept of "extensions"`
`00:36.51`	`joobie`	`im not saying extensions.conf is not beyond someone understanding.. it's just harder, coming from a general coding/scripting background`
`00:36.56`	`denon`	`joobie: if you look in the source, you'll see that the extensions.conf functions look an awful lot like the C functions already`
`00:37.09`	`[TK]D-Fender`	`joobie: Yes, especially harder when people don't read the instructions <-`
`00:38.14`	`joobie`	`TK.. the instructions suck`
`00:38.15`	`[TK]D-Fender`	`joobie: the only things AEL changes are the general removal of priorities, repeating the exten pattern at the start and giving you the illusion of a few more basic structured programming syntaxes from other languages`
`00:38.33`	`[TK]D-Fender`	`joobie: The instructions are almost completely obvious.`
`00:38.38`	`joobie`	`not for queues`
`00:38.47`	`joobie`	`i found leads to defunt functions in standard dialplans`
`00:38.52`	`joobie`	`and references saying use AEL`
`00:38.59`	`denon`	`this is such a cyclic channel`
`00:39.04`	`[TK]D-Fender`	`joobie: Read the apps and stop praying there's a way around everything. There is a hammer in front of you. Its made to hammer a nail, stop trying to use it as a screwdriver`
`00:39.04`	`joobie`	`heck the documentation on how to do queue management that came with 1.4 was all based on AEL`
`00:39.23`	`joobie`	`.. just a thought`
`00:39.24`	`joobie`	`:)`
`00:39.34`	`[TK]D-Fender`	`joobie: What doc are you referring to exactly?`
`00:39.35`	`joobie`	`that said, standard dialplan rocks`
`00:39.39`	`denon`	`joobie: you've got the source .. submit a patch.`
`00:39.41`	`joobie`	`voip-info`
`00:39.52`	`[TK]D-Fender`	`joobie: "Came with 1.4" my ass`
`00:40.06`	`[TK]D-Fender`	`joobie: WIKI is unmaintained crap contributed by "who knows"`
`00:40.13`	`joobie`	`asterisk-1.4.22.1/doc/queues-with-callback-members.txt`
`00:40.16`	`joobie`	`.that is all based on AEL`
`00:40.17`	`jaytee`	`hey! I know that guy!`
`00:40.27`	`joobie`	`i didn't want to use AEL.. but heck, the documentation forces you down that route`
`00:40.47`	`joobie`	`it's not queues-with-callback-members-aelstyle.. it's just generic, queues-with-callback-members`
`00:41.01`	`[TK]D-Fender`	`joobie: "forces"? Get real`
`00:41.06`	`joobie`	`hey jay :)`
`00:41.28`	`joobie`	`TK, ok forces is not the best selection of words :P`
`00:41.37`	`joobie`	`"steers" is more relevant`
`00:41.48`	`AsteriskDom`	`infobot: AEL`
`00:42.12`	`infobot`	`ael is probably Asterisk Extension Language - a dialplan language with 'c like' syntax?`
`00:42.12`	`[TK]D-Fender`	`joobie: Yes, and the first sentence APOLOGIZES for showing you a sample with AEL`
`00:42.12`	`joobie`	`lol :)`
`00:42.22`	`joobie`	`it does :)`
`00:43.34`	`[TK]D-Fender`	`joobie: And that sample has the very specific code showing you to RemoveQueueMemeber. Yup... you looked REALLY hard :)`
`00:44.24`	`joobie`	`TK, I used RemoveQueueMember to get rid of that Local/1000 queue member.. BUT, was just hoping that to remove a stale queue member, there would be a CLI command`
`00:44.29`	`[TK]D-Fender`	`joobie: This is a really bloated sample...`
`00:44.41`	`joobie`	`it seems limiting that you have to write a dialplan function to cleanup a stale member`
`00:44.45`	`joobie`	`it is`
`00:44.54`	`joobie`	`mine is based on it, but very much cut down..`
`00:45.04`	`joobie`	`there's a lot of crap in that one..`
`00:45.17`	`joobie`	`but as a queue noob, it gives you the full picture which is good`
`00:45.27`	`[TK]D-Fender`	`joobie: I also love how the code sample for the "pre-acknowledgement" is in STANDARD extensions.conf. Consistency = 0`
`00:45.41`	`joobie`	`hehe :)`
`00:45.56`	`[TK]D-Fender`	`and some broken/worthless logic`
`00:46.02`	`joobie`	`where is your love for AEL TK.. where is the love..`
`00:46.22`	`[TK]D-Fender`	`actually., ALL of the GotoIf's in the sample are BROKEN <-`
`00:46.56`	`*** join/#asterisk Tuxguy (n=jimi@cpe-065-184-197-243.ec.res.rr.com)`
`00:46.57`	`joobie`	`his queue-addremove() function is pretty creative though`
`00:47.09`	`joobie`	`one call for two funtions`
`00:47.20`	`[TK]D-Fender`	`joobie: All AEL does is bring more people in here who find bugs with it or can't figure it out and expect it to do things it can't. It does not add to *'s functionality`
`00:47.24`	`joobie`	`well 5 functions even`
`00:47.40`	`joobie`	`yea`
`00:47.41`	`joobie`	`i hear ya`
`00:47.50`	`[TK]D-Fender`	`joobie: What do users really want? They want SIP-B, better codec support, reduced overhead, video conferencing, and so on`
`00:48.02`	`[TK]D-Fender`	`joobie: AEL is a petty distraction by comparison`
`00:48.09`	`joobie`	`true`
`00:48.45`	`[TK]D-Fender`	`joobie: Yes, and their function is just the same worded as [macro-zomgitsamiracle]`
`00:49.07`	`joobie`	`neway i have to wrap up this queue.. speaking of distractions cough #asterisk cough :P ..`
`00:49.18`	`joobie`	`yer.. i get your point TK:)`
`00:49.21`	`joobie`	`peace`
`00:49.49`	`[TK]D-Fender`	`joobie: Yes... no from where we started... go RQM them and move on :p`
`00:53.18`	`thedonvaughn`	`My switch just crashed at work with: [Apr 22 19:37:35] WARNING[23844] asterisk.c: Accept returned -1: Too many open files`
`00:53.27`	`thedonvaughn`	`should i be raising ulimit?`
`00:54.46`	`Tuxguy`	`Anyone get this error before? configure: error: C preprocessor "/lib/cpp" fails sanity check .. cpp gcc etc is installed`
`01:04.19`	`stoked`	`anyone know how to passthrough the callerid # from the provider to an extension?`
`01:05.50`	`*** join/#asterisk umpc (n=Justin@unaffiliated/umpc)`
`01:05.51`	`stoked`	`I keep getting the extension name as the #`
`01:09.28`	`[TK]D-Fender`	`stoked: pastebin is your friend. SHOW US`
`01:09.30`	`[TK]D-Fender`	`~pb`
`01:09.30`	`infobot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/`
`01:09.32`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^`
`01:10.04`	`*** join/#asterisk grx0 (n=root@70.94.220.114)`
`01:11.45`	`Hazuki`	`wowwwwww, VoIP gateways are expensive`
`01:11.53`	`Hazuki`	`Mediatrix's 4116 SIP hub is $1100 ><`
`01:12.01`	`Hazuki`	`the boss is going to poo masonry over this`
`01:12.50`	`stoked`	`http://pastebin.com/m30b0b460`
`01:13.55`	`stoked`	`was there something output you were asking for?`
`01:13.56`	`*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a5c458dbf409ac8c)`
`01:14.33`	`stoked`	`I setup the mbu400 to connect directly to my provider, and it seems to get the CID # properly`
`01:14.43`	`stoked`	`so I must have messed up something somewhere in my asterisk config`
`01:14.52`	`[TK]D-Fender`	`Hazuki: Linksys SPA-8000 + SPA-3102 = 9FXS, 1 FX0 ~ $310 USD`
`01:15.41`	`Hazuki`	`we need one FXO and no FXS`
`01:15.42`	`[TK]D-Fender`	`stoked: Where do I see you NoOp-ing the callerid? Wheres the actual code in addition to the CLI output?`
`01:16.05`	`[TK]D-Fender`	`Hazuki: If you don't need any FXS, then WTF is that Mediatrix for?`
`01:16.32`	`Hazuki`	`[TK]D-Fender: to connect 10 phones from inside the office`
`01:16.50`	`Hazuki`	`one FXO for the business phone line, and the Mediatrix to multiplex it through * to 10 phones`
`01:16.52`	`[TK]D-Fender`	`Hazuki: Wakeup time.. thats what FXS IS <-`
`01:17.02`	`[TK]D-Fender`	`~fxs`
`01:17.03`	`infobot`	`extra, extra, read all about it, fxs is foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx`
`01:17.09`	`stoked`	`[TK]D-Fender in my extension devices, the number shows up as the name, but the extension name shows up as the number`
`01:17.10`	`[TK]D-Fender`	`Hazuki: Analog phone interface <-`
`01:17.32`	`[TK]D-Fender`	`stoked: NoOp it in your dialplan, and pastebin the code & the result`
`01:17.33`	`Hazuki`	`hm, so if I were to get a 1 FX0 10 FXS device, would that handle everything?`
`01:17.43`	`Hazuki`	`the single line in, and the 10 lines out?`
`01:17.44`	`[TK]D-Fender`	`Hazuki: Were there such a thing`
`01:17.56`	`stoked`	`sorry... I'm a asterisk newb, how do I do that`
`01:18.22`	`[TK]D-Fender`	`Hazuki: It'll take 2-3 device for anything reasonable to come to the combination you're looking for at anything resembling "cost competitive" like what I suggested at > 1/3 the cost`
`01:18.27`	`grx0`	`can't believe hazuki is still trying to use any POTS....`
`01:18.36`	`Hazuki`	`grx0, this is my boss's decision`
`01:18.41`	`Hazuki`	`he has a certain use case he wants`
`01:18.55`	`[TK]D-Fender`	`stoked: NoOp(name = "${CALLERID(name)}" number = "${CALLERID(num}")`
`01:18.58`	`Hazuki`	`I'm the new woman in the office, his first tech type...past a certain point I can't say "we shouldn't do this"`
`01:19.01`	`stoked`	`ok`
`01:20.27`	`[TK]D-Fender`	`Hazuki: Do you already have a good analog patch-panel infrastructure for your office to plug this all into?`
`01:20.57`	`Hazuki`	`He wants to do this with a single server (Vostro 200), a 1-FXO 0 FXS card, and a VoIP hub`
`01:21.16`	`Hazuki`	`he's going to have to run the phone lines from each desk down to the basement in a drop, like he did with the ethernet cables`
`01:21.58`	`[TK]D-Fender`	`Hazuki: 1st tip : ditch the term "voip hub".`
`01:22.07`	`Hazuki`	`I know it's ignorant, I`
`01:22.09`	`Hazuki`	`'m sorry ><`
`01:22.20`	`Hazuki`	`this has been a long day, he doesn't know entirely what he's getting into, this is new to me...`
`01:22.31`	`[TK]D-Fender`	`Hazuki: He wants 10 analog phones and 1 analog line. He doesn't give a rats ass about the kind of equipment that takes those in`
`01:22.58`	`Hazuki`	`[TK]D-Fender, is there a simpler, cheaper method than a Digium FXO card and this thing from Mediatrix?`
`01:23.12`	`[TK]D-Fender`	`Hazuki: So its a question of what will do the job with good quality & some semblance of cost-effectiveness.`
`01:23.18`	`Hazuki`	`Yes`
`01:23.23`	`[TK]D-Fender`	`Hazuki: Frankly having only 1 analog line is a disaster.`
`01:23.27`	`grx0`	`AMEN`
`01:23.29`	`grx0`	`:)`
`01:23.44`	`Hazuki`	`That part isn't up to me`
`01:23.49`	`[TK]D-Fender`	`Hazuki: Means 1 guy on the line and you don't get more callers in. No VM, no outbound while anyone is talking, etc`
`01:23.53`	`stoked`	`http://pastebin.com/d5cb96a20`
`01:24.15`	`stoked`	`[TK]D-Fender seems like name/number is shown correctly`
`01:24.25`	`Hazuki`	`what do you suggest he does then? I thought Asterisk was supposed to handle the heavy lifting about multiplexing all this`
`01:24.41`	`[TK]D-Fender`	`stoked: Looks like your provider doesn't send the name`
`01:24.46`	`[TK]D-Fender`	`stoked: So not *'s fault`
`01:24.58`	`stoked`	`[TK]D-Fender no I don't need the name`
`01:25.04`	`stoked`	`I need the number`
`01:25.28`	`[TK]D-Fender`	`Hazuki: If you have 1 line, * can't miraculously let you do more with it that you can with regular phones. If you're using that line no more calls come in. Plain physics`
`01:25.37`	`[TK]D-Fender`	`Hazuki: * doesn't give you more connectivity than you have`
`01:25.44`	`stoked`	`in my devices, I get the number as the name which is fine, but the the extension name shows up as the number`
`01:25.46`	`[TK]D-Fender`	`stoked: Seems you have the #`
`01:25.55`	`stoked`	`makes it pita to call a number back`
`01:26.19`	`stoked`	`yeah that's odd, I tried removing the stdexten and did a straight Dial and got the same thing`
`01:26.20`	`Hazuki`	`ugh...so what is this about concurrent calls then?`
`01:26.27`	`Hazuki`	`if there's only one call going at a time...`
`01:26.42`	`[TK]D-Fender`	`Hazuki: You call out, someone calls in. FAILURE`
`01:26.44`	`grx0`	`theres only one pots line which means 1 phone call total outside the system`
`01:26.48`	`stoked`	`but on my device I can setup mutiple voip accounts, so I set it up directly to the provider and it works fine`
`01:26.53`	`grx0`	`someone calls they get a busy signal if someones on the phone`
`01:26.57`	`Hazuki`	`[TK]D-Fender, he does have another line, but it's his personal phone`
`01:27.13`	`[TK]D-Fender`	`Hazuki: Fat load of good it does everyone else`
`01:27.20`	`Hazuki`	`[TK]D-Fender, I know.`
`01:27.27`	`Hazuki`	`You're yelling at the wrong person >>;`
`01:27.38`	`[TK]D-Fender`	`Hazuki: No.. you're the messenger... I can't yet at him :p`
`01:27.52`	`[TK]D-Fender`	`hazzPass this on, will ya... "You're a moron"`
`01:27.56`	`grx0`	`ROFL`
`01:28.14`	`sach`	`heh`
`01:28.14`	`Hazuki`	`so if he wants to have, say, 10 calls at once, one to each agent, he's more or less boned and needs 10 lines?`
`01:28.20`	`grx0`	`thinks a "This is a bad idea, thers many cost-effective ways to do this that don't involve a single pots line"`
`01:28.32`	`[TK]D-Fender`	`Hazuki: 10 calls = 10 channels to the PSTN. Do the math.`
`01:28.35`	`grx0`	`he needs to probably get a sip trunk if he's too cheap to pay for a t1/pri.`
`01:28.53`	`sach`	`grx0 i was just about to suggest that`
`01:28.58`	`[TK]D-Fender`	`Yup, most ITSP's offer good multi-channel rates`
`01:29.05`	`Hazuki`	`I thought so...I was considering telling him that. trying to weight "which is worse; the new girl contradicting him but saving his bacon later, or going along with it?"`
`01:29.08`	`sach`	`that makes the most sense. that way you can expand dynamically`
`01:29.22`	`grx0`	`you dont have to directly contradict him to show him he's wrong :D`
`01:29.35`	`[TK]D-Fender`	`Hazuki: Talk "dollars" to him. Thats what they understand. Callers frustrated at getting a busy tone = lost $`
`01:29.45`	`Hazuki`	`so instead of a business-class POTS line, he should get...what's the technical term for this, X-channel SIP line?`
`01:30.00`	`grx0`	`just tell him you have a suggestion as to a better way that you were enlightend to by a crack team of VoIP specialists :)`
`01:30.05`	`sach`	`I think it's called SIP trunking`
`01:30.07`	`rob0`	`Always be a professional. Never set your sights lower than that. Even if it gets you fired, it gets you fired for doing the right thing.`
`01:30.18`	`rob0`	`(and in the long run you'll benefit overall)`
`01:30.20`	`[TK]D-Fender`	`Hazuki: next term to drop "business class POTS". Its a F-ing dumb analog line... stop evangelizing it!`
`01:30.22`	`Hazuki`	`rob0, you're right...thank you, I'm just so nervous ><`
`01:30.34`	`Hazuki`	`[TK]D-Fender, three days ago I'd never heard any of this okay?`
`01:30.50`	`[TK]D-Fender`	`Hazuki: And it should take you less that 1 to drop :)`
`01:31.00`	`Hazuki`	`this is entirely new, worse than that day 5 years ago when I started Linux with a Gentoo install CD and the docs and not a clue`
`01:31.10`	`[TK]D-Fender`	`hangs a giant "MISSION ACCOMPLISHED"" banner`
`01:31.25`	`grx0`	`[TK]D-Fender: well the business class is like premium prices on sale...19.99$ for 20.00$!`
`01:31.37`	`grx0`	`same pots for more $`
`01:31.44`	`Hazuki`	`Okay then, what are the words the professionals of the world in general and the people here in particular prefer for just about everything?`
`01:31.47`	`[TK]D-Fender`	`grx0: Now on special at 50% off twice the original price!`
`01:32.01`	`rob0`	`42`
`01:32.39`	`[TK]D-Fender`	`Hazuki: "1 dumb analog line, 1 dumb administrative plan". That about sums it up`
`01:32.59`	`[TK]D-Fender`	`Hazuki: An analog line does what we've all seen for the past Century pretty much`
`01:33.11`	`Hazuki`	`Googling SIP trunking`
`01:33.24`	`sach`	`Hazuki, good idea`
`01:33.27`	`Hazuki`	`so he needs to find a provider who will give him...a 10-channel SIP trunk?`
`01:33.34`	`Hazuki`	`no, that would be too easy wouldn't it`
`01:33.37`	`[TK]D-Fender`	`Hazuki: T1 links have other signalling options and functionality. ITSP's let you leverage other connectivity at more competitive scaling.`
`01:33.41`	`sach`	`don't need to go that far`
`01:34.06`	`sach`	`many offer an expandable method`
`01:34.06`	`[TK]D-Fender`	`Hazuki: "Need"? No. We aren't here to tell him what he needs. Only to advise on what will FULFILL them.`
`01:34.14`	`Hazuki`	`Okay, let's go back to what he wants: one line in, 10 lines out, all talking out at once if necessary. What is the way to do this?`
`01:34.24`	`[TK]D-Fender`	`Hazuki: 1 in, 10 out. Doesn't work like that.`
`01:34.29`	`sach`	`1 line in?`
`01:34.31`	`grx0`	`quit`
`01:34.34`	`grx0`	`doh`
`01:34.40`	`[TK]D-Fender`	`Hazuki: you are mixing "out" with "# of phones I want just on the inside"`
`01:34.43`	`sach`	`^C^C^C!`
`01:35.19`	`Hazuki`	`Rephrase: "he wants to pay for one connection from the outside world and multiplex it among 10 phones inside the office, and if necessary all 10 should be able to be talking at once." Better?`
`01:35.24`	`[TK]D-Fender`	`Hazuki: What we're debating is your PSTN connection, not the device you want to use so your people can have a phone at their desk`
`01:35.38`	`drmessano`	`Make it look like a sunflower`
`01:35.50`	`[TK]D-Fender`	`Hazuki: Can YOU have an analog line carry 10 conversations to different destinations at the same time on it?`
`01:35.56`	`sach`	`that man is insande`
`01:36.06`	`Hazuki`	`[TK]D-Fender, I do not know. I don't know. This is not my expertise`
`01:36.08`	`[TK]D-Fender`	`drmessano: Can I get that in cornf-lower blue?`
`01:36.18`	`sach`	`no, not over that network`
`01:36.23`	`[TK]D-Fender`	`Hazuki: Do you have an analog phone line at home?`
`01:36.27`	`Hazuki`	`Yes`
`01:36.34`	`drmessano`	`[TK]D-Fender: You can get it in any color you want, as long as it's white`
`01:36.35`	`[TK]D-Fender`	`Hazuki: can YOU do 10 calls at once on it?`
`01:36.39`	`Hazuki`	`stop lecturing me on the physics; I'm just confused over the terminology`
`01:36.51`	`sach`	`lol`
`01:36.54`	`drmessano`	`Oh nice`
`01:36.58`	`[TK]D-Fender`	`Hazuki: 1 dumb analog line = 1 call.`
`01:37.19`	`Hazuki`	`he wants to do something where he pays for one unit of conenctivity service (however the Hell you want to put it) such that he can multiplex it among 10 phones and have them all using it if need be`
`01:37.19`	`drmessano`	`Damn analog users`
`01:37.21`	`[TK]D-Fender`	`Hazuki: that it starts with you calling out, or another call coming in, its still just 1 call`
`01:37.41`	`drmessano`	`I think hes getting mad`
`01:37.46`	`[TK]D-Fender`	`Hazuki: Oh, all will be able to use it... just limited to 1 at a time <-`
`01:37.55`	`Hazuki`	`At the same time, if need be`
`01:37.58`	`sach`	`Hazuki, you can keep the analog line for incoming calls, and then set teh outbound route to a sip trunking provider. that is probably the best way to do it`
`01:38.05`	`*** join/#asterisk Faithful (n=Faithful@121.91.182.207)`
`01:38.07`	`[TK]D-Fender`	`Hazuki: "Joe is on the phone? Oh shit, he's gonna take forever!"`
`01:38.18`	`[TK]D-Fender`	`Hazuki: No "same time". its 1 line`
`01:38.26`	`Hazuki`	`What am I misunderstanding here?`
`01:38.35`	`Hazuki`	`He can't get some big chunk of bandwidth or something?`
`01:38.43`	`[TK]D-Fender`	`Hazuki: You're at home, can someone else pick up one of your phones to place a separate call? No.`
`01:38.58`	`[TK]D-Fender`	`Hazuki: Are we talking POTS here, or an ITSP now?`
`01:39.05`	`Hazuki`	`ITSP I suppose`
`01:39.06`	`rob0`	`You can also have a telco sell you rollover lines, so a single number is called, and any (the first available) of 2 or more lines would ring.`
`01:39.15`	`Hazuki`	`since POTS seems to be a dead end for this`
`01:39.27`	`sach`	`yes, it is.`
`01:39.34`	`rob0`	`I think sach has a good idea.`
`01:39.34`	`[TK]D-Fender`	`Hazuki: Well yes, you can use your internet conenction to pass as many calls as your plan with an ITSP & bandwidth support`
`01:39.37`	`*** join/#asterisk Faithful (n=Faithful@121.91.182.207)`
`01:40.18`	`Hazuki`	`Oooookay. Where would he go to get SIP trunking, and how would he do the think sach mentioned about redirecting to an outbound provider?`
`01:40.28`	`[TK]D-Fender`	`~itsplist-us`
`01:40.28`	`infobot`	`[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net`
`01:40.49`	`[TK]D-Fender`	`Hazuki: and this wouldn't be "redirecting", this would jsut in place of bothering with analog "lines" at all`
`01:41.00`	`sach`	`thanks [TK]D-Fender (and infobot)`
`01:41.00`	`infobot`	`bitte, sach`
`01:41.10`	`sach`	`nice`
`01:41.17`	`sach`	`loves bots`
`01:41.20`	`Hazuki`	`so if he got this ITSP, how would this work? I'm utterly lost now. What kind of cables would tis ITSP run, what do they need to connect to...?`
`01:41.26`	`*** join/#asterisk siera08 (n=chatzill@218.207.141.90)`
`01:42.08`	`sach`	`Hazuki, does he have a broadband line connected to the asterisk server?`
`01:42.21`	`*** join/#asterisk Faithful (n=Faithful@121.91.182.207)`
`01:42.22`	`Hazuki`	`We don't have any of the hardware yet. He wants to have this all planned out ahead of time ><`
`01:42.29`	`[TK]D-Fender`	`Hazuki: Call via IP on your internet connection.`
`01:42.32`	`[TK]D-Fender`	`~voip`
`01:42.33`	`infobot`	`[voip] Voice over IP`
`01:42.35`	`[TK]D-Fender`	`^^^^^^^`
`01:42.43`	`Hazuki`	`I know what voip is...I talk to my love over it`
`01:42.55`	`[TK]D-Fender`	`~itsp`
`01:42.56`	`infobot`	`[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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`
`01:42.58`	`[TK]D-Fender`	`^^`
`01:43.01`	`[TK]D-Fender`	`there you go`
`01:43.30`	`Hazuki`	`so he needs a service where he can originate and terminate (the equivalents of FXS and FXO signalling?)`
`01:43.34`	`[TK]D-Fender`	`Hazuki: Shop around for the one offering the best rate package for a plan that best scales to your needs`
`01:43.43`	`[TK]D-Fender`	`Hazuki: Forget FXS/FXO no`
`01:44.00`	`sach`	`right, if he insists on the stupid analog line, then just keep that for the incoming, and route outgoing through the ITSP`
`01:44.05`	`rob0`	`~Origination`
`01:44.07`	`[TK]D-Fender`	`Hazuki: they send your calls to the PSTN (and get them from there). What devices you talk on in your office is YOUR concern`
`01:44.22`	`rob0`	`What you want is termination.`
`01:44.37`	`[TK]D-Fender`	`Hazuki: For which I'd suggest an SPA-8000 + PAP2, both from Linksys. Cost < $300`
`01:44.38`	`rob0`	`(assuming you keep the POTS line for incoming calls)`
`01:44.42`	`Hazuki`	`he wants people to be able to call in, have * route it to agents...`
`01:44.46`	`[TK]D-Fender`	`rob0: Wants BOTH`
`01:44.48`	`*** join/#asterisk Faithful (n=Faithful@121.91.182.207)`
`01:44.53`	`Hazuki`	`I would rather do this all digital honestly`
`01:44.58`	`Hazuki`	`I'll see if I can talk him out of POTS`
`01:45.10`	`Hazuki`	`since he apparently didn't understand either that 1 POTS line = 1 concurrent call`
`01:45.11`	`[TK]D-Fender`	`Hazuki: Excellent idea, wish we came up with that earlier!`
`01:45.20`	`sach`	`that would be the most sensible option`
`01:45.27`	`Hazuki`	`Cut it out. I've never dealt with any of this and he's making me nuts`
`01:46.24`	`rob0`	`We never cut anything out. Well ... if we do, we'll feed it to you.`
`01:46.25`	`Hazuki`	`What does this look like in the office? When you buy from an ISTP, do they install new cabling?`
`01:46.37`	`sach`	`falls over and dies`
`01:47.00`	`rob0`	`cuts out sach's heart`
`01:47.14`	`Hazuki`	`I'm sorry it's a stupid question ;-; this is a long goddamn day and the boss's cluelessness is getting me and ugh`
`01:47.19`	`[TK]D-Fender`	`Hazuki: I don't thikn you're following. its INTERNET TRAFFIC`
`01:47.30`	`[TK]D-Fender`	`Hazuki: Do you have a special cable for the "voip" you already do?`
`01:47.31`	`*** join/#asterisk Faithful (n=Faithful@121.91.182.207)`
`01:47.33`	`sach`	`Hazuki, they are run __THROUGH__ the internet`
`01:47.34`	`Hazuki`	`But he doesn't need a new separate DSL modem or something?`
`01:48.03`	`[TK]D-Fender`	`Hazuki: He needs an internet conenction. Doesn't matter what as long as it has the bandwidth to support the amount of calls he's expecting`
`01:48.13`	`Hazuki`	`15 mbps cable should be enough to handle his office net needs plus up to 10 simultaneous calls at say 85kbps`
`01:48.20`	`rob0`	`Does the office have computers and Internet already?`
`01:48.20`	`[TK]D-Fender`	`Hazuki: Separate? Not necessarily`
`01:48.26`	`Hazuki`	`rob0, yes`
`01:48.34`	`Hazuki`	`15 mbps TWC, 5 Ubuntu nodes`
`01:48.37`	`Hazuki`	`going to add 5 more`
`01:48.52`	`[TK]D-Fender`	`Hazuki: Ok, your current connection is more than fine`
`01:49.02`	`Hazuki`	`I did enough research to know that much at least ><`
`01:49.08`	`Hazuki`	`remember, I did read 200+ pages of the O`
`01:49.14`	`Hazuki`	`'Reilly book`
`01:50.34`	`Hazuki`	`So, he just has to order service from an ITSP, I need to open the relevant ports on the firewall, and then...?`
`01:50.41`	`*** join/#asterisk Kumbang (n=unknown@125.163.83.153)`
`01:51.36`	`sach`	`well, you have to setup aseterisk but that should be it`
`01:53.01`	`Hazuki`	`and asterisk will handle all the switching and autonegotiating and conversion and everything?`
`01:53.56`	`*** join/#asterisk pmhaddad (n=pmhaddad@24-247-41-171.dhcp.mrqt.mi.charter.com)`
`01:55.04`	`*** join/#asterisk Pegasus_RPG (n=chatzill@cpe-24-164-135-145.si.res.rr.com)`
`01:55.09`	`Pegasus_RPG`	`hello`
`01:55.35`	`Pegasus_RPG`	`Does anyone have experience with using * in conjunction with a legacy PBX behind it?`
`01:55.46`	`Pegasus_RPG`	`(With the goal of phasing out the legacy system)`
`01:57.18`	`[TK]D-Fender`	`Hazuki: Pretty much`
`01:57.39`	`*** join/#asterisk nauticalthinker (n=mratliff@c-76-122-200-95.hsd1.tn.comcast.net)`
`01:57.48`	`*** part/#asterisk Tuxguy (n=jimi@cpe-065-184-197-243.ec.res.rr.com)`
`01:58.05`	`[TK]D-Fender`	`Hazuki: * processes calls the way you set it up to with whaver you configure to have it talk to. ITSP's, analog or digital interfaces, SIP phones, etc... a call is a call is a call. All the same to *.`
`01:58.17`	`Hazuki`	`So I'm still going to need the 4116 to connect all 10 phones to, but no need for an FXO card, and the Asterisk server will need to be on the network because all its traffic comes/goes from it?`
`01:58.56`	`[TK]D-Fender`	`Hazuki: Get multiple DID's from your ITSP and want one treated differently than another? Sure. Want your calls handled differently on tuesdays nights where the Cubs won their last home-game and its currently raining?> Sure`
`01:59.05`	`[TK]D-Fender`	`Hazuki: Forget the 4116... overprices.`
`01:59.25`	`Hazuki`	`Okay, just need something that can connect 10 phones to Asterisk and allow calls in/out in any permutation of them`
`01:59.58`	`[TK]D-Fender`	`Hazuki: Phones connected to * is 1 part. * can handle ALL of them placing calls at one. Where they GO is another matter`
`02:00.35`	`*** join/#asterisk blkry (n=blkry@24-241-112-018.dhcp.gnvl.sc.charter.com)`
`02:00.40`	`Hazuki`	`Yes, but I need to know what kind of hardware is needed to connect all 10 phones to the server. And where they go...well, if he gets SIP trunking from an ITSP I can just point *'s outbound handler at that?`
`02:00.44`	`nauticalthinker`	`any of you guys know nec 2400 programming? I need to send all voicemail from nec to asterisk server`
`02:00.56`	`[TK]D-Fender`	`Hazuki: http://www.voipsupply.com/linksys-spa8000-g1 <- 8 down for $250`
`02:01.16`	`[TK]D-Fender`	`Hazuki: Yes * can handle it all...`
`02:01.33`	`Hazuki`	`[TK]D-Fender, thank you :) Sorry I'm so ignorant on this, it's that he's rushing me and I had no time for thorough research ><`
`02:01.53`	`Hazuki`	`so any ATA will do it, so long as it has ethernet in back for the server and FXS in front for the phones?`
`02:02.15`	`[TK]D-Fender`	`Hazuki: http://www.voipsupply.com/linksys-pap2t-na <- 1 of these for your other 2`
`02:02.37`	`[TK]D-Fender`	`Hazuki: Well the one you were looking at is BIG. More than you needs`
`02:02.54`	`Hazuki`	`I know, but no one makes a bloody 10-FXS port ATA ><`
`02:03.02`	`[TK]D-Fender`	`Hazuki: My way is < 1/3 the cost`
`02:03.15`	`[TK]D-Fender`	`Hazuki: Not ECONOMICALLY priced that is`
`02:03.17`	`Hazuki`	`Yes, I saw...that will help temper this "Boss, we got a problem, we need SIP trunking" email a lot`
`02:04.10`	`[TK]D-Fender`	`Hazuki: Look at it as the fact you can recuperate the cost of paying for that line and your overall connectivity will flourish without extra hardware costs`
`02:04.37`	`Hazuki`	`Yes, the $320 or so on those two plus an SIP trunk will pay for itself in less than half a year of not having 10 bloody analog phone lines`
`02:06.45`	`[TK]D-Fender`	`Hazuki: Well do confirm the actual connectivity costs. Each provider bundles up their services differently .`
`02:06.50`	`Hazuki`	`nods`
`02:07.19`	`Hazuki`	`what kind of plan would I be looking for? It's not "one channel = one concurrent call" is it?`
`02:07.41`	`*** join/#asterisk Kumbang (n=unknown@125.163.83.153)`
`02:08.51`	`[TK]D-Fender`	`Hazuki: Yes, it is.. its a question about how many channels your plan supports at a price you'd prefer to pay vs the alternatives :)`
`02:09.17`	`Hazuki`	`So whatever we decide on in the end, it has to be "10-channel SIP trunking?" Still not sure of the right words for this`
`02:09.35`	`[TK]D-Fender`	`Hazuki: Yeah, that term is accurate enough`
`02:11.05`	`Hazuki`	`okay, if I'm using AsteriskNow, would having an 8-FSX and a 2-FSX ATA connected make it harder to configure?`
`02:11.55`	`*** join/#asterisk chendy (n=chatzill@58.251.100.254)`
`02:13.31`	`[TK]D-Fender`	`Hazuki: not really...`
`02:13.58`	`[TK]D-Fender`	`Hazuki: those 2 devices are pretty close tot the same to configure and each port acts largely like a separate device anyway`
`02:14.20`	`Hazuki`	`So it wouldn't be insanely complicated compared to having a single ATA hooked up? I can tell both of them to respond on the same incoming number but, say if the extension was 0001-0008 it goes to the first one and if 0009 or 0010 to the second?`
`02:14.46`	`Pegasus_RPG`	`Is is possible to have the telco service into the * box and be able to transfer calls to extensions on the PBX if all I have between them is analog FXS/FXO connections?`
`02:15.02`	`*** join/#asterisk JuStIcIa_ (i=john@cbl-sd-74-96.aster.com.do)`
`02:15.58`	`[TK]D-Fender`	`Hazuki: Each port is independent`
`02:16.28`	`Hazuki`	`[TK]D-Fender, so it wouldn't matter much what combination of ATAs were connected to * and how many portd each had since * will configure each port separately?`
`02:17.04`	`[TK]D-Fender`	`Hazuki: Each IS separate from the other. how you want calls to be sent to them is up to you.`
`02:17.31`	`Hazuki`	`The devices with 8 are just more convenient, compact forms than 8 single FXS adaptors then?`
`02:18.03`	`[TK]D-Fender`	`Hazuki: Yes, you may as well consider that as 8 independant devices duct-taped together with a common web interface to configure`
`02:18.23`	`Hazuki`	`Wonderful! I was hoping it was like that(nice analogy too ^^)`
`02:18.29`	`[TK]D-Fender`	`Hazuki: a pretty well layed out tabbed admin page actually`
`02:19.02`	`[TK]D-Fender`	`Hazuki: I've configured 2 of these remotely already`
`02:19.02`	`Hazuki`	`Nice o.o`
`02:19.15`	`Hazuki`	`I am going to need a second NIC for second ATA though rigth?`
`02:19.46`	`[TK]D-Fender`	`Hazuki: These all just get thrown on the same switch as your * box. No need for dedicated NIC's`
`02:19.55`	`Hazuki`	`Oh!`
`02:20.22`	`Hazuki`	`I should have realized that...how will it know how to communicate with the * server though? I thought the ATA had to go into the server`
`02:20.54`	`*** join/#asterisk ingenius (n=alektro@69.90.72.173)`
`02:21.37`	`Hazuki`	`...or do I just set the router to give the ATA a static IP and tell the * server to find it thereZ?`
`02:22.20`	`[TK]D-Fender`	`Hazuki: IP <_ Its all just IP`
`02:22.40`	`[TK]D-Fender`	`Hazuki: * doesn't need to know where the ATA is inherently, the reverse is the norm`
`02:23.05`	`[TK]D-Fender`	`Hazuki: Your devices "register" to * and * knows where they are then`
`02:23.16`	`Hazuki`	`even if their IP addresses are changing?`
`02:23.29`	`[TK]D-Fender`	`Hazuki: Geernally they won't reall... and yes`
`02:23.51`	`[TK]D-Fender`	`Hazuki: the devices will reregister frequently and on getting a new IP, etc`
`02:24.19`	`Hazuki`	`and they're smart enough to seek out a running instance of *, on a network, when needed? Spooky`
`02:26.44`	`[TK]D-Fender`	`Hazuki: No... thats what we call "configuring my device" :)`
`02:26.50`	`[TK]D-Fender`	`Hazuki: It isn't magic.`
`02:27.02`	`[TK]D-Fender`	`Hazuki: IP, user, pass. Thats the usual minimum`
`02:27.09`	`Hazuki`	`I suppose I'll see it when it comes...it can't be much harder than DD-WRT or Tomato`
`02:34.59`	`[TK]D-Fender`	`Hazuki: To configure the device, a lot less. Configureing ASTERISK is another matter.`
`02:35.04`	`[TK]D-Fender`	`Hazuki: Keep reading!`
`02:35.26`	`Hazuki`	`I thought AsteriskNOW provided enough GUI tools to make it pretty painless >>;`
`02:37.28`	`[TK]D-Fender`	`Hazuki: LOL... far from`
`02:37.51`	`[TK]D-Fender`	`Hazuki: Actually... if you use FreePBX with it, that is at least largely "complete".`
`02:38.02`	`[TK]D-Fender`	`Hazuki: Asterisk-GUI is far from`
`02:38.10`	`Hazuki`	`AstLinux?`
`02:40.18`	`*** part/#asterisk dzup (i=dzup@bnc1.shellium.org)`
`02:40.51`	`*** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus)`
`02:44.09`	`[TK]D-Fender`	`Hazuki: Not familiar with. Just be aware that you do sell away the control * offers you when you run a GUI`
`02:44.29`	`Hazuki`	`I know. And as a Gentoo user that idea grates on me. But I also don't have time to learn *`
`02:44.47`	`Hazuki`	`it's less a PBX solution and more a PBX programming language, and I just don't have the time now`
`02:45.11`	`[TK]D-Fender`	`Hazuki: Good understanding.`
`02:45.39`	`Hazuki`	`I like the idea. It's great. But sometimes you need a house fast and don't have the skills to build it brick by brick with no manual`
`02:45.41`	`[TK]D-Fender`	`Hazuki: I'd say to grab the latest AsteriskNOW ISO and install that using the FreePBX interface to start and see how long that floats you`
`02:45.53`	`Hazuki`	`nods That's the plan...I'm reading docs on the site as we speak`
`02:45.56`	`[TK]D-Fender`	`Hazuki: just know that configuring it is not supported here, but in their own channel if you need support`
`02:46.02`	`[TK]D-Fender`	`~freepbx`
`02:46.03`	`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`
`02:46.05`	`*** join/#asterisk captiancrash (n=jonmoore@adsl-074-181-189-229.sip.owb.bellsouth.net)`
`02:46.16`	`Hazuki`	`Thanks for the warning, before I started asking about it :p`
`02:47.43`	`[TK]D-Fender`	`Hazuki: we can still help with your hardware and general understanding questions, but actual configuration will land you in "don't ask" territory :)`
`02:48.07`	`Hazuki`	`nods Thanks so much for your help. My head hurts from all the new knowledge but it's a good hurt`
`02:48.14`	`Hazuki`	`and knowing this stuff will make me more valuable`
`02:48.27`	`*** join/#asterisk Mw3 (i=mw3@mw3.hu)`
`02:49.27`	`Hazuki`	`curls up in front of the fire with her Hatsumi plush and snoozes z_z`
`02:56.28`	`*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)`
`02:58.18`	`[TK]D-Fender`	`loves his new (used) Roland XV-3080`
`03:04.44`	`*** part/#asterisk Pegasus_RPG (n=chatzill@cpe-24-164-135-145.si.res.rr.com)`
`03:14.40`	`*** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio)`
`03:22.25`	`sach`	`ok off to lunch.`
`03:23.08`	`*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)`
`03:26.00`	`*** join/#asterisk tdjacobs (n=tiagoj@200.213.73.100)`
`03:39.02`	`*** join/#asterisk Kumbang (n=unknown@125.163.83.153)`
`03:43.23`	`*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)`
`03:49.07`	`jplank`	`would you believe this is actually in one of my companies fluff pieces about asterisk "Based on Linux, sparing the IT group the need to constantly install software patches for the Windows operating system. "`
`03:49.20`	`drmessano`	`lol`
`03:49.41`	`jplank`	`I loved it`
`03:49.46`	`jplank`	`lol`
`03:50.18`	`rob0`	`An unsecured Linux can be as bad, if not worse, than a Windows.`
`03:51.41`	`rob0`	`Not that I really disagree with the selling point, I guess.`
`03:54.02`	`drmessano`	`As if "Not being windows" is really a selling point for something not traditionally run on windows\`
`03:54.20`	`drmessano`	`My toast "isnt windows".. so fuckin what`
`03:54.26`	`drmessano`	`My toastER "isnt windows".. so fuckin what`
`03:55.11`	`florz`	`no, toasters are netbsd, of course`
`03:57.37`	`[TK]D-Fender`	`jblack: Next, Asterisk is a piece os SOFTWARE, not an operating system, and you'r right... doesn't require patches, only completely recompiling including kernel modules`
`03:57.46`	`[TK]D-Fender`	`jplank: Next, Asterisk is a piece os SOFTWARE, not an operating system, and you'r right... doesn't require patches, only completely recompiling including kernel modules`
`03:57.48`	`[TK]D-Fender`	`rather`
`03:59.06`	`jplank`	`Well it said "based" on linux`
`03:59.16`	`[TK]D-Fender`	`jplank: Whic it isn't`
`03:59.22`	`jplank`	`but I guess anything that doesn't run on windows is ++1`
`03:59.54`	`[TK]D-Fender`	`jplank: Linux is based on UNIX in the sense that it at least implements most POSIX style interfaces and happens to be an OS kernel as well`
`04:00.01`	`florz`	`jplank: 1 is not an lvalue`
`04:00.03`	`[TK]D-Fender`	`jplank: But * CAN run on Windows.`
`04:00.24`	`[TK]D-Fender`	`jplank: Sorry little fish, but you're swimming upstream on this one...`
`04:00.32`	`jplank`	`hey now, I didn't write it, I thought it was funny`
`04:00.51`	`pdmmm`	`[TK]D-Fender: u can argue Linux isnt based on UNIX actually`
`04:00.51`	`jplank`	`* can also be run inside VM, doesn't mean it should :)`
`04:00.59`	`pdmmm`	`its more of the stepchild of UNIX`
`04:01.16`	`florz`	`[TK]D-Fender: BTW, Linux is a piece of SOFTWARE, too.`
`04:01.37`	`[TK]D-Fender`	`pdmmm: "based" in the sense of at least trying to emulate in some larger capacity, not an implied derivaive`
`04:01.47`	`[TK]D-Fender`	`derivative*`
`04:02.26`	`[TK]D-Fender`	`pdmmm: I'll lump this in the same family as nit-picking what WINE is ;)`
`04:02.38`	`pdmmm`	`ha`
`04:04.18`	`pdmmm`	`i actually made my statement to be funny`
`04:04.18`	`pdmmm`	`:)`
`04:04.18`	`pdmmm`	`but i agree`
`04:04.19`	`pdmmm`	`Asterisk is arguably a software application, which too has needs to be upgraded as development continues.`
`04:04.29`	`*** join/#asterisk j_kroon (n=jkroon@dsl-240-187-242.telkomadsl.co.za)`
`04:04.53`	`jplank`	`I think they were getting at the whole update every second tuesday of every month thing, and still break everything`
`04:05.30`	`jplank`	`but I'd rather build and secure a linux box over a windows box any day of the week`
`04:06.13`	`pdmmm`	`mac has a update every tuesday :)`
`04:06.32`	`jplank`	`does mac's updates break something every tuesday?`
`04:06.43`	`pdmmm`	`my heart. :(`
`04:06.46`	`jplank`	`lol`
`04:07.05`	`pdmmm`	`hehe`
`04:07.06`	`pdmmm`	`funny`
`04:07.17`	`pdmmm`	`im a little loony, i've been putting in long hours @ work`
`04:08.39`	`jplank`	`its hard to get mad at that though, working late goes synonymous with any type of tech career`
`04:10.54`	`pdmmm`	`yeah`
`04:14.44`	`[TK]D-Fender`	`[00:06]<jplank>does mac's updates break something every tuesday? <- in 2005 OS X 10.4.2 Killed my marketing department when an SMB issue caused Adobe Illustrator to CRASH constantly.`
`04:14.55`	`pdmmm`	`neener`
`04:14.56`	`[TK]D-Fender`	`jplank: so... YES, it was a very bad tuesday`
`04:15.38`	`*** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net)`
`04:17.15`	`jplank`	`I'm no way near a mac person, but if you need to go back to 2005 to find the last time a update broke something, makes it a hard comparison against windows`
`04:18.06`	`drmessano`	`I dont remember a critical break in windows like that going back at least that long`
`04:18.48`	`drmessano`	`Unless you count XP SP2, but thats 2004`
`04:19.51`	`[TK]D-Fender`	`SP2 never posed a problem for me.`
`04:19.54`	`jplank`	`windows botched attempt at the daylight savings time update`
`04:20.46`	`jplank`	`or windows sending out an update, without notifying anyone it changes the way permissions worked, and TONS of businesses losing email access`
`04:21.24`	`jplank`	`I remember that day very well, our IT guy was in Japan, and everyone was looking at me to fix the mail server`
`04:21.52`	`drmessano`	`I remember nothing of the sort`
`04:22.03`	`drmessano`	`Certainly not "tons" of businesses`
`04:22.44`	`drmessano`	`and the DST updates were prior to the time change, so while they were wrong, there was no effect from it`
`04:23.13`	`jplank`	`tell that to the tons of people who were sending calendar invites into the future`
`04:23.35`	`drmessano`	`Where are you getting these numbers?`
`04:23.42`	`drmessano`	`TONS and TONS?`
`04:23.57`	`jplank`	`the permissions issue effected EVERY bes install`
`04:24.04`	`jplank`	`(I'm looking it up)`
`04:24.07`	`drmessano`	`I remember very few issues from any of the updates.. most stemming from LACK of updates`
`04:24.14`	`*** join/#asterisk chendy (n=chatzill@58.251.100.254)`
`04:24.19`	`*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)`
`04:24.41`	`jplank`	`and the DST issue, I'm judging by the weeks of receiving calendar invites from various companies that had the wrong date/time`
`04:25.01`	`drmessano`	`Again, lack of updates`
`04:26.50`	`stoked`	`is there a setting in Asterisk to set the "Contact" attribute in the SIP INVITE packet?`
`04:27.24`	`[TK]D-Fender`	`stoked: its typically the userid for the peer`
`04:27.46`	`stoked`	`[TK]D-Fender ah thanks... I'll try that... think that's what my callerid problem is`
`04:27.56`	`stoked`	`I compared tcpdumps in wireshark`
`04:28.11`	`jplank`	`can sipaddheader modify the contact field?`
`04:28.22`	`[TK]D-Fender`	`stoked: And you still haven't shown your configs or the SIP debug of the call * places to your phone`
`04:28.49`	`[TK]D-Fender`	`stoked: Now I DO know a very specific parameter that many people set and shouldn't than can cause this precise thing.`
`04:29.02`	`[TK]D-Fender`	`stoked: So please PB up your peer masking only passwords`
`04:29.13`	`stoked`	`ok`
`04:29.23`	`jplank`	`don't forget to mask the fact that your using trixbox ;)`
`04:29.33`	`jplank`	`fender loves that`
`04:30.15`	`thehar`	`freepbx`
`04:30.28`	`thehar`	`haw`
`04:30.31`	`thehar`	`lurks more`
`04:33.58`	`stoked`	`http://pastebin.com/d4e08620d`
`04:34.54`	`[TK]D-Fender`	`fromuser=mbu400 <- CULPRIT`
`04:34.56`	`[TK]D-Fender`	`YUP`
`04:35.14`	`stoked`	`oh what should I use?`
`04:35.16`	`[TK]D-Fender`	`stoked: and "type=peer" <- what it should be generally`
`04:35.39`	`[TK]D-Fender`	`stoked: just REMOVE the "fromuser" line`
`04:35.43`	`stoked`	`ohhh`
`04:35.50`	`stoked`	`k will try`
`04:36.49`	`stoked`	`bingo!`
`04:36.51`	`stoked`	`that works now`
`04:36.52`	`stoked`	`thanks`
`04:37.08`	`[TK]D-Fender`	`stoked: You're welcome`
`04:37.21`	`[TK]D-Fender`	`stoked: And the real issue is the Fron: header being overriden`
`04:37.28`	`[TK]D-Fender`	`from`
`04:37.45`	`stoked`	`yeah I noticed that in my tcpdumps`
`04:38.14`	`stoked`	`awesome thanks man, much appreciated`
`04:38.38`	`[TK]D-Fender`	`stoked: No prob`
`04:46.06`	`*** join/#asterisk Beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org)`
`04:50.12`	`*** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128544476.dsl.bell.ca)`
`04:51.24`	`Beave`	`hey gang....`
`04:51.56`	`Beave`	`does anyone know of a voip adapter (1/2 port) that supports SRTP or IAX2 encryption (less likely I know) ?`
`04:54.10`	`jplank`	`is there any command that would let me read from a list, and check if an entry is in the list, from the dial plan?`
`04:55.12`	`[TK]D-Fender`	`jplank: depends on what this "list" is`
`04:56.47`	`jplank`	`just a list of phone numbers`
`04:57.12`	`Beave`	`Writing a AGI comes to mind...`
`04:58.18`	`Beave`	`jblack: would be trival to do....`
`05:00.28`	`jplank`	`I was hoping for something that worked like authenticate`
`05:02.01`	`Beave`	`jplank: not that I can think of right off... but again, an AGI.....`
`05:02.04`	`[TK]D-Fender`	`jplank: ok, maybe the concept doesn't come so quickly to you. It doesn't matter what data you have in your list, it matter what your list is STORED IN.`
`05:02.30`	`[TK]D-Fender`	`jblack: What is it? a DB? A text file? Somehting you have to lookup off a web server? Pulled out of thin air?`
`05:02.42`	`[TK]D-Fender`	`jplank: ^^`
`05:02.47`	`[TK]D-Fender`	`dangit`
`05:03.03`	`jplank`	`ohhh, I got you`
`05:03.17`	`jplank`	`I meant just a text file (hence my reference to authenticate)`
`05:03.35`	`jplank`	`but really, a text file, DB entry, whatever could work`
`05:03.51`	`jplank`	`I'd use astDB, but doesn't that get cleared after a restart?`
`05:04.06`	`[TK]D-Fender`	`jplank: No`
`05:05.06`	`[TK]D-Fender`	`jplank: "core show application authenticate"`
`05:05.12`	`jplank`	`err actually I don't think astDB would work, I could only do get and put`
`05:05.28`	`[TK]D-Fender`	`jplank: Put data in, and you can check if its there <-`
`05:05.31`	`jplank`	`authenticate prompts for a password`
`05:05.41`	`[TK]D-Fender`	`jplank: Yes & no.`
`05:05.46`	`jplank`	`no?`
`05:05.55`	`[TK]D-Fender`	`jplank: Read it again and open you mind a bit.`
`05:06.18`	`jplank`	`oh wait, astDB would work huh, I could just make the item I'm searching for the key`
`05:06.26`	`jplank`	`and the value is irrelevant`
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`07:29.25`	`Psychobilly`	`hello, i have a prob with * 1.4, i have this in my ael code and its not working: SEC=${STRFTIME(${EPOCH},,%S)};`
`07:29.38`	`Psychobilly`	`i get this error`
`07:29.40`	`Psychobilly`	`<PROTECTED>`
`07:29.40`	`Psychobilly`	`[Apr 22 22:14:31] WARNING[28515]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '<token>'; Input:`
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`07:31.05`	`Psychobilly`	`in the other hand this wirks perfectly: Saynumber(${STRFTIME(${EPOCH},,%S)});`
`07:31.23`	`Psychobilly`	`any ideas?`
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`07:32.27`	`stochastik`	`If I have a TRUNK=SIP/bob and TRUNK=SIP/matt, is there a way to load balance which trunk gets used?`
`07:37.32`	`wdoekes`	`for dialing Dial(${EXTEN}@${TRUNK}) in your extensions?`
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`07:42.28`	`[gnubie]`	`can anyone highlight the advantage of using 1.6 over 1.4?`
`07:42.33`	`[gnubie]`	`waves`
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`08:25.38`	`proxium`	`Hi everybody, I have to install Vicidial and astGUI/Vicidial on a separate server, but I can't found any document or Help about it, can anyone tell me if is it possible to do so, and how it's done ?`
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`08:30.29`	`popolinux`	`Hello`
`08:34.25`	`popolinux`	`I try to send fax with txfax(), but I have an error : "set_format: Unable to find a codec translation path from alaw to unknown"`
`08:34.40`	`popolinux`	`Anybody know this problem ?`
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`09:26.42`	`EmleyMoor`	`When I have sip debug on, do all headers appear regardless of the core verbose setting?`
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`09:31.40`	`hi365`	`all sip headers, iirc`
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`10:40.28`	`plundra`	`Can't you get announcing in a queue, when using ringing instead of moh?`
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`10:48.42`	`aryonoco`	`hi, I have a question about compiling dahdi-linux under Xen (Amazon EC2 to be exact)`
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`10:49.26`	`aryonoco`	`I have the source of the kernel installed and available, but I don't know how to set KVERS in the makefile to compile for that kernel`
`10:49.33`	`aryonoco`	`can anyone help?`
`10:55.39`	`tzafrir_laptop`	`aryonoco, yes, you, by helping to test http://bugs.digium.com/view.php?id=13930`
`10:55.47`	`tzafrir_laptop`	`It is reported to work`
`10:56.19`	`tzafrir_laptop`	`KVERS defaults to the output of uname -r`
`10:56.45`	`tzafrir_laptop`	`it is also the name of the directory under /lib/modules`
`10:58.58`	`aryonoco`	`ok, if it defaults to uname -r then it should be fine`
`10:59.06`	`aryonoco`	`just another question about dahdi-tools`
`10:59.18`	`aryonoco`	`the documentation only mentions two ./configure options for it`
`10:59.26`	`aryonoco`	`./configure --without-ncurses CC="gcc-4.10"`
`10:59.41`	`aryonoco`	`is there anything else I can set?`
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`11:06.55`	`aryonoco`	`dahdi-linux compiled fine, but I'm having issues compiling dahdi-tools`
`11:07.48`	`aryonoco`	`I got the current dahdi-tools tarball, 2.1.0.2, and did the following`
`11:08.14`	`aryonoco`	`./configure cc=gcc-4.0.2`
`11:08.15`	`aryonoco`	`make`
`11:08.17`	`aryonoco`	`make install`
`11:08.18`	`aryonoco`	`make config`
`11:08.49`	`aryonoco`	`now any program that I want to run can't find libstdc++.so.6 and other standard libraries`
`11:09.15`	`aryonoco`	`for example apt-get says: apt-get: /usr/local/lib/libstdc++.so.6: no version information available (required by apt-get)`
`11:09.29`	`aryonoco`	`any hints on what I'm missing here?`
`11:10.15`	`tzafrir_laptop`	`aryonoco, aptitude install build-essentials`
`11:10.39`	`tzafrir_laptop`	`nots of the existing, pre-built dahdi packages`
`11:11.07`	`tzafrir_laptop`	`http://updates.xorcom.com/pkg-voip/`
`11:11.45`	`aryonoco`	`thanks for that tzafrir_laptop`
`11:11.46`	`tzafrir_laptop`	`aryonoco, though those do not contain this patch`
`11:11.57`	`aryonoco`	`but these are all Debian packages`
`11:12.03`	`aryonoco`	`are there any Ubuntu ones?`
`11:22.16`	`tzafrir_laptop`	`aryonoco, no`
`11:22.31`	`tzafrir_laptop`	`though it is probably mostly a matter of rebuilding`
`11:23.03`	`tzafrir_laptop`	`(and then adding backport/* scripts)`
`11:27.12`	`aryonoco`	`tzafrir_laptop, thanks`
`11:27.28`	`aryonoco`	`but do you know of any particular reason why compiling from source doesn't work?`
`11:34.16`	`Psychobilly`	`i have a prob with * 1.4, i have this in my ael code and its not working: SEC=${STRFTIME(${EPOCH},,%S)};`
`11:34.19`	`Psychobilly`	`i get this error`
`11:34.32`	`Psychobilly`	`[Apr 23 14:34:10] WARNING[29206]: ast_expr2.fl:411 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.`
`11:34.32`	`Psychobilly`	`<PROTECTED>`
`11:34.49`	`Psychobilly`	`on the other hand this works perfectly: Saynumber(${STRFTIME(${EPOCH},,%S)});`
`11:38.16`	`proxium`	`<PROTECTED>`
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`12:01.51`	`merlin8282`	`Hi ! Is there a way for a non-queuemember to pickup a call that is ringing in a queue ?`
`12:01.58`	`merlin8282`	`using Asterisk 1.4.22`
`12:02.42`	`freh`	`merlin8282, Joining the queue.`
`12:03.19`	`freh`	`you could add an extension that executes the AddQueuemember() application`
`12:04.16`	`merlin8282`	`ok. In fact, i'm trying to program an extension that offers the possibility to pickup a ringing extension without knowing its number (such as *8XXX without knowing the XXX)`
`12:04.57`	`freh`	`Then someone just dials that extension and ends up as a queuemember`
`12:05.19`	`*** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903)`
`12:05.43`	`merlin8282`	`ok.`
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`12:06.31`	`freh`	`merlin8282, I have no experience with that. Here I have an extension that just adds queuemembers, and another that removes them`
`12:06.49`	`merlin8282`	`Ok, i'll search this way.`
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`12:07.37`	`freh`	`But you're idea might be possible. I'm just starting to use queues myself`
`12:09.05`	`makafre`	`good morning all, I would need help with voicemail messages that are not attached to e-mails`
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`12:09.11`	`creativx`	`merlin8282: just to pick up any ringing extension in a given queue?`
`12:09.30`	`merlin8282`	`creativx: yes`
`12:09.43`	`creativx`	`should be possible`
`12:10.12`	`creativx`	`but whats your motivation`
`12:10.25`	`creativx`	`if you have 2 queue members in a call, and a 3rd call comes into the queue`
`12:10.33`	`creativx`	`do you want a backup person to be able to snap up that 3rd call`
`12:11.03`	`merlin8282`	`something like this, yes.`
`12:11.04`	`makafre`	`if I delete all messages from INBOX and leave a voicemail, file msg0000.wav is created but asterisk looks to attach msg0001.wav to the e-mail`
`12:11.20`	`*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)`
`12:11.24`	`creativx`	`morning [TK]D-Fender`
`12:12.03`	`merlin8282`	`creativx: in fact, we don't really have a hotline, but I use a queue instead of Dial() a variable containing all members I want.`
`12:12.18`	`merlin8282`	`Dial()ing*`
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`12:12.24`	`creativx`	`merlin8282: well.. im not sure how that would be done in extensions.conf`
`12:13.17`	`Quintana`	`je suis le maÃ®tre du monde !`
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`12:13.31`	`merlin8282`	`I already am able to pickup intern ringing phones (internal calls), but not calls coming from external.`
`12:13.37`	`merlin8282`	`Quintana: menteur`
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`12:16.40`	`makafre`	`guys, if I delete all messages from INBOX and leave a voicemail, file msg0000.wav is created but asterisk looks to attach msg0001.wav to the e-mail, what is wrong?`
`12:16.44`	`merlin8282`	`What I'm doing is picking up the last ringing phone (each time a call is placed or received I update a database variable)`
`12:17.30`	`merlin8282`	`...and then Pickup() with this variable does.`
`12:18.05`	`merlin8282`	`My problem is now setting this variable when an external call comes in, into my queue.`
`12:18.12`	`tzafrir_laptop`	`Psychobilly, please use verbosity level 3 and see what you actually run. Also: can you pastebin how the generated dialplan looks? dialplan show <name of context>`
`12:18.41`	`Quintana`	`merlin8282, ;)`
`12:18.49`	`Psychobilly`	`tzafrir_laptop the erors i pasted where from the console with ver set to 3`
`12:18.52`	`merlin8282`	`This variable is set to the phone that has to be rung.`
`12:19.20`	`merlin8282`	`Hence this problem : which number does a queue have ? XD`
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`12:19.49`	`atis_work`	`Psychobilly: ok, what was your question? i recall some knowledge of STRFTIME :)`
`12:19.53`	`Psychobilly`	`tzafrir_laptop and thats the generated dialpan : 4. Set(SEC=$[${STRFTIME(${EPOCH},,%S)}]) [pbx_ael]`
`12:20.14`	`merlin8282`	`An alternative for me could be to not use a queue and come back to a Dial() based solution.`
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`12:20.20`	`Psychobilly`	`atis_work: i have this in my ael code and its not working: SEC=${STRFTIME(${EPOCH},,%S)};`
`12:20.24`	`Psychobilly`	`i get this error`
`12:20.29`	`Psychobilly`	`[Apr 23 14:34:10] WARNING[29206]: ast_expr2.fl:411 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.`
`12:20.32`	`Psychobilly`	`<PROTECTED>`
`12:20.41`	`Psychobilly`	`on the other hand this works perfectly: Saynumber(${STRFTIME(${EPOCH},,%S)});`
`12:22.20`	`atis_work`	`Psychobilly: i never used pure variable assignations in AEL, it's weird..`
`12:22.32`	`atis_work`	`Psychobilly: try using Set(a=${...}) in AEL`
`12:22.39`	`Psychobilly`	`hm`
`12:22.42`	`Psychobilly`	`ok let me try`
`12:24.14`	`Psychobilly`	`it worked!`
`12:24.18`	`Psychobilly`	`thx atis_work`
`12:24.47`	`*** join/#asterisk Pan3D (n=Pan3D@node2.sensoryresearch.net)`
`12:25.08`	`Psychobilly`	`but again why the first assigment method didnt worked... strange`
`12:26.05`	`Psychobilly`	`the old line was interpreted as Set(SEC=$[${STRFTIME(${EPOCH},,%S)}]) the new as Set(SEC=${STRFTIME(${EPOCH}\|\|%S)})`
`12:26.30`	`jjshoe`	`in voicemail.conf does format= also tell it which greeting to play first? like busy.wav vs busy.WAV or is there some other order of priority going on?`
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`12:27.26`	`atis_work`	`Psychobilly: not sure, AEL tends to wrap everything it considers an expression in $[...]`
`12:27.27`	`[TK]D-Fender`	`jjshoe: automatic the same as codec negotiation`
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`12:31.32`	`defsdoor`	`hello - remember my problem yesterday - no voice on sangoma a101 pri - turned off hardware echo canx and it works`
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`12:32.25`	`J4zen`	`Hi there, quick question. I remember there being an AGI tool for asterisk which users could use to check their voicemail and forward calls, pretty basic and easy to use. Does anyone remember the name of such an application`
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`12:32.39`	`J4zen`	`it might have been bundled with a previous version of Trixbox or so`
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`12:33.33`	`ariel_`	`Morning everyone`
`12:35.01`	`[TK]D-Fender`	`defsdoor: In some rare cases teh HWEC buffer can get locked up and all voice gets stuck spooling in there. there is a fluch command for this but i don't recall what it is`
`12:35.57`	`[TK]D-Fender`	`J4zen: "forward calls" = not unless you have a whole lot of other dialplan, and if we're talking about trixbox this is not even a question to ask here.`
`12:36.22`	`[TK]D-Fender`	`J4zen: "Asterisk" has no such thing, its 3rd party`
`12:36.51`	`[TK]D-Fender`	`J4zen: "core show application voicemailmain" <- there's how to check your voicemail.`
`12:38.05`	`J4zen`	`True, but it was a standalone agi script that is used commonly with Asterisk. It's not a part of Trixbox, as i recall it used to be bundled with it.. might have been something else. Since im merely asking about the name of an AGI script people commonly use with Asterisk vanilla i figured this would be the best place to ask.`
`12:38.09`	`J4zen`	`Correct me if i'm wrong`
`12:39.37`	`[TK]D-Fender`	`J4zen: Not possible.`
`12:39.52`	`[TK]D-Fender`	`J4zen: there is no "forwarding" unless your entire dialplan revolves around it`
`12:40.15`	`J4zen`	`you're right, i missed that.`
`12:40.17`	`J4zen`	`thanks [TK]D-Fender`
`12:40.35`	`J4zen`	`Must have been trixbox/freepbx after all`
`12:41.28`	`*** join/#asterisk propellerhead (n=yogurt2u@host170.190-31-201.telecom.net.ar)`
`12:42.58`	`[TK]D-Fender`	`J4zen: Every dialplan is custom so you can't jsut throw around an AGI that is expected to have any impact on other aspects of your dialplan unless you're in a complete cokie-cutter GUI world. Was kinda silly to ask...`
`12:43.28`	`ariel_`	`I would like to know if anyone has had any issues in sending calls via sip to IP address of an asterisk system from a Cisco 3845. We get the call starts ringing but as soon as it's answered it hangs up.`
`12:43.31`	`J4zen`	`yeah you're absolutely right, completely missed that. busy day i suppose :)`
`12:44.49`	`[TK]D-Fender`	`ariel_: If you're going to ask then you should really already have a pastebin of a complete failed call attempt at verbose 10, and SIP debug enabled ready to show us....`
`12:45.07`	`*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)`
`12:45.14`	`[TK]D-Fender`	`ariel_: c'mon... you know how this works...`
`12:45.36`	`*** join/#asterisk juanIMP (n=Juancho@200.71.41.22)`
`12:46.27`	`ariel_`	`[TK]D-Fender: well yes I know but what is the biggest issue is that they all show it's working normal. Only sign it has different is that the Cisco sends a Sip 487 which is a hang up. At this stage I am just fishing as all appears normal.`
`12:47.21`	`ariel_`	`I will have all the logs from the 3845 later on today as my provider is going to be sending them. Just wondering off handed if anyone has seen this issue. As google and other voip FAQ don't have any clue.`
`12:48.41`	`[TK]D-Fender`	`ariel_: Don't fire shots off randomly hoping there is a documented case with your model # directly attached to it.`
`12:48.54`	`[TK]D-Fender`	`ariel_: this is almost never the case. So Go provide backup.`
`12:55.05`	`*** join/#asterisk t0rrieri (n=Torrieri@nelug/crew/torrieri)`
`12:57.27`	`*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)`
`13:01.22`	`*** join/#asterisk brad_mssw (n=brad@216.155.101.90)`
`13:02.06`	`*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`13:08.53`	`*** join/#asterisk mazpe (n=mazpe@c-71-196-32-22.hsd1.fl.comcast.net)`
`13:09.05`	`*** join/#asterisk matsk (n=matkar@c-ad89e253.174-6-64736c10.cust.bredbandsbolaget.se)`
`13:09.19`	`*** join/#asterisk captiancrash (n=jonmoore@70.159.118.70)`
`13:10.40`	`mazpe`	`is 'username' deprecated from sip.conf?`
`13:10.48`	`mazpe`	`i keep seen: chan_sip.c: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser' on my 1.6.x`
`13:12.08`	`*** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-10-130.w90-56.abo.wanadoo.fr)`
`13:13.40`	`creativx`	`mazpe: that would be a wild guess?`
`13:15.07`	`[TK]D-Fender`	`"What is this warning under the big red button labeled 'Don't Touch' trying to tell me?"`
`13:15.22`	`mazpe`	`exactly my point.`
`13:15.28`	`mazpe`	`no red button :)`
`13:15.46`	`mazpe`	`i was just wondering if it changed for sip peers... or all sip types.`
`13:15.50`	`*** join/#asterisk AsteriskDom (n=Asterisk@75.112.88.200.m.sta.codetel.net.do)`
`13:18.07`	`EmleyMoor`	`If I wanted to have a go at rewriting my dialplan in ael, for 1.4, is that AEL v2?`
`13:19.09`	`eppigy`	`DONDE ESTA`
`13:19.35`	`*** join/#asterisk Iskorptix_ (n=iskorpti@d205.csc.lt)`
`13:19.37`	`Iskorptix_`	`hello`
`13:20.00`	`Iskorptix_`	`when I change timezone in voicemail.conf , messages still comming with bad time`
`13:20.10`	`Iskorptix_`	`what else I have to change to make an impact to time ?`
`13:20.14`	`[TK]D-Fender`	`mazpe: in 1.6 "peer" is the only real type`
`13:20.20`	`Iskorptix_`	`(system has uses time)`
`13:20.27`	`[TK]D-Fender`	`mazpe: the concept of "user" & "friend" are being done away with`
`13:20.34`	`Iskorptix_`	`(system uses good time*)`
`13:20.40`	`EmleyMoor`	`Iskorptix_: What do you mean by "bad time"?`
`13:21.14`	`*** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com)`
`13:21.16`	`mazpe`	`[TK]D-Fender: i see`
`13:21.48`	`*** join/#asterisk alvar (n=quassel@81.221.180.109)`
`13:22.04`	`Iskorptix_`	`EmleyMoor: for example I'm getting such messages from asterisk : <...> Date: Thursday, April 23, 2009 at 01:10:42 PM <...> , but I know that time is ok , I mean the time should be 16:10pm`
`13:22.52`	`EmleyMoor`	`Iskorptix_: What time zone are you in?`
`13:23.12`	`Iskorptix_`	`Isreal/Jerusalem`
`13:23.14`	`*** join/#asterisk qdk (n=qdk@81.7.168.130)`
`13:23.27`	`Iskorptix_`	`gmt+2`
`13:23.27`	`mazpe`	`this is my sip account: http://rafb.net/p/NLwoIl87.html`
`13:23.40`	`EmleyMoor`	`So 14:10 UTC...`
`13:23.48`	`Iskorptix_`	`EmleyMoor: yes`
`13:23.50`	`mazpe`	`my phone registration is just username: gc100 and password: 100`
`13:23.54`	`mazpe`	`that should work right?`
`13:24.30`	`captiancrash`	`Iskorptix_, not sure if it's here, but i have a lot of issues with time + linux when my hardware clock isn't set correctly....`
`13:24.40`	`mazpe`	`all i get is: chan_sip.c:18390 handle_request_register: Registration from '<sip:gc100@1.1.1.1;transport=UDP>' failed for '2.2.2.2' - No matching peer found`
`13:25.09`	`mazpe`	`sip:phone100`
`13:25.29`	`EmleyMoor`	`Iskorptix_: What name have you used for the timezone in voicemail.conf?`
`13:25.36`	`Aiatek`	`take off`
`13:25.40`	`Aiatek`	`dial=SIP/phone_100`
`13:25.58`	`Aiatek`	`deny=0.0.0.0/0.0.0.0`
`13:26.08`	`EmleyMoor`	`(Oh - was that the time 15 mins ago? If so you are UTC+2 (UTC 13:10))`
`13:26.16`	`EmleyMoor`	`UTC}3 even`
`13:26.16`	`*** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk)`
`13:26.19`	`EmleyMoor`	`+3`
`13:26.27`	`mazpe`	`ok`
`13:26.36`	`Iskorptix_`	`EmleyMoor: tz=central \|\|\|\|\|\|\| central=Israel/Jerusalem\|'vm-received' Q 'digits/at' IM`
`13:27.08`	`Iskorptix_`	`captiancrash: I bet hardware clock is not important here, because I have few other applications running there and they provide me ideal time`
`13:27.28`	`EmleyMoor`	`Iskorptix_: Hold on...`
`13:28.30`	`EmleyMoor`	`Iskorptix_: Change it to Asia/Jerusalem`
`13:28.34`	`*** join/#asterisk oej (n=olle@80.251.192.3)`
`13:30.09`	`Iskorptix_`	`EmleyMoor: same`
`13:30.34`	`Iskorptix_`	`EmleyMoor: oh no`
`13:30.36`	`*** join/#asterisk captiancrash (n=jonmoore@70.159.118.70)`
`13:30.38`	`Iskorptix_`	`it worked`
`13:30.44`	`Iskorptix_`	`thanks a million EmleyMoor !`
`13:31.00`	`*** join/#asterisk arnuld (n=arnuld@unaffiliated/arnuld)`
`13:31.01`	`mazpe`	`Aiatek: still not authenticating`
`13:31.07`	`arnuld`	`Hello`
`13:31.14`	`Aiatek`	`permit=0.0.0.0/0.0.0.0`
`13:31.19`	`Aiatek`	`remove it too`
`13:31.47`	`*** join/#asterisk arnuld (n=arnuld@unaffiliated/arnuld)`
`13:31.52`	`Aiatek`	`and make a reload after thtat in the CLI`
`13:32.02`	`mazpe`	`no luck`
`13:32.04`	`arnuld`	`I did a lot of google search but did not come up with anything`
`13:32.26`	`mazpe`	`yet if i change the username in the device to phone_101 it works`
`13:32.34`	`Aiatek`	`yes`
`13:32.46`	`Aiatek`	`im eas looking that right now`
`13:33.08`	`*** join/#asterisk arnuld (n=arnuld@unaffiliated/arnuld)`
`13:33.22`	`arnuld`	`If I am using "asterisk" to make some calls then it sends return code 5 for busy call and 4 for successful pick-up of the call`
`13:33.22`	`EmleyMoor`	`(I think it was giving time at the IDL!)`
`13:33.25`	`Aiatek`	`username=phone_100`
`13:33.29`	`*** join/#asterisk lanning (n=lanning@173.8.187.197)`
`13:33.48`	`arnuld`	`but asterisk does not return anything if user picks-up and then disconnectes the call`
`13:34.12`	`mazpe`	`the username has to the same as the sip name?`
`13:34.14`	`EmleyMoor`	`(Yankee time)`
`13:35.08`	`arnuld`	`how can I know if the user has disconnected the call`
`13:35.25`	`arnuld`	`because on disconnection I am not getting anything as return code`
`13:35.36`	`arnuld`	`or is this a feature of asterisk`
`13:35.38`	`arnuld`	`?`
`13:35.52`	`arnuld`	`and I need to do something else to get this thing done`
`13:36.14`	`angryuser`	`arnuld, what do you mean that you dont get anything ? no sip message "BYE" ?`
`13:36.51`	`angryuser`	`or CANCEL`
`13:37.42`	`arnuld`	`angryuser, Actually I am using Astersk Manager API`
`13:38.33`	`angryuser`	`arnuld, i cant help you then, have you enabled all rights for your client ?`
`13:38.46`	`angryuser`	`in manager.conf`
`13:38.52`	`arnuld`	`angryuser, yes`
`13:38.58`	`*** join/#asterisk n3hxs (n=HAMming@63.68.135.4)`
`13:39.26`	`angryuser`	`arnuld, that's all guess i got`
`13:41.39`	`arnuld`	`oops!`
`13:41.42`	`arnuld`	`thanks anyay`
`13:43.02`	`*** join/#asterisk kchehab (n=CK@212.98.141.199)`
`13:43.16`	`kchehab`	`My scenario is incoming call to asterisk which asterisk in its term will dial it through its trunk .`
`13:43.16`	`kchehab`	`I recognized that Asterisk is sending two invites to My Trunk GW IP as you can see in the debugging below`
`13:43.16`	`kchehab`	`The first is the default and the second when asterisk receives a 200 OK`
`13:43.16`	`kchehab`	`Why Asterisk(B2BUA) is acting like that, and from where I can get the asterisk sip dial call flow`
`13:46.25`	`*** join/#asterisk jmacz (n=jmacz@190.144.75.22)`
`13:47.03`	`mazpe`	`Aiatek: any other ideas?`
`13:48.07`	`*** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk)`
`13:49.00`	`[TK]D-Fender`	`kchehab: Enable SIP debug at * CLI and pastebin a complete failed call attempt from beginning to end.`
`13:49.02`	`[TK]D-Fender`	`~pb`
`13:49.03`	`infobot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/`
`13:54.12`	`*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)`
`13:55.42`	`*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)`
`13:58.15`	`MatBoy`	`when you can hear a "non ok line message" on a asterisk box but no sound during a call ? is sound actually working well than or not ?`
`13:59.14`	`*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)`
`14:01.34`	`Chainsaw`	`MatBoy: DTMF tones and busy/congestion signals might well be carried out of band.`
`14:01.51`	`Chainsaw`	`MatBoy: In which case, no, you can not rely on these tones as an indication that the audio stream setup is working correctly.`
`14:02.41`	`kchehab`	`[TK]D-Fender kindly find my debugging at http://www.binpaste.com/v.php?id=s7qfb`
`14:03.05`	`kchehab`	`[TK]D-Fender its an etherral debug`
`14:03.21`	`kchehab`	`ethereal*`
`14:03.26`	`[TK]D-Fender`	`kchehab: and I told you to get * SIP DEBUG from CLI.`
`14:03.33`	`kchehab`	`ok i will`
`14:03.37`	`[TK]D-Fender`	`kchehab: This does not provide any packet detail`
`14:04.08`	`[TK]D-Fender`	`kchehab: And please use pastebin.com , I don't want to scroll the a textbox the size of a postage-stamp`
`14:04.13`	`[TK]D-Fender`	`through*`
`14:05.43`	`*** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)`
`14:06.27`	`*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)`
`14:08.40`	`*** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk)`
`14:10.40`	`kchehab`	`[TK]D-Fender kindly find it at http://www.binpaste.com/v.php?id=sebja`
`14:10.58`	`kchehab`	`[TK]D-Fender its a sip debug`
`14:12.04`	`[TK]D-Fender`	`kchehab: link is bad`
`14:17.23`	`MatBoy`	`Chainsaw: ok... than I have to look further`
`14:17.26`	`*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)`
`14:17.47`	`jaytee`	`anyone needing a good laugh go to the end of the first or top of the second page of Digg and look for a link regarding Denise Richards. One of the funniest videos I've seen lately.`
`14:19.26`	`[TK]D-Fender`	`jaytee: Seen the "Flamewar" vid on CH?`
`14:19.35`	`*** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173)`
`14:20.56`	`captiancrash`	`when i'm using any fuction that playsback a file (such as Playback, or even entering voicemail) the first second or so is cut off as heard from the other end. normally, i solved this with Wait(1) in my dialplan, but this seems like a bad solution..`
`14:21.06`	`jaytee`	`[TK]D-Fender, nope. CH?`
`14:21.20`	`[TK]D-Fender`	`jaytee: http://www.collegehumor.com/video:1907543`
`14:21.27`	`[TK]D-Fender`	`jaytee: 100% Comedy Gold`
`14:21.39`	`captiancrash`	`is there a better way of inserting a delay before playing back a file? (the problem happens when going into voicemail, joining a conference, using followme, etc..`
`14:21.46`	`alvar`	`hi all`
`14:23.02`	`[TK]D-Fender`	`captiancrash: better than WHAT?`
`14:23.34`	`*** join/#asterisk riddlebox (n=user@159.251.13.3)`
`14:24.23`	`jaytee`	`omg! that is priceless!`
`14:24.24`	`captiancrash`	`[TK]D-Fender, right now, i do something like "8100,1,Wait(1)" before VoiceMail() to solve the voicemail problem. this solution doesn't work other places`
`14:24.43`	`[TK]D-Fender`	`captiancrash: Playback(silence/1) ; or 2, etc`
`14:25.10`	`*** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)`
`14:25.37`	`[TK]D-Fender`	`jaytee: And though she seems a little young for me, I'd bang that chick in the Bustedtees video ad like a screen-door in a hurricane :D`
`14:26.03`	`*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)`
`14:27.29`	`kchehab`	`[TK]D-Fender kindly find it on http://pastebin.com/m6e7f454 and sorry for the Inconvenience`
`14:27.35`	`*** join/#asterisk zeeesh (n=zeeesh@203.215.179.43)`
`14:28.05`	`jaytee`	`[TK]D-Fender, have you seen this one? http://www.collegehumor.com/video:1907543`
`14:28.09`	`captiancrash`	`[TK]D-Fender, when I use the followme app, the same thing happens. The "Press 1 to accept 2 to ignore" starts to playback before i start hearing audio on the phone... i pick up the receiver and it takes about half a second or so before i hear anything.`
`14:28.24`	`captiancrash`	`[TK]D-Fender, i don't see a way to insert the wait before that... am i missing something?`
`14:28.24`	`angryuser`	`hello, i have a person wiilling to play message for his clients, the problem is that he has really many clients, and for the second i dont find any good autodialer soultion to a given volume with a goo administration interface, 3 mil of clients 20 sec message, any ideas ?`
`14:29.00`	`[TK]D-Fender`	`jaytee: .... same vid...`
`14:29.21`	`drmessano`	`HAHAHAH`
`14:29.22`	`*** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de)`
`14:29.39`	`[TK]D-Fender`	`captiancrash: Sorry can't comment on that apps usage. its nothing yuo can't do yourself in pure dialplan anyway`
`14:30.45`	`jaytee`	`[TK]D-Fender, sorry. I kept watching the following vids. guess the number doesn't change`
`14:31.08`	`[TK]D-Fender`	`jaytee: whats the name?`
`14:31.26`	`jaytee`	`it's this one`
`14:31.28`	`jaytee`	`www.collegehumor.com/video:1904510`
`14:31.54`	`jaytee`	`the part at the end is my favorite part`
`14:32.15`	`[TK]D-Fender`	`jaytee: Yup... kinda funny, and somewhat true actually... mind you I've been playing guitar for almost 20 years now :)`
`14:32.19`	`eppigy`	`STACK GUAP`
`14:32.23`	`kchehab`	`[TK]D-Fender did you see it ?`
`14:32.36`	`*** join/#asterisk ThoMe (i=tm@tm.muc.de)`
`14:32.37`	`ThoMe`	`hello`
`14:32.50`	`ThoMe`	`ist the zaptel-dummy-driver in the package of asterisk-1.4.24.1 ?`
`14:36.23`	`ThoMe`	`hello?`
`14:36.49`	`ThoMe`	`dahdi ?`
`14:36.51`	`SuPrSluG`	`um,it's in the zaptel package`
`14:36.59`	`SuPrSluG`	`that too`
`14:37.23`	`ThoMe`	`SuPrSluG: where is the package?`
`14:37.37`	`SuPrSluG`	`asterisk.org`
`14:37.38`	`ThoMe`	`http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/telephony/dahdi-linux/releases/dahdi-linux-2.1.0.4.tar.gz here?`
`14:37.44`	`ThoMe`	`is the renamed in dahdi ?`
`14:38.04`	`[TK]D-Fender`	`~dahdi`
`14:38.05`	`infobot`	`[~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav`
`14:38.09`	`[TK]D-Fender`	`ThoMe: old news`
`14:38.13`	`ThoMe`	`[TK]D-Fender: hello.`
`14:38.17`	`SuPrSluG`	`yes and the tools have an app to change the dialplan instances`
`14:38.26`	`SuPrSluG`	`of zaptel to dahdi`
`14:38.35`	`JerJer`	`who's your dah dee`
`14:38.41`	`ThoMe`	`[TK]D-Fender: for meetme i need dahdi, correct?`
`14:38.49`	`[TK]D-Fender`	`ThoMe: Ye`
`14:38.51`	`[TK]D-Fender`	`s`
`14:38.57`	`ThoMe`	`[TK]D-Fender: i sir! have a nice day!`
`14:39.46`	`*** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-82-2.hag.east.verizon.net)`
`14:40.06`	`SuPrSluG`	`TK, I thought you we're going way old skool w/ the Ye`
`14:41.16`	`[TK]D-Fender`	`SuPrSluG: Going even older-school back to the days when people knew how to spell :)`
`14:41.29`	`kchehab`	`[TK]D-Fender HELLO`
`14:41.32`	`*** part/#asterisk Joe_CoT (n=joecot@ubuntu/member/pdpc.bronze.joeterranova)`
`14:41.33`	`[TK]D-Fender`	`kchehab: Not sure of whats going on in there.`
`14:42.39`	`SuPrSluG`	`Shakespeare Day 23rd April`
`14:43.10`	`kchehab`	`[TK]D-Fender is it normal or not ?`
`14:43.14`	`[TK]D-Fender`	`SuPrSluG: Or as we like to think of it : Happy Ubuntu 9.04 Day :)`
`14:43.24`	`ThoMe`	`works asterisk of a XEN-machine?`
`14:43.29`	`[TK]D-Fender`	`kchehab: Not sure, it looks like the call is bouncing around in there.`
`14:43.51`	`kchehab`	`ok`
`14:43.53`	`kchehab`	`thanks`
`14:46.49`	`*** join/#asterisk Pingu-five (n=o@86.72.21.83)`
`14:48.26`	`*** join/#asterisk Dealer2mogette (n=Dealer2m@16.104.80-79.rev.gaoland.net)`
`14:51.06`	`*** join/#asterisk dethstar (n=Obi-Wan@unaffiliated/dethstar)`
`14:51.11`	`*** join/#asterisk pwebguy (n=pwebguy@200.110.240.130)`
`14:51.30`	`*** join/#asterisk moy (n=moy@74.12.124.89)`
`14:53.57`	`pwebguy`	`Hello all. Can someone point me to a resource for verifying encryption for IAX channels?`
`14:54.34`	`pwebguy`	`I have done packet captures with wireshark, and cannot see any differences`
`14:54.42`	`pwebguy`	`Maybe I am looking for the wrong thing?`
`14:55.23`	`pwebguy`	`I found this . http://www.panoramisk.com/85/iax-trunk-and-voice-ciphering/en/`
`14:56.13`	`pwebguy`	`But definitely do not see the same packets`
`14:57.03`	`*** part/#asterisk gego (n=rick@b238085.customer.hansenet.de)`
`14:57.28`	`haib`	`Can you send smdi over serial with asterisk, or can it only receive? Looking at the smdi.conf it seems like it can only receive, it can't send smdi over serial, correct?`
`14:59.04`	`*** part/#asterisk pwebguy (n=pwebguy@200.110.240.130)`
`15:01.47`	`*** join/#asterisk crevetor (n=antoiner@bureau.ubity.com)`
`15:01.54`	`crevetor`	`Hi everybody`
`15:02.17`	`crevetor`	`quick question : is it possible to specify menuselect choices non-interactively`
`15:02.19`	`crevetor`	`?`
`15:02.46`	`crevetor`	`I would like to select the french sounds for instance but without having to enter menuselect`
`15:04.20`	`pmhaddad`	`there might be an option you can pass to ./configure`
`15:04.23`	`pmhaddad`	`let me check`
`15:04.37`	`crevetor`	`pmhaddad: I didn't see anything in the ==help`
`15:04.38`	`Psychobilly`	`cre yes, these options are saved in menuselect.makeopts file`
`15:04.42`	`Psychobilly`	`creativx *`
`15:05.12`	`jaytee`	`so you could overwrite the existing file with a template of the options you want from another file using a script`
`15:05.48`	`crevetor`	`jaytee: that's what I thought but I was looking for an easier way to do this..`
`15:06.29`	`crevetor`	`./configure option would be great but I didn't see anything in the --help`
`15:06.50`	`jaytee`	`easier? how hard is it to run make menuselect one time, copy the file and then create a two line shell script to overwrite?`
`15:07.04`	`Psychobilly`	`or a sed onliner`
`15:07.14`	`jaytee`	`are you creating some kind of custom distribution?`
`15:07.45`	`pmhaddad`	`crevetor, yeah there is no ./configure option`
`15:08.08`	`pmhaddad`	`i thought it might let you choose the langauge before you got to actually compiling, but apperently not`
`15:08.43`	`crevetor`	`jaytee: not that hard but my build environnment is different from my work environnment`
`15:09.06`	`crevetor`	`jaytee: so the resulting menuselect.makeopts are different`
`15:09.33`	`crevetor`	`jaytee: yes, I'm creating my own packaged version`
`15:10.02`	`jaytee`	`crevetor, is the packaged version meant to match the options for the work environment?`
`15:11.16`	`makafre`	`jaytee: by the way, if you want to keep the same menuselect options between builds you can copy menuselect.makeopts to /etc/asterisk.makeopts and menuselect will take that file as your defaults for future builds`
`15:11.35`	`crevetor`	`jaytee: the packaged version should include everything I need but I don't want to install all the dependencies in my work environnment`
`15:11.51`	`jaytee`	`makafre, didn't know that. that's very handy item for future reference. thanks!`
`15:12.28`	`makafre`	`np, it sure is handy :)`
`15:12.44`	`*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)`
`15:12.55`	`crevetor`	`makafre: from what I've read that's also true with a ~/. file`
`15:13.10`	`makafre`	`crevetor: yes`
`15:13.14`	`jaytee`	`crevetor, since most packaged versions of * are precompiled I don't see where the problem really lies, you compile once with the options you need and that becomes the binary template for the package.`
`15:13.23`	`*** part/#asterisk makafre (n=makafre@modemcable056.198-203-24.mc.videotron.ca)`
`15:14.14`	`crevetor`	`jaytee: The issue is to make sure that the sounds get built when asterisk gets build in the build environnment`
`15:14.47`	`crevetor`	`Anyway I think I have the solution`
`15:15.01`	`jaytee`	`crevetor, I understand that part but what I meant was when you build the "package" is the package precompiled?`
`15:15.32`	`jaytee`	`if you compile * with the sound options you want and then just include /var/lib/asterisk/sounds as part of the package, you're there!`
`15:15.51`	`crevetor`	`jaytee: no, it builds in a build environnment and then it's automatically packaged`
`15:16.17`	`*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)`
`15:16.34`	`jaytee`	`crevetor, and this "automatic" packaging tool? it doesn't allow custom configuration of the package?`
`15:17.20`	`*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)`
`15:17.20`	`crevetor`	`It does, that's not really the issue. The issue is that the build tool is non interactive`
`15:17.20`	`*** join/#asterisk ruben23 (n=AGENT@124.107.3.178)`
`15:17.46`	`crevetor`	`jaytee: FYI I am using pbuilder to create an ubuntu package`
`15:18.42`	`jaytee`	`crevetor, you might ask about options in the Ubuntu forum or if there's a pbuilder forum for ways to customize the build options for the package.`
`15:18.50`	`jaytee`	`most people in here just compile`
`15:19.18`	`crevetor`	`Ok, thanks for the help`
`15:19.37`	`jaytee`	`you welcome, wish I'd had a better answer for ya`
`15:20.05`	`angryuser`	`i am still searching for a good autodial solution ;)`
`15:20.29`	`Dealer2mogette`	`Hello everybody`
`15:20.47`	`angryuser`	`does someone know one ? maybe some kind if .call file generato with database connexion ?`
`15:21.06`	`crevetor`	`Dealer2mogette: huhu I love this tune`
`15:21.23`	`Dealer2mogette`	`i have a question concerning a script i would do but i don't know if i must ask here or on #asterisk-dev`
`15:21.36`	`Dealer2mogette`	`crevetor: you are french ?`
`15:21.51`	`crevetor`	`Dealer2mogette: french but I live in Quebec`
`15:22.14`	`Dealer2mogette`	`ok ^^ i don't know that this song was known in Quebec`
`15:23.23`	`crevetor`	`Dealer2mogette: I've only lived here for a year, I've known this song in France`
`15:25.27`	`Dealer2mogette`	`no answer so i ask here :`
`15:25.29`	`Dealer2mogette`	`I want to do a script which is locate on the client and send a message to the asterisk server. The message is to tell to the server if the client is "online" or "away". Do you know how can i do this ? Actually I have read about Asterisk Manager Interface and Asterisk Gateway Interface but do you know over possibility to do this ? How will you do this ?`
`15:27.05`	`[TK]D-Fender`	`Dealer2mogette: Depends how you want to indicate "away". What do you want to do on the server side?`
`15:27.26`	`[TK]D-Fender`	`Dealer2mogette: How would the server lookup this state? (lets forget about how the client will toggle it for a moment)`
`15:27.54`	`ruben23`	`hi anyone can help setting my asterisk to use public IP....for its eth0 and local IP for it eth1....to serve my Sip client.`
`15:27.56`	`*** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com)`
`15:28.16`	`crevetor`	`Dealer2mogette: implement the hinting part of a sip client`
`15:28.19`	`[TK]D-Fender`	`ruben23: * will use both interfaces jsut fine, there is no "setup"`
`15:29.10`	`Dealer2mogette`	`[TK]D-Fender: ok so clients are a support team ! so when one is available asterisk can send the call to it`
`15:29.18`	`ruben23`	`[TK]D-Fender: ill just add up the IP values on both ethernets then automatically...it will communicate...`
`15:29.31`	`ruben23`	`no NAT setup..or other config`
`15:29.50`	`ruben23`	`my SIP client gateway would be the local IP eth1...am i right..>?`
`15:30.00`	`[TK]D-Fender`	`Dealer2mogette: Yes, well how does * make the DECISION to call?`
`15:30.27`	`*** join/#asterisk LeddyHM (n=NONE@75.63.105.141)`
`15:31.59`	`Dealer2mogette`	`[TK]D-Fender: i'm sorry, i don't understand yout question :/`
`15:33.14`	`[TK]D-Fender`	`Dealer2mogette: You say you want to indicate a "state" of being "available". Where are you having Asterisk CHECK for this?`
`15:33.34`	`casix`	`ruben23: no nat is nedeed you have too legs one from * to sip on public ip and anoter from * to the other sip device on lan side`
`15:33.35`	`*** join/#asterisk clintc (n=clintc@n128-227-53-108.xlate.ufl.edu)`
`15:34.47`	`ruben23`	`casix: i have this problem for a month honestly....can setup a public IP on my asterisk box`
`15:35.22`	`ruben23`	`still now...my asterisk box is in local..behind nat...want to change it into public..`
`15:35.34`	`casix`	`ruben23: bindaddr=0.0.0.0 in your sip.conf`
`15:35.45`	`ruben23`	`yes..i have that`
`15:35.58`	`casix`	`is asterisk listen in both interfaces?`
`15:36.26`	`ruben23`	`casix:what you mean...?`
`15:36.45`	`casix`	`the machine have two interfaces, no?`
`15:36.59`	`ruben23`	`the machine have 2 interface right..`
`15:37.12`	`casix`	`and is asterisk listen in both?`
`15:38.31`	`ruben23`	`casix: i dont have idea...how do i setup to listen on both interface..`
`15:38.50`	`Dealer2mogette`	`[TK]D-Fender: i don't know, it's not me which have configure this. But do you know if with AGI, AMI or another thing, i can "change the state" of a client in the server ?`
`15:38.54`	`casix`	`ruben23: netstat -na`
`15:38.59`	`*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)`
`15:39.03`	`ruben23`	`ok..`
`15:39.11`	`[TK]D-Fender`	`Dealer2mogette: Ok, you seem to lack some key understanding of things.`
`15:39.20`	`Dealer2mogette`	`yes :s`
`15:39.27`	`*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)`
`15:39.32`	`[TK]D-Fender`	`Dealer2mogette: there is no global flag to say that a device is "unabailable". You have to CODE IT in yuor dialplan yourself`
`15:40.15`	`[TK]D-Fender`	`Dealer2mogette: The decision on wheter to try to call a device or not is all dialplan.`
`15:40.21`	`ruben23`	`casix: can i PM you..? if its ok..`
`15:40.36`	`Dealer2mogette`	`ok and how can i do this simply ?`
`15:40.43`	`casix`	`PM?`
`15:41.30`	`[TK]D-Fender`	`Dealer2mogette: go read the book and decide how you want to store this "status" you are going to invent, and look at where you want to check for it in your dialplan.`
`15:41.33`	`[TK]D-Fender`	`~book`
`15:41.33`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`15:41.36`	`proxium`	`Hi, I have this message in asteriskcli "> Channel SIP/1111-0a438390 was never answered." what does it stand for?`
`15:42.01`	`[TK]D-Fender`	`proxium: You placed a call, they didn't answer. What is there to fail to understand here?`
`15:42.01`	`ruben23`	`casix:Private message`
`15:42.04`	`*** join/#asterisk CunningPike (n=CunningP@204.239.10.119)`
`15:42.44`	`casix`	`I prefer not... maybe someon know somthing that I don't no, well maybe not... shure :P`
`15:42.53`	`proxium`	`[TK]D-Fender: But it only rings one time`
`15:43.11`	`[TK]D-Fender`	`ruben23: i can't believe you are still not finished with this. THERE IS NOTHING TO CONFIGURE.`
`15:43.48`	`[TK]D-Fender`	`ruben23: What don't you get? you do "bindaddr=0.0.0.0" and * will listen on all interfaces. this is what is used in just about ever example and is almost certain to already be done in yours.`
`15:43.57`	`[TK]D-Fender`	`ruben23: there is no "configuring".`
`15:44.11`	`[TK]D-Fender`	`ruben23: You are INVENTING work that doesn't exist.`
`15:44.31`	`Dealer2mogette`	`[TK]D-Fender: ok thanks ! but i have read some chapters of this book and i found some interresting things on AGI and AMI. But if i have well understood i can't do this with AGI and with AMI it's possible but all accounts must be "manager" ?`
`15:44.44`	`*** join/#asterisk matrix1233 (n=Administ@196.203.44.3)`
`15:44.53`	`matrix1233`	`hello evry body`
`15:45.08`	`matrix1233`	`how can i spy a channel with asterisk 1.2`
`15:45.09`	`matrix1233`	`??`
`15:45.20`	`matrix1233`	`i have a ZAP channel`
`15:45.22`	`casix`	`ruben23: if this is you netstat -na asterisk is no listening or is not listening in port 5060 (default)`
`15:45.27`	`matrix1233`	`ZAP/g1`
`15:45.42`	`[TK]D-Fender`	`Dealer2mogette: Stop thinking of AMI & AGI as magic. your dialplan makes decisions about how to processes a call. its your job to create it.`
`15:46.22`	`*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)`
`15:47.19`	`ruben23`	`http://pastebin.com/m1cd1a153`
`15:47.43`	`*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`15:47.44`	`*** join/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com)`
`15:48.23`	`matrix1233`	`so`
`15:48.36`	`casix`	`??`
`15:48.38`	`matrix1233`	`spy channel ?? :-$`
`15:49.10`	`ruben23`	`casix: its not listening to the ports 5060..?`
`15:51.24`	`casix`	`matrix1233: http://tinyurl.com/crtj75`
`15:52.13`	`ThoMe`	`emm, i try to load dahdi dummy`
`15:52.14`	`ThoMe`	`vm03:/var/log/asterisk# insmod /lib/modules/2.6.26-1-xen-686/dahdi/dahdi_dummy.ko`
`15:52.17`	`ThoMe`	`insmod: error inserting '/lib/modules/2.6.26-1-xen-686/dahdi/dahdi_dummy.ko': -1 Unknown symbol in module`
`15:52.20`	`ThoMe`	`ideas?`
`15:52.31`	`casix`	`ruben23: http://tinyurl.com/ddesla`
`15:52.57`	`mazpe`	`~pb`
`15:52.57`	`infobot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/`
`15:53.11`	`*** join/#asterisk oej (n=olle@ns.webway.se)`
`15:54.26`	`mazpe`	`need a little help... i'm trying to setup an asterisk server to host trunks for my clients to connecto too this is my configuration:`
`15:54.28`	`mazpe`	`http://pastebin.com/m2cf17eb3`
`15:54.52`	`mazpe`	`does it make any sense the way is configure? or am i missing something`
`15:55.13`	`[TK]D-Fender`	`mazpe: the register has to come BEFORE all of your peer entries`
`15:57.42`	`*** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com)`
`15:58.22`	`mazpe`	`[TK]D-Fender: correct.`
`15:58.30`	`mazpe`	`this is the errors i get`
`15:58.31`	`mazpe`	`http://pastebin.com/m5bb3fada`
`15:59.09`	`*** join/#asterisk bijit (n=chatzill@190.10.115.50)`
`15:59.38`	`bijit`	`where does asterisk look for the Authenticate file?`
`15:59.51`	`ThoMe`	`[TK]D-Fender: emm, works dahdi not with XEN?`
`16:00.27`	`ThoMe`	`No hardware timing source found in /proc/dahdi, loading dahdi_dummy`
`16:00.27`	`ThoMe`	`Running dahdi_cfg: done.`
`16:01.01`	`ronator`	`/etc/asterisk.conf`
`16:01.31`	`ThoMe`	`ronator: ?`
`16:02.06`	`mazpe`	`[TK]D-Fender: am i using the wrong the username?`
`16:02.09`	`[TK]D-Fender`	`mazpe: maybe.. just MAYBE you should make your entry called [user1] <-`
`16:03.50`	`*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)`
`16:05.10`	`[TK]D-Fender`	`bijit: common-sense guess : absolute-path`
`16:05.26`	`bijit`	`:(`
`16:05.48`	`*** part/#asterisk matrix1233 (n=Administ@196.203.44.3)`
`16:06.30`	`*** join/#asterisk Vec (n=Vec@87.74.7.50)`
`16:12.11`	`mazpe`	`[TK]D-Fender: hmm.. i thought that you have an entry name with a different username.`
`16:12.14`	`mazpe`	`[TK]D-Fender: thanks`
`16:12.25`	`mazpe`	`i'm not going to fight with nature :) i'm sure theres a good reason`
`16:12.40`	`[TK]D-Fender`	`mazpe: If you're asking for "Fred", there'd better be a "Fred" on the other side.`
`16:13.02`	`[TK]D-Fender`	`mazpe: For you to think otherwise by default isn't too bright`
`16:13.10`	`ronator`	`ThoMe : wrong console ... ehm, window ^^`
`16:14.15`	`*** join/#asterisk \|Cybex\| (n=John@80.100.126.176)`
`16:15.44`	`ronator`	`one thing is the _extension_ , which could also be a name (e.g. in SIP) ; username is just an option and is for displaying (or blocking ;)`
`16:16.16`	`ronator`	`correct me if i am wrong ^^`
`16:17.55`	`*** join/#asterisk juanIMP (n=Juancho@200.71.41.22)`
`16:18.27`	`*** join/#asterisk af_ (n=getsmart@88-149-240-185.dynamic.ngi.it)`
`16:18.58`	`*** join/#asterisk codefreeze-lap (n=murf@72.21.67.40)`
`16:22.52`	`*** join/#asterisk blkry (n=blkry@64.147.222.130)`
`16:23.11`	`*** join/#asterisk \|Cybex\| (n=John@80.100.126.176)`
`16:24.21`	`*** join/#asterisk blkry (n=blkry@64.147.222.130)`
`16:27.42`	`*** join/#asterisk iratik (n=root@74-84-99-12.client.mchsi.com)`
`16:30.41`	`iratik`	`If i have dial plan which tries a number on several different trunks. I am noticing that if trunk 1 is busy, the call is going to trunk 2. Thats silly, if the called number is busy then the dialplan should just respond by playing a busy signal to the caller using Busy()... just not sure how to implement... or maybe i'm not even on the right track. http://www.pastie.org/455946`
`16:30.44`	`ruben23`	`casix...?`
`16:32.22`	`jameswf`	`wtf?`
`16:34.27`	`[TK]D-Fender`	`iratik: Clearly not even on a track. You aren't checking why it failed in between.`
`16:35.07`	`casix`	`ruben23: ?`
`16:35.35`	`iratik`	`Thank you for letting me know that i'm not even on track.. So i need to check DIALSTATUS and if its busy then play busy... i understand`
`16:36.19`	`[TK]D-Fender`	`iratik: Well if three is a condition to be be checked.... well go check it. that's what DIALSTATUS & GotoIf are for...`
`16:36.25`	`[TK]D-Fender`	`there*`
`16:36.57`	`jameswf`	`1800 1700 of which are newbs users in #ubuntu hurts the head`
`16:37.22`	`*** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net)`
`16:40.13`	`defsdoor`	`anyone here use polycom soundstation - I've got an ip6000 and am completely lost on configuring it`
`16:40.19`	`iratik`	`[TK]D-Fender: Am I more on track? http://www.pastie.org/455946`
`16:45.00`	`_brent_`	`defsdoor: it's a polycom--it's a pain`
`16:45.18`	`_brent_`	`if you don't have any kind of central provisioning, you'll have to use the phone's web UI`
`16:45.25`	`defsdoor`	`yeah - I think I've just found a simple explanation`
`16:45.49`	`defsdoor`	`I have tftp etc.. but done fancy getting involved in the polycoms xml stuff`
`16:45.55`	`defsdoor`	`dont*`
`16:47.10`	`_brent_`	`IMO, polycom needs to step up their game when it comes to user interfaces`
`16:47.53`	`defsdoor`	`I had to bring it home with me from an install yesterday as I hadn't got time to get it working`
`16:48.09`	`defsdoor`	`of course I don't have psu for it or poe injector`
`16:48.27`	`*** join/#asterisk trippssss (n=tripps@66.60.235.100)`
`16:48.29`	`defsdoor`	`taking it to another site to sort out tomorrow`
`16:48.35`	`_brent_`	`at least it's not a 501 that requires their special poe injector powersupply`
`16:48.43`	`_brent_`	`the newer stuff uses regular poe`
`16:48.51`	`defsdoor`	`yeah`
`16:49.09`	`defsdoor`	`I bought a dlink poe thing that /isn't/ poe`
`16:49.29`	`ruben23`	`casix: YES.. my asterisk now is litening to port 5060..`
`16:49.37`	`defsdoor`	`thought it was an injector but it's some proprietary psu down cat5 nonsense`
`16:49.40`	`*** join/#asterisk keebler (n=keebler@h247.235.20.98.dynamic.ip.windstream.net)`
`16:49.47`	`trippssss`	`so still dealing with this not being able to dial global crossing issue from my PRI. solved the e.164 presentation they wanted to see, now they're telling me the call traps show the call going out at 3.1 KHz instead of speech, though all other calls go out to speech. any ideas how this could happen?`
`16:50.41`	`ruben23`	`casix: if ill have an asterisk server using dual ethernet port...i dont need NAT & dhcp if...my client dont need to access to internet...just voice..for asterisk...`
`16:51.09`	`ruben23`	`if the clinet PC needs to access the net then....i should setup a nated & dhcp on my asterisk box..`
`16:51.33`	`trippssss`	`what is the proper coder for speech? g.711u-law?`
`16:54.02`	`casix`	`ruben23: for voice no need nat, dhcp is not the same as nat and it depends who is your network if it have static ip or it use dhcp to get the ip address`
`16:54.48`	`coppice`	`trippssss: what would you expect other than 3.1kHz audio for a voice call?`
`16:54.51`	`ruben23`	`casix: finally i got it....`
`16:55.43`	`*** join/#asterisk ScarEye (n=scareye@12.27.87.66)`
`16:56.07`	`ruben23`	`but on my situation..i got htis type of network: internet===>E1 modem===>asterisk box===>switch====SIP client..(voice and data)`
`16:56.42`	`ruben23`	`so i need to setup NAt...and dhcp on it for the client able to access the net..aside form voice..`
`16:57.04`	`*** part/#asterisk Holos (n=cosmond@209.167.131.35)`
`16:57.05`	`trippssss`	`coppice, call trap by telco says it shouldn't show as 3.1KHz, but "speech"`
`16:57.20`	`trippssss`	`coppice, perhaps their software means that's a data bearing coder`
`16:58.05`	`trippssss`	`coppice, he says many providers "block these 3.1 KHz to stop spam faxers that robo dial"`
`16:59.16`	`casix`	`ruben23: then set it but voice will not use it. internet clients will connect to WAN interface and lan clients will connect to lan interface`
`16:59.28`	`casix`	`theres no need of nat for voice in your network`
`16:59.31`	`casix`	`I have to go`
`16:59.32`	`casix`	`bye`
`17:00.12`	`*** part/#asterisk macli (n=macli@nmc.brc.ubc.ca)`
`17:02.29`	`drmessano`	`~happyclownpbx`
`17:02.30`	`infobot`	`[HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone`
`17:04.30`	`bpgoldsb`	`Why is it that when I execute a macro, my extension changes from the dialed extension (112) to s? Is there any way to prevent this? It seems to have a negative impact on my cdr records`
`17:05.06`	`*** join/#asterisk nullable_type (n=nullable@hq.verbx.net)`
`17:05.26`	`nullable_type`	`Is there anyway i can tell Asterisk to use a different IP for RTP?!`
`17:07.40`	`trippssss`	`so basically it appears my outbound calls to this number are somehow presenting themselves as data/fax calls rather than speech. but I cannot figure out for the life of me why that would be the case`
`17:08.21`	`[TK]D-Fender`	`iratik: 1 down, 3 to go`
`17:08.45`	`nullable_type`	`D-Fender > Is there anyway i can tell Asterisk to use a different IP for RTP?!`
`17:08.48`	`*** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net)`
`17:09.44`	`[TK]D-Fender`	`nullable_type: * doesn't hav split srvices, its a B2BUA, not a proxy`
`17:10.12`	`iratik`	`oh... NOANSWER and CHANUNAVAIL ... and ...`
`17:10.24`	`iratik`	`hangup?`
`17:11.47`	`[TK]D-Fender`	`iratik: You'd almost think this was documented or something :)`
`17:13.17`	`iratik`	`I know its documented... I looked it up. Thank you for your patiennce. If the callee hangs up, there is no dialstatus for that?... I'm probably off track as far as how i think asterisk works if there isn't a dialstatus for that`
`17:14.18`	`[TK]D-Fender`	`iratik: unless you tell * otherwise at the end of your call the channel just dies`
`17:14.33`	`[TK]D-Fender`	`irkDialplan only resumes if you specify the "g" option and the callee hangs up`
`17:15.09`	`iratik`	`congestion is the one i was missing, if i use the g flag... what will dialstatus be if the caller hungup... or will i have to just assume or check hangupcause?`
`17:16.51`	`*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)`
`17:17.29`	`*** join/#asterisk bl4 (n=kim@64.0.29.254.ptr.us.xo.net)`
`17:18.31`	`*** join/#asterisk jsgoecke (n=jsgoecke@c-67-180-102-94.hsd1.ca.comcast.net)`
`17:18.38`	`jsgoecke`	`Hello everyone.`
`17:19.13`	`jsgoecke`	`Has anyone had luck getting MOH & Beeps working with the patched app_conference module published here: http://rubyurl.com/U0wK ?`
`17:19.27`	`iratik`	`Is it possible for asterisk to report a destination trunk as responding as busy... but the call connects anyway?`
`17:19.59`	`jsgoecke`	`I compiled, installed the patched version, use 'conference(myconf\|nh)' and yet no MOH`
`17:20.11`	`*** join/#asterisk MrTelephone (n=test@h697179-171.picriverisp.net)`
`17:20.26`	`*** join/#asterisk Erol_ (n=x@88.235.144.246)`
`17:20.30`	`Erol_`	`hi`
`17:20.40`	`[TK]D-Fender`	`iratik: You don't WANT to continue on a completed call`
`17:21.00`	`MrTelephone`	`do you disable call forward within the client. Can you disable it in asterisk by ignoring 302 moved temporarily messages?`
`17:22.01`	`iratik`	`[TK]D-Fender: thanks for your help btw`
`17:22.16`	`Erol_`	`i am reading a book about asterisk and it says that I should use at least one FXO card even if I dont use pstn for making asterisk have a good timing signal, is that true for todat?`
`17:22.16`	`[TK]D-Fender`	`iratik: You're welcome...`
`17:22.20`	`Erol_`	`today i mean`
`17:22.28`	`iratik`	`[TK]D-Fender: I can't imagine how excruciating it must be to be so helpful with people that honestly feel like they have a clue... but don't`
`17:22.38`	`[TK]D-Fender`	`Erol_: Depends on your volume of calls and how much you need timing`
`17:23.03`	`Erol_`	`[TK]D-Fender: actually it doesnt tell about timing, what is it?`
`17:23.07`	`[TK]D-Fender`	`iratik: Yup, makes you want to stab yourself in the eye with a rusty spork sometimes...`
`17:23.11`	`jsgoecke`	`Erol_ That is if you are using apps, like MeetMe, that need timing`
`17:23.17`	`jsgoecke`	`I use app_conference, and need no timer, works fine`
`17:23.33`	`[TK]D-Fender`	`Erol_: Generally just MeetME and IAX2 Trunk mode`
`17:23.58`	`Erol_`	`and how can this card make a good timing even if you dont use it with a line?!`
`17:24.04`	`[TK]D-Fender`	`jsgoecke>Has anyone had luck getting MOH & Beeps working with the patched app_conference module published here: http://rubyurl.com/U0wK ? <- aaprently "sorta fine"`
`17:24.09`	`[TK]D-Fender`	`apparently*`
`17:24.11`	`[TK]D-Fender`	`gah`
`17:27.15`	`Erol_`	`[TK]D-Fender: does asterisk use this card even if you dont plug a pstn line to it?`
`17:27.40`	`jsgoecke`	`[TK]D-Fender Not sure how to read what you are saying?`
`17:28.04`	`jsgoecke`	`Erol_ Because it gets used as a timing device as opposed to ZTDummy`
`17:28.15`	`[TK]D-Fender`	`jsgoecke: You say "works fine" right after asking about your problems with it :)`
`17:28.31`	`jsgoecke`	`Erol_ http://www.voip-info.org/wiki/view/Asterisk+timer+ztdummy`
`17:28.34`	`[TK]D-Fender`	`Erol_: As a timing source, yes`
`17:28.43`	`jsgoecke`	`app_conference works fine`
`17:28.50`	`MrTelephone`	`Noone here has problems with people forwarding their sip clients to alternate numbers?`
`17:28.52`	`jsgoecke`	`Now just trying to add the patched version, to get MOH, and that is not`
`17:33.29`	`*** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu)`
`17:34.18`	`[TK]D-Fender`	`will have Ubuntu 9.04 server & desktop i368 / AMD64 before the end of the work-day.`
`17:34.21`	`[TK]D-Fender`	`\o/`
`17:34.27`	`[TK]D-Fender`	`i386 even!`
`17:35.10`	`Psychobilly`	`u make baby debian swirl cry`
`17:36.19`	`jblack`	`[TK]D-Fender: About time.`
`17:36.24`	`Erol_`	`[TK]D-Fender: according to that wiki I dont need a zaptel hardware for timing because i can use ztdummy`
`17:36.43`	`jblack`	`I've got another 10 hours or so to finish the jaunty upgrade.`
`17:36.56`	`[TK]D-Fender`	`Erol_: Correct, but YMMV on it`
`17:37.18`	`Erol_`	`[TK]D-Fender: YMMV?`
`17:37.32`	`jblack`	`Your mileage may vary.`
`17:37.34`	`[TK]D-Fender`	`jblack: Tmorrow night I'm going to do a live upgrade on my home desktop from 8.10`
`17:37.54`	`jblack`	`I can top that. ;)`
`17:38.03`	`Erol_`	`what is YMMV?`
`17:38.04`	`jblack`	`In a few days, I'm going to see how jaunty does on an eepc.`
`17:38.16`	`jblack`	`Erol_: Your Mileage May Vary.`
`17:39.05`	`jblack`	`It's a way of saying performance varies according various factors.`
`17:39.43`	`Erol_`	`so better use hardware`
`17:39.44`	`trippssss`	`telco telling me that all my outbound calls display as 3.1 KHz, which again he says many customers' main lines specifically block this to prevent getting robo dialed fax calls to their main number. How do I specifically configure * to send all calls so they show up as speech calls?`
`17:39.57`	`jblack`	`I think ztdummy is fine.`
`17:41.14`	`Pingu-five`	`[TK]D-Fender, are you there ?`
`17:41.23`	`[TK]D-Fender`	`looks around...`
`17:41.28`	`[TK]D-Fender`	`Pingu-five: uuhhhh.... no?`
`17:41.37`	`Pingu-five`	`I knew it`
`17:42.00`	`Pingu-five`	`I'm a shoolmate of the guy who asked stuff about being "away" and all`
`17:42.06`	`Pingu-five`	`um... schoolmate.`
`17:42.20`	`jblack`	`shoolmate was more interesting.`
`17:42.53`	`Pingu-five`	`ah, sorry :/`
`17:43.26`	`jblack`	`It's shokay.`
`17:44.06`	`Pingu-five`	`Anyway, we are trying to find out how to make ppl at the support team join an asterisk queue (="available") and make them leave when needed (= "away" status)`
`17:44.11`	`*** join/#asterisk macros73_ (n=cs@c-71-61-74-104.hsd1.pa.comcast.net)`
`17:44.34`	`jblack`	`ppl? How's he doing?`
`17:44.45`	`Pingu-five`	`Cause, they have to 'manually' join the queue by typing stuff on their phone and they are lazy`
`17:45.09`	`Pingu-five`	`Fine, thx :D`
`17:45.21`	`Pingu-five`	`You have his best regards :D`
`17:45.28`	`jblack`	`The hotdesking stuff I wrote refused to take/receive calls unless someone is logged in. Just gripe that the agent is not logged in.`
`17:45.52`	`jblack`	`But that's not the stuff that's built into *. Perhaps you can check a channel variable.`
`17:46.23`	`jblack`	`if it's about people coming into the work, and doing nothing, the appropriate thing is use logins and logouts as their timecard, and only pay for time they are present.`
`17:46.48`	`*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)`
`17:46.58`	`jblack`	`* = a phone system, not middle management. =)`
`17:47.35`	`Pingu-five`	`What is this hotdesking thing you wrote ?`
`17:48.10`	`jblack`	`Just that. Agents log in, agents log out. phone doesn't work otherwise, appointment are tracked.`
`17:48.36`	`Pingu-five`	`But, agents still have to type stuff on their phones to login/out, right ?`
`17:48.37`	`[TK]D-Fender`	`Pingu-five: And the reason you targeted me directly for this is...?`
`17:48.38`	`jblack`	`there's a rudimentary hotdesking app built into * as well. You might want to take a look at it.`
`17:49.06`	`[TK]D-Fender`	`Pingu-five: Second how/when/why would you log them in when they are too lazy to do it themselves from their own phones?`
`17:49.22`	`jblack`	`Sure. They get to work, they dial 'Hi'. And entered a 3 number pin. At the end of the day, they dial 'Bye', and go home.`
`17:49.47`	`Pingu-five`	`[TK]D-Fender, because you told my schoolmate stuff about lacking key understanding and all. I saw the chatlog whiel coming back home and voila`
`17:49.57`	`jblack`	`To indicate they got an appointment (to qualify for bonuses), after a call, they'd dial 'apt'`
`17:50.20`	`jblack`	`Nothing special. Typical stuff that any typical * admin can whip up in a weekend.`
`17:51.16`	`Pingu-five`	`[TK]D-Fender, because they are too lazy to leave the queue when they are taking a break and such stuff. So they want 'something' to put them in/out of the queue with no fingerwork`
`17:51.38`	`[TK]D-Fender`	`Pingu-five: entirely true. He asked how to tell * remotely that a device should be considered "busy", without the first clue that he is responsible for * looking at any sort of flag you have invented as the basis of deciding whether or not to actually call a device that that given exten might otherwise normally just do`
`17:51.41`	`jblack`	`Pingu-five: Ok. There's a highly technical process to make that problem much easier.`
`17:51.49`	`trippssss`	`is at a loss`
`17:52.03`	`jblack`	`Pingu-five: It's called "If you don't log out, you're gonna join the unemployment ranks"`
`17:52.12`	`[TK]D-Fender`	`Pingu-five: Really? And Asterisk is supposed to be PSYCHIC if they are there and not TELLINg it when they arrive & leave?`
`17:52.46`	`jblack`	`repeats that * is a phone system, not middle management.`
`17:53.06`	`jblack`	`The last thing you want to do with lazy employees is use them as a buffer for customers.`
`17:53.14`	`Pingu-five`	`[TK]D-Fender, umm... psychic ? like mewtwo pokemon ?`
`17:53.49`	`jblack`	`[TK]D-Fender: We could get rich if you and I made a chair with a sensor with a * interface.`
`17:53.50`	`[TK]D-Fender`	`Pingu-five: No, psychic like Jo-Jo Savard`
`17:54.13`	`[TK]D-Fender`	`jblack: I'm too busy running 22v to my coworkers' :p`
`17:54.14`	`Pingu-five`	`Anyway, the 'unemplyement' stuff cannot be used. Well.. on me. But thats all.`
`17:54.16`	`[TK]D-Fender`	`220*`
`17:54.41`	`trippssss`	`is there another call frequency to use than 3.1 KHz?`
`17:54.59`	`KyleK`	`if i was going to start using * would I go with 1.6 or 1.4?`
`17:55.28`	`[TK]D-Fender`	`KyleK: What is your goal?`
`17:55.30`	`Pingu-five`	`Anyway, they said something like "We'd like a program that kicks us out the queue when our screensaver starts or something, and put us back in when the screensaver disappears"`
`17:55.33`	`jblack`	`Pingu-five: Again, for the third time.... if they're not worth treating as an adult human, then wtf are they doing being an employee?`
`17:55.53`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`17:55.54`	`[TK]D-Fender`	`Pingu-five: You have code to detect that and make the decision?`
`17:55.54`	`jblack`	`kylek: Either should be fine.`
`17:56.17`	`jsgoecke`	`Okay, I got notifications working with app_conference using this command Conference(myconf/Snh)`
`17:56.25`	`Pingu-five`	`Detecting the screensaver and all ? Not yet, but thats not the hardest thing.`
`17:56.35`	`jsgoecke`	`I thought valid options for an application were ',' or '\|', does '/' also work or needed for some???`
`17:56.36`	`[TK]D-Fender`	`Pingu-five: Actually, yes, it actually is.`
`17:57.17`	`Pingu-five`	`jblack, I am a student, I dont have my word to say "Do that fking thing if u wanna good marks!"`
`17:57.53`	`jblack`	`Well, if you're not the boss, then what business do you have bossing people around, trying to make them do stuff anyways?`
`17:57.56`	`[TK]D-Fender`	`Pingu-five: Have your monitoring app connect via AMI and set an AstDB key value to change their "state", which you will have to check for in dialplan. If you are using queues instead use an AMI Originate to dial a local channel to do the logout command.`
`17:59.06`	`Pingu-five`	`jblack, their company asked my school to do it. So their sofware needs becomes our (graded) homework.`
`18:00.04`	`*** join/#asterisk mib_ar5eeyq2 (i=43bd1417@gateway/web/ajax/mibbit.com/x-f4dbf49ef514d726)`
`18:00.12`	`Pingu-five`	`Hmm...`
`18:00.21`	`Pingu-five`	`But, the AMI needs a manager account, right ?`
`18:00.29`	`[TK]D-Fender`	`Pingu-five: Correct`
`18:01.07`	`mib_ar5eeyq2`	`Hello all! I am having a problem getting with asterisk and my sangoma FXO card... the problem is, i believe, that trix is trying to use a zap channel too quickly after it has been hung up. i have tried putting a w in the outbound string to delay it, but I beleive there is no dial tone on the line yet, so the numbers are being passed before the line was even t`
`18:01.08`	`Pingu-five`	`So I have to make a manager account for each tech support guy ?`
`18:01.20`	`mib_ar5eeyq2`	`anyone have any ideas to ensure that the zap channel is ready to take a call? like a wait 20s before using channel again, or check for silence first?`
`18:01.27`	`[TK]D-Fender`	`Pingu-five: No.`
`18:01.51`	`Pingu-five`	`So a single account for everyone is fine too ?`
`18:01.57`	`[TK]D-Fender`	`Pingu-five: Yes`
`18:02.07`	`Pingu-five`	`Good news for me`
`18:02.15`	`mazpe`	`I'm getting a "username mismatch" when connecting 2 asterisk servers via multiple sip accounts... here is my config and CLI log: http://pastebin.com/m6440de22`
`18:03.12`	`*** join/#asterisk voxter (n=voxter@76.77.91.250)`
`18:03.15`	`mazpe`	`what i'm trying to accomplish is to use via to route my clients calls via 2 different trunks depending on the call [client1] and [client2]`
`18:03.21`	`KyleK`	`[TK]D-Fender: I'm going to hook an FXO up to * and configure it to work as a answering machine, currently looking at a SPA3102 as hardware`
`18:03.28`	`*** part/#asterisk beek_ (n=klinebl@pdpc/supporter/professional/beek)`
`18:03.31`	`Pingu-five`	`But... if I give AMI acces to every tech support guy, I more or less give them admin powers on the server ... right ?`
`18:03.34`	`*** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu)`
`18:03.41`	`mazpe`	`it had worked earlier with just one sip account.. when i added the second one.. is when all hell broke loose`
`18:03.54`	`[TK]D-Fender`	`KyleK: Save your money and effort and jsut buy a dumb answering machine.`
`18:04.21`	`[TK]D-Fender`	`Pingu-five: You give it to your APP. Do they have access to the source?`
`18:04.34`	`KyleK`	`dumb ones wont email me the messages though`
`18:04.52`	`Pingu-five`	`Well... I dont really know. I'm just a dev slave`
`18:06.16`	`Pingu-five`	`Hmm... I'm gonna put a "W8! Dun give sourcecode!" post-it and i'm fine I think.`
`18:06.22`	`[TK]D-Fender`	`Pingu-five: Who is writing the app?`
`18:06.48`	`Pingu-five`	`Me and the other students of my "team".`
`18:06.48`	`[TK]D-Fender`	`Pingu-five: that'll do`
`18:07.20`	`Pingu-five`	`Cool`
`18:08.34`	`Pingu-five`	`So, when I'm conncted via AMI, I just have to... put them in the queue, and voila. right ?`
`18:08.46`	`*** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com)`
`18:10.37`	`*** join/#asterisk lanning (n=lanning@nat/yahoo/x-5be002010cbf06d6)`
`18:10.40`	`Pingu-five`	`I've seen "QueueAdd" and "QueueRemove" in the list of AMI commands. Should I use that ? Or should I just edit the database containing the queue members ?`
`18:11.08`	`Pingu-five`	`Ive seen things such as "DBget"`
`18:16.01`	`*** part/#asterisk Pingu-five (n=o@86.72.21.83)`
`18:16.01`	`*** join/#asterisk Pingu-five (n=o@86.72.21.83)`
`18:16.47`	`Pingu-five`	`Major lag here >_>`
`18:17.29`	`mazpe`	`The error: "check_auth: username mismatch" seems to go away when i switch the sip trunks from "type=peer" to "type=friend" here is my config as type=peer http://pastebin.com/m6440de22`
`18:18.28`	`Pingu-five`	`So, should I use the db or queueadd ?`
`18:19.22`	`bpgoldsb`	`Whats the CLI command to dump all the information/variables about a specific call/channel?`
`18:26.41`	`mib_ar5eeyq2`	`need help configuring a wait on zap channels`
`18:26.54`	`[TK]D-Fender`	`Pingu-five: depends how you deal with memebers, what ver of *, etc`
`18:27.32`	`[TK]D-Fender`	`KyleK: SPA will be a little tricky, but not a big deal.`
`18:27.50`	`[TK]D-Fender`	`bpgoldsb: "core show channel [channel]"`
`18:28.08`	`bpgoldsb`	`[TK]D-Fender, <3`
`18:28.16`	`Pingu-five`	`[TK]D-Fender, Ok OK. So both ways are possible ? none is pure foolish nonsense ?`
`18:28.33`	`[TK]D-Fender`	`Pingu-five: Depends how you want things to work`
`18:28.54`	`trippssss`	`found out it's my sip gateway that's doing the audio 3.1 xmission rather than speech (mediant 1000)`
`18:29.18`	`trippssss`	`however changing it to speech makes inbound calls die :-\|`
`18:29.49`	`Pingu-five`	`[TK]D-Fender, Anyway, thank you very much for your help.`
`18:34.50`	`*** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix)`
`18:35.42`	`*** join/#asterisk CapriCoRN^80 (i=administ@209.8.41.155)`
`18:36.06`	`dni`	`exten => s,n,Read(Secret,plsenter,10) is the 10, the maximum buffer length ? how can i define for it to read from 3-10 ?`
`18:38.08`	`[TK]D-Fender`	`dni: ther is no minimum.`
`18:38.55`	`[TK]D-Fender`	`dni: check the result and loop it yourself. Or make your own read routine as an IVR`
`18:38.55`	`dni`	`just did it`
`18:38.55`	`dni`	`it reads the data fine`
`18:39.03`	`dni`	`basically im trying to set a variable to contain digits pertaining to the caller id`
`18:39.11`	`dni`	`and then dial out using disa with that caller id`
`18:39.18`	`dni`	`but anything less than 10 digits gets rejected`
`18:39.22`	`dni`	`might be my sip trunk provider tho`
`18:39.49`	`dni`	`but i dont see why,. cuz all they should care about si the number im dialing,. not my caller id`
`18:39.51`	`jplank`	`There is still sip providers who allow spoofing of caller ID?`
`18:40.32`	`dni`	`yea`
`18:40.50`	`jplank`	`they can't be top tier providers thought, right?`
`18:41.01`	`jplank`	`is it just a mom and pop shop that have an asterisk and a PRI?`
`18:41.20`	`dni`	`who would you consider top tier?`
`18:41.39`	`jplank`	`non-itsps`
`18:41.40`	`dni`	`tbh, im not 100% sure on their customer base and how big of a company they are.`
`18:42.43`	`jblack`	`top tier company: Any company so much larger than yours, that they could care less whether or not you're a customer.`
`18:43.08`	`dni`	`heh,. nah these people actually give decent support`
`18:43.12`	`jplank`	`lol`
`18:43.15`	`*** part/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com)`
`18:43.22`	`dni`	`but thats probably cuz they are trying to sign my company on as a customer`
`18:43.23`	`jplank`	`depends on how much you will with them though`
`18:43.50`	`jplank`	`we get XO to bend to our every whim all the time`
`18:44.12`	`dni`	`shit,. its taken at&t 3 months to isntall our Metro-E`
`18:44.20`	`jplank`	`sounds about right`
`18:44.23`	`dni`	`and thats just a point to point`
`18:44.32`	`dni`	`that turnaround seemed ridiculous to me`
`18:44.41`	`jplank`	`our normal metro ethernet installs are around 90 days`
`18:44.52`	`jplank`	`dni: is AT&T the lec for your area?`
`18:44.58`	`dni`	`have you heard of wifi max ? if so what ar eyour thoughts on it ?`
`18:44.59`	`jplank`	`or just another ilec?`
`18:45.06`	`jplank`	`do you mean wimax?`
`18:45.18`	`dni`	`they a lec for business i believe`
`18:45.24`	`dni`	`yea wimax**`
`18:45.30`	`jplank`	`are they the local lec, the RBOC`
`18:45.37`	`jplank`	`wimax is cool`
`18:45.48`	`dni`	`our primary lec is bellsouth`
`18:45.56`	`jplank`	`I met a guy at astricon who's company was doing voice over it and said it worked great`
`18:46.04`	`jplank`	`bellsouth = at&t`
`18:46.26`	`dni`	`err yea i guess you are right,.. but they operate seperately afaik`
`18:46.36`	`dni`	`we get bills from at&t and seperate bills from bellsouth`
`18:46.48`	`*** join/#asterisk oej (n=olle@ns.webway.se)`
`18:46.51`	`jplank`	`I forgot where he said he was from, but he said wimax was the only feasible high speed internet connection where he is though`
`18:47.02`	`*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)`
`18:47.05`	`dni`	`http://slingbroadband.com/ these are the peoples im going to go with`
`18:47.17`	`dni`	`the price is about the same`
`18:47.22`	`dni`	`but the isntall takes only 7 days`
`18:47.23`	`jplank`	`I know they started integrating their networks (at&T and bellsouth) but I don't know how far along they are`
`18:47.33`	`jplank`	`where are you?`
`18:47.37`	`dni`	`south florida`
`18:47.48`	`jplank`	`ahhh`
`18:48.11`	`jplank`	`what like Lauderdale or miami area?`
`18:48.35`	`dni`	`miami to be exact`
`18:48.47`	`dni`	`how about you ?`
`18:49.02`	`jplank`	`I work from SC, but my company is based out of NY`
`18:49.27`	`jplank`	`in an area like miami, why are you looking at wimax?`
`18:49.32`	`jplank`	`T1's are cheap down there`
`18:49.46`	`dni`	`because we wanted the full 10mbs, and at&t takes 3 months`
`18:49.46`	`dni`	`:)`
`18:49.49`	`*** join/#asterisk doug (i=doug@breakout.telerama.com)`
`18:50.07`	`dni`	`not o0nly that,. but we might move office buildings in which case a regular metro e would not be effificient`
`18:50.14`	`jplank`	`have you noticed all the major plays have jumped out of wimax?`
`18:50.15`	`dni`	`with the wimax its just a matter of moving the receiver`
`18:50.19`	`doug`	`what's the trick to let my iax softphone to send parens in the extension to dial?`
`18:50.29`	`doug`	`i.e. rewrite the extension to remove parens`
`18:50.35`	`doug`	`and periods, dashes, etc.`
`18:50.53`	`doug`	`i can't seem to find a "s//" function for extensions.conf`
`18:50.56`	`jplank`	`if you could wait, you are A LOT better off with metro e`
`18:51.02`	`*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)`
`18:51.39`	`dni`	`jplank, how come ?`
`18:52.04`	`dni`	`if uptime is 99.9% and speed is the same`
`18:52.05`	`jplank`	`its a lot more reliable`
`18:52.12`	`jplank`	`latency is lower`
`18:52.17`	`Katty`	`moo`
`18:52.21`	`jplank`	`99.9% uptime my ass`
`18:52.29`	`dni`	`i thought the same as yoi`
`18:52.31`	`dni`	`*you`
`18:52.33`	`Katty`	`moooooooo`
`18:52.50`	`jblack`	`I wouldn't consider 99.9 as good enough for phone.`
`18:53.08`	`jplank`	`I don't think he's doing voice over the wimax`
`18:53.12`	`dni`	`jblack, i have fail over solution`
`18:53.14`	`jplank`	`if he is, run, run away fast`
`18:53.15`	`Katty`	`moo?`
`18:53.19`	`Katty`	`infobot: moo?`
`18:53.20`	`infobot`	`ACTION mooooooooo! I am cow, hear me moo, I weigh twice as much as you. I am cow, eating grass, methane gas comes out my ass`
`18:53.24`	`eppigy`	`hello Katty`
`18:53.25`	`dni`	`jplank, yea i wanted to do data and voice`
`18:53.33`	`Katty`	`eppigy: ello dave`
`18:53.37`	`eppigy`	`ALLO`
`18:53.44`	`Katty`	`allohow'reyou`
`18:53.50`	`jplank`	`dni: theres a reason all the major players left the wimax idea behind`
`18:54.17`	`jplank`	`its a pain in the ass to keep latency low, the network efficient, and still make a profit`
`18:54.19`	`eppigy`	`how's she cuttin`
`18:55.06`	`jblack`	`what's with all that bright light outside....`
`18:56.27`	`Katty`	`it's called sunshine`
`18:56.29`	`Katty`	`you should go get some.`
`18:56.38`	`*** part/#asterisk doug (i=doug@breakout.telerama.com)`
`18:56.48`	`jblack`	`Where? amazon?`
`18:56.51`	`eppigy`	`vitamin d`
`18:56.55`	`Katty`	`not funny`
`18:56.56`	`eppigy`	`nature's prozac`
`18:56.58`	`Katty`	`schooch`
`18:56.59`	`Katty`	`your tial`
`18:57.00`	`Katty`	`outside`
`18:57.05`	`Katty`	`pushes jblack out the door`
`18:57.44`	`jblack`	`No way. It's way too bright out there. And there's some sort of thing going on causing there to be a lot of cars on the street.`
`18:57.49`	`jplank`	`jblack does it bother you that they call me your name all the time?`
`18:57.55`	`*** join/#asterisk mocker (n=kyle@shell.mocker.org)`
`18:57.59`	`eppigy`	`haha`
`18:58.08`	`jblack`	`nah. I'd be lonely if it weren't for my friends.`
`18:58.21`	`mocker`	`Anyone here messed with intergrating Asterisk 1.4 and OCS?`
`18:58.55`	`jblack`	`oh, jplank.. Nah. I don't care.`
`18:59.12`	`jblack`	`at least as long as you don't care if I answer.. :)`
`18:59.41`	`jplank`	`go ahead`
`18:59.52`	`jblack`	`considers sending a complaint to the city about wasting all this electricy on all that sunshine stuff`
`18:59.53`	`jplank`	`they are usually yelling at me anyway :P`
`19:00.01`	`jplank`	`lol`
`19:00.08`	`eppigy`	`i am getting confused`
`19:00.09`	`jblack`	`no problem. people yell at me too.`
`19:01.07`	`jblack`	`as long as we're consistent (perhaps "o'rlly? wtf? dude chill. k thx bai.`
`19:02.22`	`drmessano`	`mocker: You need 1.6`
`19:02.34`	`drmessano`	`mocker: Need SIP TCP.. Shit only works on 1.6`
`19:02.36`	`Qwell`	`seanbright: day 2. not 100% satisfied. probably...80? I'm going with user error for now.`
`19:02.52`	`Qwell`	`well, day 1 really`
`19:03.17`	`drmessano`	`Qwell: You get your electric butt hair shaver?`
`19:03.24`	`drmessano`	`Qwell: Not QUITE doin it?`
`19:04.38`	`eppigy`	`oh he went there`
`19:05.15`	`drmessano`	`Word of advice.. Shave AGAINST the grain`
`19:05.22`	`drmessano`	`Its night and day`
`19:06.56`	`*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)`
`19:07.13`	`drmessano`	`So now that Ubuntu 9.04 is out`
`19:07.18`	`drmessano`	`Who cares?`
`19:07.36`	`[TK]D-Fender`	`drmessano: I do. Got desktop & sever for 386/64`
`19:07.44`	`haib`	`Can you send smdi over serial with asterisk, or can it only receive? Looking at the smdi.conf it seems like it can only receive, it can't send smdi over serial, correct?`
`19:07.54`	`bijit`	`is there a way I can see on CLI the numbers dialed when Authenticate asks? dtmf?`
`19:08.07`	`drmessano`	`[TK]D-Fender: Oh youre one of those`
`19:08.30`	`[TK]D-Fender`	`drmessano: by "those" you mean "people who use it"? If so.. yes`
`19:08.32`	`drmessano`	`Karma Minus Minus [TK]D-Fender`
`19:08.40`	`drmessano`	`Bah, its broken`
`19:08.44`	`[TK]D-Fender`	`drmessano: I only use it for my home desktop mind you`
`19:08.54`	`[TK]D-Fender`	`drmessano: And what part of it is "broken"?`
`19:09.08`	`drmessano`	`The Karma Minus Minus`
`19:09.11`	`drmessano`	`Got nothing`
`19:09.17`	`jplank`	`fender: did they automatically push out that update ;)`
`19:10.02`	`seanbright`	`Qwell: have you smoked otherwise?`
`19:10.26`	`drmessano`	`I prefer distros named after articles of clothing, types of cheese, or intestinal tract worms`
`19:10.34`	`jblack`	`drmessano: I'm downloading.`
`19:11.39`	`eppigy`	`I use ubuntu desktop on my laptop`
`19:11.45`	`drmessano`	`jblack: I can see that.. If you cared enough to spite me you would, if you didnt care enough to spite me, you would`
`19:11.45`	`eppigy`	`its pretty sweet`
`19:11.54`	`drmessano`	`Basically, you would`
`19:11.54`	`eppigy`	`I will never use ubuntu server though`
`19:12.49`	`Qwell`	`seanbright: 2`
`19:12.53`	`drmessano`	`I choose not to use Ubuntu simply because those expecting me to would be surprised and those not expecting me to would be surprised at those expecting me to`
`19:13.20`	`jplank`	`drmessano: is lindows your os of choice?`
`19:13.32`	`seanbright`	`Qwell: you a 1 pack a day guy?`
`19:13.36`	`jblack`	`Yup.`
`19:13.36`	`Qwell`	`seanbright: one last night (~2 hours in) and one this morning (because my first cart "ran out")`
`19:13.38`	`Qwell`	`seanbright: yeah`
`19:14.01`	`drmessano`	`I'm pretty stuck on BarbieLinux`
`19:14.02`	`jblack`	`The only time I care what other people use is when they try to get me to fix it. :)`
`19:14.17`	`seanbright`	`Qwell: well from 20 to 2 is a good start`
`19:14.35`	`jblack`	`Which version of barbielinux? Ken, or dreamhouse?`
`19:14.51`	`jplank`	`lol`
`19:15.12`	`drmessano`	`I keep all my files in the Barbie Mansion, I use the Barbie Corvette Browser.. and I LOOOVE that I can play dress up (AKA, change themes) with a single click`
`19:15.25`	`jplank`	`are you guys really pulling a joke from a 2006 april fools day prank?`
`19:16.01`	`jblack`	`drmessano: Yeah, I'm sure it's absolutely great, with all that complicated math shit stripped out.`
`19:16.03`	`drmessano`	`jblack: I am using BarbieLinux 2.3: Malibu Madness`
`19:16.20`	`jblack`	`ohhhh. pink AND purple. SooOOOooo cute!`
`19:17.31`	`drmessano`	`BarbieLinux is cool, but DO NOT.. I repeat.. DO NOT install Asterisk on it`
`19:17.45`	`drmessano`	`Talk about some bitches running up a phone bill`
`19:17.51`	`*** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67)`
`19:18.26`	`jblack`	`I can't top that.`
`19:19.10`	`jblack`	`Other than to drop just a little bit of PFE....`
`19:19.15`	`jblack`	`"Hello Kitty Linux"`
`19:19.27`	`andrebarbosa`	`anyone notice that if you have dynamic features set, the dtmf's to outboudn IVR's are blocked waiting for a timeout that will never happen`
`19:19.27`	`andrebarbosa`	`:s`
`19:19.29`	`drmessano`	`I would run that`
`19:19.41`	`jblack`	`Pink background, lots of oversized flowers... And the mouse icon.. well, that would be kitty of course.`
`19:20.02`	`jblack`	`Comes with a free gadget waif...`
`19:20.31`	`jblack`	`andrebarbosa: I've never noticed a problem with features, other than the default timeout being far too short.`
`19:21.35`	`andrebarbosa`	`the timeout you can configure`
`19:21.56`	`jblack`	`a default is, by definition, unconfigured. =)`
`19:22.02`	`andrebarbosa`	`ya sure ;)`
`19:22.12`	`andrebarbosa`	`but that is not a big problem imho`
`19:22.18`	`andrebarbosa`	`but the stange thing`
`19:22.39`	`andrebarbosa`	`is that in 1.4.24 the dynamic features blocks outbound dtmf's`
`19:22.41`	`andrebarbosa`	`:S`
`19:22.57`	`andrebarbosa`	`i have a dyn feature with code #79`
`19:23.00`	`jblack`	`I don't agree.`
`19:23.19`	`andrebarbosa`	`and to send the "#" dtmfs to outbound line i need to press twice the #`
`19:23.32`	`jblack`	`correct.`
`19:23.44`	`andrebarbosa`	`if I disable the dyn features, it works fine`
`19:24.03`	`jblack`	`what would you expect? * to read your mind and know when you mean a feature, and when you don't?`
`19:24.37`	`andrebarbosa`	`no`
`19:24.41`	`andrebarbosa`	`I press #`
`19:24.49`	`andrebarbosa`	`then the digitfeaturetimeout expires`
`19:24.56`	`andrebarbosa`	`and the # is sent to the bridge channel`
`19:24.57`	`andrebarbosa`	`:S`
`19:25.38`	`jblack`	`that's what you want, or what you see? (that's the intended behaviour)`
`19:25.53`	`jblack`	`if a feature isn't matched within timeout, shove it out.`
`19:26.32`	`andrebarbosa`	`yea`
`19:26.36`	`andrebarbosa`	`but is not happening`
`19:26.42`	`andrebarbosa`	`asterisk blocks and never timeouts`
`19:26.48`	`andrebarbosa`	`for dynamic features`
`19:27.01`	`jblack`	`hmm. I see. No idea.`
`19:27.17`	`andrebarbosa`	`for the default features, like blind transfer and atx transfer`
`19:27.19`	`jblack`	`perhaps you're matching a feature listed lower.`
`19:27.19`	`andrebarbosa`	`it works fine`
`19:29.10`	`andrebarbosa`	`I was trying to fix it, with no luck till now`
`19:29.11`	`andrebarbosa`	`:(`
`19:30.35`	`dni`	`has anyone seen this WARNING,. i see it in the console every time i hit a dtmf key. [Apr 23 15:28:25] WARNING[7868]: chan_sip.c:11427 handle_request_info: Unable to parse INFO message from CXC-388-69b571f0-3c81aac-13c4-49f0c085-c64514b2-5e06141f@4.68.250.148. Content Ø¤¢·?è®`
`19:31.21`	`andrebarbosa`	`looks like your phone\sip device is sending a broken SIP INFO packet`
`19:31.44`	`andrebarbosa`	`try to change it to rfc2833 dtmf mode`
`19:31.58`	`dni`	`hmm ,. the calls go's thru fine,. and it recognizes the dtmf i enter,.`
`19:32.04`	`dni`	`ok ill try that`
`19:35.06`	`dni`	`for dtmf method all i got is : inband, AVT, info, auto, inband+info, avt+info ,. . i have it set at auto`
`19:35.24`	`dni`	`those are the only dtmf options i hav eon mysip phone`
`19:35.27`	`andrebarbosa`	`AVT`
`19:35.33`	`dni`	`ok`
`19:35.34`	`andrebarbosa`	`and set asterisk to rfc2833`
`19:36.32`	`dni`	`dtmfmode=inband in sip.conf correct?`
`19:36.42`	`dni`	`under [general] ?`
`19:37.44`	`oej`	`info can be other tings than DTMF.`
`19:38.23`	`andrebarbosa`	`dtmfmode=rfc2833`
`19:38.38`	`dni`	`i think its just particular to the DISA i have setup`
`19:38.44`	`andrebarbosa`	`you are right oej`
`19:39.04`	`dni`	`cuz when i initially dial a phone i dont se those warnings,. its only when iut coems to my disa context`
`19:39.14`	`*** join/#asterisk Schreiber1337 (n=SCHREIBE@216.169.165.178)`
`19:39.19`	`*** join/#asterisk profxavier (n=MyNick@unaffiliated/neverblue)`
`19:40.16`	`*** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net)`
`19:40.25`	`*** join/#asterisk pwebguy (n=pwebguy@190.149.24.99)`
`19:40.45`	`dni`	`andrebarbosa, thanks`
`19:40.48`	`dni`	`that seemed to be it`
`19:40.57`	`profxavier`	`Asterisk call queues. When selected 'fewest calls', and I have a fallback person (if no one answers, they will get the call), does this try one person in the call (fewest calls), then go to the fallback person, or will it go through the entire group?`
`19:41.01`	`oej`	`dni: turn on SIP debug and capture a full INFO packet and place it in pastebin so I can study it, thanks`
`19:41.05`	`andrebarbosa`	`no prob`
`19:41.19`	`andrebarbosa`	`oej I have a question for you`
`19:41.27`	`oej`	`Shoot`
`19:41.32`	`andrebarbosa`	`a few days ago, i made a small patch for * 1.4`
`19:41.47`	`haib`	`Is it possible to send out of band smdi through serial, or can asterisk only receive it?`
`19:41.52`	`andrebarbosa`	`to support linksys g729 conferences`
`19:42.26`	`[TK]D-Fender`	`profxavier: what is a "fallback person"?`
`19:42.27`	`Schreiber1337`	`Can someone help me with a extensions.conf that won't load?`
`19:42.31`	`bpgoldsb`	`Is there any way to create a visual map of my dialplan from the dialplan code itself?`
`19:42.45`	`profxavier`	`Fender, I explained in the () brackets`
`19:42.59`	`andrebarbosa`	`it's working for a few weeks with success, if you don't mind to take a look at the patch, it maybe useful for someone else`
`19:43.06`	`[TK]D-Fender`	`profxavier: says nothing about HOW you do this`
`19:43.35`	`profxavier`	`Fender how about we just discuss how fewestcalls works, rather than my methodology?`
`19:44.43`	`[TK]D-Fender`	`profxavier: You mean regardless of any possibility that your method may break the functionality we are describing?`
`19:44.52`	`Schreiber1337`	`My extensions.ael is loading but my extensions.conf is not... please help!`
`19:45.03`	`profxavier`	`Fender, please no long assist with my question, thanks`
`19:45.23`	`pwebguy`	`Hello, all - I am tryting to verify the media encryption provided by IAX using wireshark but encrypted and unencrypted calls all look the same. Can anyone point me to a resource that will describe what I should be looking for?`
`19:46.05`	`[TK]D-Fender`	`profxavier: fewest calls should bang that 1 person as long as they are available until they answer. If you cheat the process somehow, thats another amtter`
`19:46.19`	`*** join/#asterisk Beighto (n=chatzill@120.155-62-69.ftth.swbr.surewest.net)`
`19:46.59`	`profxavier`	`anyone else ?`
`19:47.19`	`*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)`
`19:47.42`	`dni`	`pwebguy, are you dumping ascii ?`
`19:47.46`	`dni`	`tcpdump -s0 -A`
`19:48.06`	`pwebguy`	`no, just reading the logs provided by wireshark`
`19:48.14`	`*** join/#asterisk blebleble (n=godie@216.222.194.7)`
`19:48.25`	`pwebguy`	`better to do tcpdump?`
`19:48.42`	`blebleble`	`is there any decent graphical tools to search /var/spool/asterisk/monitor for queue recordings then looking up who answered the call via the uniqueid and the asteriskcdrdb?`
`19:48.46`	`pwebguy`	`Was using this page as my guide: http://www.panoramisk.com/85/iax-trunk-and-voice-ciphering/en/`
`19:49.20`	`Beighto`	`Does anybody have any recommendations for SIP termination under $.01 for US?`
`19:49.37`	`dni`	`i know tcpdump syntax better than wireshark`
`19:49.42`	`dni`	`thats why i gave that as an example`
`19:49.57`	`dni`	`tcpdump -s0 -A port whatever`
`19:50.06`	`dni`	`you should see plain text if its not encrypted`
`19:50.21`	`pwebguy`	`got it; thanks - I have not used tcpdump, but will definitely give it a try.`
`19:50.44`	`[TK]D-Fender`	`profxavier: "; fewestcalls - ring the one with fewest completed calls from this queue" <- from the sample directions`
`19:50.47`	`pwebguy`	`what will show if it is encrypted?`
`19:50.55`	`[TK]D-Fender`	`profxavier: Sure looks like it means what it says.`
`19:51.02`	`dni`	`pwebguy, encrypted data`
`19:51.19`	`*** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net)`
`19:51.40`	`pwebguy`	`ok - I will give it a try and see what I can find out. Thank you dni`
`19:51.47`	`dni`	`no prob`
`19:52.42`	`profxavier`	`Fender, i asked nicely`
`19:53.04`	`*** join/#asterisk n3hxs (n=HAMming@63.68.135.4)`
`19:53.08`	`[TK]D-Fender`	`profxavier: and I answered nicely.`
`19:53.21`	`profxavier`	`i wasn't referring to my question`
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`19:53.29`	`profxavier`	`I was hoping you could just drop it`
`19:57.37`	`Schreiber1337`	`[TK]D-Fender: Can I bug you for just a minute?`
`19:58.19`	`[TK]D-Fender`	`Schreiber1337: 1,5,& 20 minute blocks are available :)`
`19:59.31`	`Schreiber1337`	`[TK]D-Fender: Cool... I just put a new box in place running 1.6.0.9... coppied my configs over from the old box running 1.6.0.1... and now extensions.conf is not loading ... just extensions.ael`
`20:00.15`	`[TK]D-Fender`	`Schreiber1337: Go look at your * startup and see what it spews out`
`20:00.43`	`[TK]D-Fender`	`Schreiber1337: check your file permissions, spelling, etc`
`20:00.57`	`[TK]D-Fender`	`Schreiber1337: try to do a reload. Check which modules have loaded.`
`20:01.12`	`oej`	`andrebarbos: Tell me why we need special patches to support it - please. Do you mean local conferencing on the Linksys?`
`20:01.15`	`*** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43)`
`20:01.46`	`*** join/#asterisk SomethingIsodd (n=dan@72.172.174.69)`
`20:01.56`	`SomethingIsodd`	`hello anyone happen to know of any irc channels that deal with Gnugk?`
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`20:03.11`	`andrebarbosa`	`oej: linksys phones don't support two simulatenous g729 calls, so when you hit the conference button, it will send a re-invite to asterisk with only g711u`
`20:03.41`	`andrebarbosa`	`the asterisk default behaviour is to reply with a 200OK with g729, which causes the linksys phones to BYE the call`
`20:03.57`	`oej`	`That's a bug.`
`20:04.08`	`oej`	`Please open a bug in the tracker and upload your patch, so we can check it.`
`20:04.15`	`andrebarbosa`	`ok`
`20:04.18`	`andrebarbosa`	`:)`
`20:04.25`	`oej`	`Also, it would be very helpful if you could take a debug file with "sip debug" of the transaction.`
`20:04.40`	`andrebarbosa`	`I've fix it by sending the jointcapabilty in the 200OK`
`20:05.19`	`andrebarbosa`	`but I have also to send a reinvite to the other peer to change the call to g711u`
`20:05.29`	`Mersault`	`I'm hoping someone here has some insight for me. I have a Sangoma A200 card and I'm getting loud static, but only on the asterisk side, PSTN side doesn't hear it. I've played with the rx and tx gains, but they don't help.`
`20:05.34`	`andrebarbosa`	`so both peers talk g711u after the re.invite is finnished`
`20:05.40`	`andrebarbosa`	`and it works well`
`20:05.57`	`Schreiber1337`	`[TK]D-Fender: http://www.spectrumcontrol.com/zero/pastbin/reolad.txt`
`20:05.58`	`Schreiber1337`	`Doesn't look like pbx_config.so did anything...`
`20:06.09`	`Mersault`	`I've tried dahdi_tool on the channels, and there's lots of signal on the rx side, even for channels that aren't hooked into the PSTN`
`20:07.56`	`[TK]D-Fender`	`Schreiber1337: go see if its there. check extensions.conf , try loading the module manually. Look at your modules.conf to see if its specifically expluded, etc`
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`20:19.25`	`Katty`	`i'm restless.`
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`20:21.21`	`seanbright`	`i'm sean.`
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`20:28.05`	`wpbrown504`	`the young and the restless`
`20:32.52`	`[TK]D-Fender`	`checkout time, BBIAB`
`20:33.04`	`wpbrown504`	`I have a question. It is probably something simple that I have over looked. I have a 4 port Digium FXO card. Dahdi see's it. It is properly loaded. I have edited the dial plan and extensions.conf as per the oriley book. Dahdi tools sees the card with no errors. When I call the thing it doesn't answer. Anyone see something obvious that I might have missed?`
`20:33.41`	`wpbrown504`	`Compiled libpri,dahdi, and Asterisk in that order.`
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`20:35.24`	`eppigy`	`Katty: COME PLAY`
`20:38.13`	`Pan3D`	`throws a kickball into the channel`
`20:44.10`	`beek`	`wpbrown504: What's happening at the console? asterisk -vvvvr`
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`20:51.38`	`wpbrown504`	`beek: she isn't showing anything.`
`20:51.48`	`wpbrown504`	`when I dial it.`
`20:53.37`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)`
`20:55.16`	`wpbrown504`	`wb Fender`
`20:56.45`	`[TK]D-Fender`	`wpbrown504: ty`
`20:56.55`	`joesuffceren`	`running asterisk 1.4.22. I'm having a weird issue where my users will answer calls but the calling party still hears ringing and the two parties are not "connected" for 3-5 seconds after my user answers. It's not every call, and it happens both on my cisco 7940s that I've been using for years and my xlite soft phone, so I know it's not an endpoint-specific issue. This system has been in...`
`20:56.56`	`joesuffceren`	`...production for 6+ months without having this issue. Only recent change was the addition of CDR logging to local mysql. Before I was logging CDR to remote postgres. Now, I'm logging to both locations.`
`21:06.19`	`*** join/#asterisk xpot (n=james@70.91.210.233)`
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`21:12.08`	`wpbrown504`	`I have a question. It is probably something simple that I have over looked. I have a 4 port Digium FXO card. Dahdi see's it. It is properly loaded. I have edited the dial plan and extensions.conf as per the oriley book. Dahdi tools sees the card with no errors. When I call the thing it doesn't answer. Anyone see something obvious that I might have missed?`
`21:12.27`	`wpbrown504`	`Compiled libpri,dahdi, and Asterisk in that order.`
`21:12.56`	`wpbrown504`	`asterisk -vvvr shows no activity while dialing`
`21:14.42`	`[TK]D-Fender`	`wpbrown504: Go prove the module is loaded`
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`21:15.11`	`[TK]D-Fender`	`wpbrown504: "dahdi show channels" , "dahdhi show status" , etc`
`21:16.12`	`wpbrown504`	`dahdi show channels is blank`
`21:16.31`	`wpbrown504`	`dahdi show status show the wildcard`
`21:17.17`	`[TK]D-Fender`	`wpbrown504: Then you haven't defined any channels`
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`21:32.30`	`*** join/#asterisk ambush276 (n=ambush27@ip70-181-112-218.oc.oc.cox.net)`
`21:32.34`	`ambush276`	`hey guys`
`21:32.36`	`ambush276`	`i have a question.`
`21:33.20`	`vjr`	`shoot`
`21:33.23`	`ambush276`	`ok basically i have my SIP setup so that i get incomming calls great and goes right to my PBX system... my outgoign calls just dont work. I am using the same SIP provider for outgoing calls, but how do i setup so that lets say ext. "12" is my cell phone, how do i get it to relay to my cell phone?`
`21:34.26`	`vjr`	`is your * box behind a nat?`
`21:34.55`	`ambush276`	`...`
`21:35.08`	`ambush276`	`it might sound dumb but im not sure`
`21:35.16`	`ambush276`	`i mean i have a router setup but its on DMZ`
`21:35.18`	`ambush276`	`for that IP`
`21:35.36`	`KyleK`	`so whats the exact problem, you're supposed to be able to dial 12 and get your cellphone?`
`21:35.53`	`*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)`
`21:35.57`	`ambush276`	`well im not sure how to set that up`
`21:36.05`	`ambush276`	`like i can dial like SIP/100`
`21:36.08`	`ambush276`	`to go to extension 100`
`21:36.10`	`ambush276`	`but that is just local`
`21:36.17`	`vjr`	`ambush276: You have to be able to dial out first from the * box to reach your cell phone.`
`21:36.18`	`ambush276`	`how do i setup so that i can send out t`
`21:36.26`	`ambush276`	`correct..`
`21:36.29`	`ambush276`	`that is my question...`
`21:36.31`	`ambush276`	`how?`
`21:36.43`	`vjr`	`can access anything to the net from your * box?`
`21:36.56`	`ambush276`	`what do you mean?`
`21:36.58`	`ambush276`	`i mean i can dial my number..`
`21:37.07`	`ambush276`	`from my house phone the asterisk number that is hooked up to the SIP line`
`21:37.13`	`ambush276`	`and get to my menu system`
`21:37.22`	`ambush276`	`i just dont know how to setup outgoing calls`
`21:37.57`	`vjr`	`ambush276: can you log on to your * box in the dmz and ping to the outside world or get a web site using w3m or whatever?`
`21:38.19`	`ambush276`	`yes.`
`21:38.21`	`ambush276`	`i can ping`
`21:38.22`	`ambush276`	`i mean`
`21:38.24`	`ambush276`	`i can call the box`
`21:38.26`	`ambush276`	`with my cell phone`
`21:38.28`	`ambush276`	`fore xample`
`21:38.30`	`ambush276`	`and get to the menu system`
`21:38.39`	`ambush276`	`and its on a linux machine`
`21:38.39`	`KyleK`	`vjr: I dont think he has set up any dialing rules for dailing out of *`
`21:38.50`	`ambush276`	`yea`
`21:38.54`	`ambush276`	`kyleK is right`
`21:39.00`	`ambush276`	`im not sure what to do..`
`21:39.27`	`EmleyMoor`	`ambush276: Are you trying to dial from * or from a phone connected to it?`
`21:42.24`	`*** join/#asterisk pwebguy (n=pwebguy@190.149.24.99)`
`21:42.29`	`vjr`	`ambush276: well look at the example configs and there's a free asterisk book you can download.`
`21:42.41`	`ambush276`	`from asterisk`
`21:42.50`	`ambush276`	`i tried`
`21:42.56`	`ambush276`	`but im confused do i need to make a Trunk?`
`21:43.08`	`ambush276`	`cause i found out that like the dial(SIP/__))`
`21:43.30`	`ambush276`	`but that is just to signify the SIP? or what, im not sure in the context of what im supposed to do to have it call an outbound line?`
`21:44.54`	`EmleyMoor`	`ambush276: Do you have a calling account with a provider? Do they offer any advice on configuring * for it?`
`21:45.56`	`vjr`	`ambush276: you do something like exten => _9NXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) in extensions.conf`
`21:46.15`	`ambush276`	`i do have a calling account with a provider (SIP) and no instructions for asterisik`
`21:46.24`	`ambush276`	`right but im not sure wht the TRUNK is for?`
`21:46.45`	`ambush276`	`i knw it sounds nubbish but i cant seem to figure out what to do w/ it`
`21:46.50`	`mazpe`	`hmm.. my codec_g729a.so doesnt seem to loading the license at all. Any ideas what could be wrong?`
`21:47.03`	`mazpe`	`here is what shows on the log and the actual files`
`21:47.04`	`mazpe`	`http://pastebin.com/m7a95f895`
`21:47.48`	`vjr`	`ambush276: TRUNK in my case is set to Zap/3 which is an FXO port wired to a vonage adapter.`
`21:47.49`	`*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk)`
`21:47.51`	`EmleyMoor`	`ambush276: If you want to call a PSTN destination via a SIP account that is what the trunk is for`
`21:48.30`	`EmleyMoor`	`Or yes - Zap/DAHDI FXO if you have them`
`21:49.15`	`ambush276`	`ok but vjr that is for a vonage adapter.. my SIP is through the net there is no modem attachment, i want to dial out through teh SIP,, but like setting up a trunk.. ?`
`21:49.18`	`vjr`	`ambush276: in the case of a software setup TRUNK might be set to something like SIP/vonage where vonage is a context in sip.conf`
`21:49.22`	`Schreiber1337`	`Hello.... does anyone know what this means... I can't find it anywhere on the net...`
`21:49.22`	`Schreiber1337`	`<PROTECTED>`
`21:49.49`	`ambush276`	`trunk might be SIP/provider and then that is the same info for register => in sip?`
`21:49.53`	`ambush276`	`not really sure what to do about that?`
`21:49.58`	`vjr`	`ambush276: you really must study the example configs or the free asterisk book to understand how it works. good luck.`
`21:51.25`	`vjr`	`ambush276: substitute SIP/context for TRUNK. try it.`
`21:52.10`	`ambush276`	`ok so _9NXXXXXX,2,Dial(${sip/provider}/${EXTEN:${TRUNKMSD}})`
`21:52.23`	`ambush276`	`im still not understand like where do i setup a TRunk and or what do i set it up to?`
`21:52.24`	`EmleyMoor`	`ambush276: Not quite`
`21:52.59`	`Schreiber1337`	`Qwell: Could I bug you for a minute?`
`21:53.36`	`EmleyMoor`	`_9NXXXXXX,2,Dial(SIP/provider/${EXTEN:${TRUNKMSD}}) , where provider is the name of the context in sip.conf where you have entered the details`
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`21:58.00`	`Qwell`	`Schreiber1337: about?`
`21:58.23`	`bpgoldsb`	`Using AEL, I have 2 variables. The first is GROUP_SALES=SIP/100&SIP/101, the second is GROUP_SUPPORT=SIP/200&SIP/201. I want to make a GROUP_ALL that has both GROUP_SALES and GROUP_SUPPORT. When I do GROUP_ALL=${GROUP_SALES}&${GROUP_SUPPORT}, asterisk doesn't expand the variables. Any idea what I'm doing wrong?`
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`21:59.56`	`Schreiber1337`	`Qwell: I keep getting " NOTICE[5538]: utils.c:967 ast_wait_for_output: Timed out trying to write" in my asterisk log... I can't find anything about it on the net or in the forums..`
`22:00.19`	`ambush276`	`ok what is the trunk msd`
`22:00.25`	`*** part/#asterisk iratik (n=root@74-84-99-12.client.mchsi.com)`
`22:00.32`	`Schreiber1337`	`Qwell: Does it have something to do with the AGI?`
`22:00.36`	`Qwell`	`Schreiber1337: so why ask me?`
`22:00.48`	`jaytee`	`cuz you know everything`
`22:00.50`	`rOfLz`	`hello there`
`22:01.35`	`Qwell`	`but yes, fix your AGI`
`22:01.35`	`rOfLz`	`I wannt remove and add prefix for my outgoing calls`
`22:02.08`	`Schreiber1337`	`Qwell: Because you are all knowing... can you maybe point me somewhere to fix my AGI...`
`22:02.17`	`*** join/#asterisk denon (i=denon@synapse.subneural.net)`
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`22:03.54`	`Schreiber1337`	`Qwell: or where to look.`
`22:04.28`	`ambush276`	`also`
`22:04.30`	`ambush276`	`so that extension`
`22:04.36`	`ambush276`	`_9NXXXXXX,2,Dial(${sip/provider}/${EXTEN:${TRUNKMSD}})`
`22:04.39`	`vjr`	`ambush276: substitute 1 for ${TRUNKMSD}`
`22:04.40`	`ambush276`	`how do i set that as an extension?`
`22:05.13`	`ambush276`	`1 is for w/e extension i want`
`22:05.23`	`ambush276`	`EXTEN:17 is for extension 17`
`22:05.47`	`[TK]D-Fender`	`${sip/provider} <- very wrong`
`22:05.55`	`vjr`	`ambush276: in extensions.conf create a context called [outgoing] and put that extension in it. then put a context=outgoing in the entry for the sip phone.`
`22:06.02`	`[TK]D-Fender`	`shouldn't even bother with the variable`
`22:06.15`	`ambush276`	`ok TK what should i do then`
`22:06.20`	`ambush276`	`in extensions or SIP vjr?`
`22:06.34`	`vjr`	`ambush276: then you should be able to call your cell phone if it's a local call.`
`22:07.36`	`rOfLz`	`can any 1 tell me how can I remove 91 and add 0 for my outgoing calls == exten => 91\|X.,1,Dial(SIP/${EXTEN}@1234)`
`22:07.51`	`vjr`	`ambush276: an dial 9 first`
`22:07.59`	`ambush276`	`..`
`22:08.01`	`ambush276`	`ok`
`22:08.03`	`ambush276`	`its not like tha`
`22:08.07`	`ambush276`	`like if im calling from my HOUse phone`
`22:08.10`	`ambush276`	`to my asterisk PBX`
`22:08.17`	`ambush276`	`and i want extension 12 on that PBX to be my cellphone....`
`22:08.21`	`ambush276`	`(not from SIP phone)`
`22:09.09`	`vjr`	`ambush276: baby steps first ;). Again read the example configs and the free asterisk book. Go to go.`
`22:09.40`	`rOfLz`	`can any 1 tell me how can I remove 91 and add 0 for my outgoing calls == exten => 91\|X.,1,Dial(SIP/${EXTEN}@1234)`
`22:10.26`	`ambush276`	`ok id ont have SIP phone configd. yet`
`22:10.30`	`ambush276`	`im jsut setting this system up.`
`22:10.39`	`ambush276`	`so under extensions`
`22:10.41`	`ambush276`	`make a contect`
`22:10.51`	`ambush276`	`[outoing] then context=outgoing`
`22:11.04`	`ambush276`	`then under the dial plan in my extension (called sp331)`
`22:11.15`	`ambush276`	`[sp331] then menu system, then my extensions`
`22:11.26`	`ambush276`	`if i want extension 17 to be my cell phone... what is the line for that?`
`22:11.38`	`ambush276`	`>_9NXXXXXX,2,Dial(${sip/provider}/${EXTEN:${17}})`
`22:14.38`	`rOfLz`	`can any 1 tell me how can I remove 91 and add 0 for my outgoing calls == exten => 91\|X.,1,Dial(SIP/${EXTEN}@1234)`
`22:14.56`	`joesuffceren`	`any ideas to improve mysql CDR logging performance?`
`22:19.24`	`*** join/#asterisk f0ner00t (i=f0ner00t@c-67-187-154-111.hsd1.ca.comcast.net)`
`22:19.47`	`f0ner00t`	`Why would I be getting Sip/2.0 404 Not Found on my sip debug?`
`22:22.32`	`*** join/#asterisk juanIMP (n=Juancho@200.71.41.22)`
`22:24.50`	`*** part/#asterisk pwebguy (n=pwebguy@190.149.24.99)`
`22:26.52`	`rOfLz`	`how can I remove 91 and add 0 for my outgoing calls == exten => 91\|X.,1,Dial(SIP/${EXTEN}@1234)`
`22:33.52`	`Qwell`	`~book`
`22:33.53`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`22:34.10`	`Qwell`	`Start reading.`
`22:37.55`	`f0ner00t`	`Qwell : How do I set up auth on my incomming again.`
`22:42.44`	`chiwawa_42`	`I have a SIP 423 error on my SIP trunk (interval too brief). I've added "defaultexpirey=1800" in my trunk's context in sip.conf. What else may solve this issue ? Can it be NAT related ? As the message appears every 30sec on asterisk" CLI, it looks to me like the defaultexpirey isn't working, how to fix that ?`
`22:45.24`	`*** join/#asterisk ScriptFanix (i=vincent@cl-54.mrs-01.fr.sixxs.net)`
`22:47.03`	`*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)`
`23:03.26`	`*** join/#asterisk Aiatek (n=munoz@190.6.143.194)`
`23:13.09`	`Defraz`	`So I call into my system put myself on hold and I hear hold music only when I am talking or make a sound`
`23:13.35`	`jsgoecke`	`Figured out my app_conference MOH issue, turns out to be the 'm' option and not the 'h' one http://forums.digium.com/viewtopic.php?p=129102#129102`
`23:13.46`	`Defraz`	`Anyone have any idea?`
`23:19.28`	`*** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de)`
`23:29.26`	`*** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl)`
`23:29.36`	`VaGoNeTaS`	`hello buddys`
`23:29.42`	`VaGoNeTaS`	`i've just installed asterisk with a redfone quad box`
`23:29.51`	`VaGoNeTaS`	`i havent pluged the E1 line yet`
`23:30.03`	`VaGoNeTaS`	`but id like to know if redfone is being loaded by asterisk`
`23:30.08`	`VaGoNeTaS`	`how should i know that?`
`23:30.35`	`VaGoNeTaS`	`besides fonulator -V command on the shell`
`23:30.49`	`*** part/#asterisk Schreiber1337 (n=SCHREIBE@216.169.165.178)`
`23:30.54`	`VaGoNeTaS`	`there is some kind of "load fonulator.so" or something on the asterisk console?`
`23:33.01`	`VaGoNeTaS`	`i even have a loopback cable connected to my redfone`
`23:39.37`	`*** join/#asterisk eliel (n=eliels@120-17-235-201.fibertel.com.ar)`
`23:39.37`	`*** join/#asterisk juanIMP (n=Juancho@200.26.152.222)`
`23:40.19`	`*** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk)`
`23:45.10`	`jsgoecke`	`VaGoNeTaS Have you checked with the Redfone folks?`
`23:46.32`	`VaGoNeTaS`	`nop dude`
`23:47.08`	`Qwell`	`seanbright: just went from 80% to about 95% after fixing it`
`23:48.42`	`Qwell`	`pre-filled ones suck apparently`
`23:48.57`	`*** join/#asterisk ltd (n=z@pat.transact.net.au)`
`23:49.02`	`VaGoNeTaS`	`?`
`23:49.17`	`VaGoNeTaS`	`asterisk-prodata*CLI> dahdi show status`
`23:49.18`	`VaGoNeTaS`	`Description Alarms IRQ bpviol CRC4`
`23:49.18`	`VaGoNeTaS`	`DAHDI_DUMMY/1 (source: HRtimer) 1 UNCONFIGUR 0 0 0`
`23:49.30`	`VaGoNeTaS`	`shouldnt be showing that`
`23:49.45`	`VaGoNeTaS`	`should've been YEL alarm or something like that`
`23:51.20`	`VaGoNeTaS`	`root@asterisk-prodata:~# fonulator -vq`
`23:51.20`	`VaGoNeTaS`	`Detecting foneBRIDGE`
`23:51.20`	`VaGoNeTaS`	`Connection to device timed out! Check network or device power.`
`23:51.20`	`VaGoNeTaS`	`statusInitalize: Internal foneBRIDGE library error`
`23:53.47`	`jameswf`	`~pb`
`23:53.47`	`infobot`	`[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/`
`23:57.01`	`VaGoNeTaS`	`is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ]`
`23:57.31`	`*** join/#asterisk Aiatek (n=munoz@190.6.143.194)`

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