irclog2html for #asterisk-doc on 20061129

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15:38.25blitzragejsmith: good morn'!
15:39.03jsmithIs it?!?
15:39.06jsmithHow do you know?
15:39.52blitzrage:D
15:39.55blitzrageI guessed
15:40.02blitzrageso you want to know something annoying?
15:40.33blitzrageWhen you use one-touch recording, Asterisk uses the EPOCH of when the recording was started, so you have no method of knowing the filename of the one-touch recorded file
15:41.36jsmithI think there's a channel variable you can set to change that ;-)
15:41.41jsmithAt least, I seem to remember that.
15:41.59blitzrageluckily, I can change the 'src'-'dst' part of the string to whatever I want with TOUCH_MONITOR, so I can at least put in a unique value for the file (thinking maybe the unique channel identifier), which I can then pass to an AGI script on hangup, find the file, rename it appropriately, and place an entry in the DB
15:42.27blitzrageyah-- you're thinking TOUCH_MONITOR... and it only changes the part after the auto-EPOCH-...   part
15:43.04blitzrageand I can't upgrade to 1.4 because HASH() is only in 1.2 (svncommunity)
15:43.19blitzragenot that upgrading to 1.4 would help in this case afaik, but it'd be nice to test on it :)
16:07.03blitzragejsmith: ping... still around?
16:07.12jsmithMostly
16:07.17jsmithGotta start teaching here in a minute
16:07.20blitzrageok... php question for you... :)
16:07.20jsmithWhat's up?
16:07.52blitzragedo I need to iterate through every file in my ./monitor directory, looking for my "uniqueid" in the filename, or is there a way I can say, "hey, look for a file with this string in it"
16:08.02jsmithgrob
16:08.06jsmithglob, that is.
16:08.06blitzragemerci!
16:08.11blitzragewill check it out
16:10.52jsmithphp.net/glob
16:12.22blitzragejsmith: does it need to check the entire filesystem?
16:13.58blitzrageahhh, looks like I can do $path/"search_parm"
16:14.02jsmithNope... just the current directory
16:14.04jsmithOr the path you show
16:14.57blitzrageright -- the example on the page didn't say anything about paths, but another example shows glob($path.$match)
16:16.05jsmithYup -- glob("/var/lib/asterisk/monitor/$uniqueid*");
16:16.31blitzrageand it returns the whole filename right?
16:16.47blitzrageahhhh yes it does
16:16.48blitzrageperfect
16:16.50blitzragethanks!
16:50.07blitzragedoh!!!  guess Asterisk only mixes the two recorded channels once the whole call is hungup
16:53.36jsmithUsing Monitor or MixMonitor?
16:53.42jsmithOh, with one-touch recording?
16:53.45jsmithI think you can choose.
16:54.42blitzragethe way it works is it records the files separately, then mixes them with sox automatically when complete, but my DeadAGI() is getting to the files before the mixing
16:55.05blitzrageif I could use mixmonitor in the one-touch, that'd be ideal, because then its already mixed right away
16:55.27jsmithShould be a way to do it.
16:55.28blitzragewonder how I configure asterisk to use mixmonitor instead...
16:55.31blitzragejsmith: either that, or I mix them myself inside the DeadAGI()
16:56.23jsmithThat might work as well.
16:57.09blitzrageyah... I'm afraid what asterisk is going to do when I mix it, then it goes looking to mix the files, and they don't exist :0
16:57.13blitzrageerrr :)
17:41.36blitzragenice... so asterisk didn't freak out
17:41.48blitzrageand advantage, I converted all the wav's to mp3 now
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18:33.38jsmithblitzrage: Was I supposed to help you w/ something after I got done teaching?
18:35.26blitzragejsmith: nope, you gave me the hint with glob() before you taught
18:35.31blitzragethanks for checking though
18:35.39blitzrageI got my AGI all complete, and it converts to mp3 and everything :)
18:36.09blitzragelunchtime!!!
18:36.39blitzrageso call recording is done. Going to move onto playing with HINTS
18:37.17blitzragethen after that, gonna see about remapping the keys of my polycom so when someone hits the DND button, it'll send a command to Asterisk instead of setting the phone into DND mode........
18:37.23blitzragehoping thats possible
18:37.26blitzrageanyways... salad time!
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20:29.56blitzragejsmith: ever used the buddy watch system on polycoms?
20:37.40blitzragehrmmmm.... seems it works with numbers, but not with strings (at least not strings with a # in them)
20:37.44blitzragethat sucks...
20:57.14blitzrageok.... so I got it working with <ext>_<pbx>
20:57.21blitzragemaybe first char needs to be an int
20:57.29blitzrageoh well, I can work with that at least
20:57.50blitzragenow I need to create a script that will generate the -directory.xml files and the .cfg files for polycoms
20:58.19blitzrageshouldn't be too bad, not changing all that much right now in the files
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23:35.25blitzragew00t
23:35.30blitzragePolycom?
23:36.05AlricYeah
23:36.13blitzrage1.2 or 1.4?
23:36.15AlricIf someone calls me, I get one-way audio
23:36.17Alric1.4 beta3
23:36.21AlricIf I call them, it works fine
23:36.25blitzragetry changing your friends to peers
23:36.33Alricwhy?
23:36.42blitzragegive it a shot and if it works I'll tell you
23:36.50AlricI've already updated to svn
23:36.56Alricsvn 1.4
23:37.03AlricOr, rather, in the process of updating.
23:37.07blitzragesame answer
23:37.12AlricI guess I could try it, haven't stopped the server yet.
23:37.42blitzragedoesn't matter if its beta or svn -- I've been seeing a bunch of talk about changing your friends to peers for hints and such
23:38.06AlricNope
23:38.07AlricStill busted.
23:38.12blitzragehrmmmm
23:38.34Alricand I can't leave until this gets fixed :/
23:38.45AlricI suspect it may be my provider though.
23:38.47blitzrageasterisk behind nat?
23:38.53AlricBecause they just did an upgrade last night on some stuff.
23:39.00Alricand afaik, it worked before then.
23:39.04blitzragehow is the routing
23:39.06AlricNo
23:39.21AlricVoIP switch <-> Asterisk <-> Phone
23:39.30AlricAsterisk has 2 interfaces, one public one private, phone is on private
23:39.36blitzrageif you call them it works fine, and if they call anywhere else its one way audio?
23:39.54Alricif they call me its one-way
23:39.58Alricthey can hear me, I cannot hear them
23:40.06Alricit shows me sending RTP packets
23:40.09Alricbut receiving none
23:40.42blitzragethey are offsite right?
23:40.46Alricoffsite?
23:40.52Alricthe caller?
23:40.54blitzragelike is your phone part of the same private network?
23:41.30Alricas the pBX?
23:41.32Alricor as the caller?
23:42.19blitzrageok... your phone is connected to asterisk which then goes to a provider, and the provider is delivering the call to their pbx -> phone?
23:42.42Alriccaller -> pstn -> my provider -> asterisk -> my phone
23:42.55Alricthat answer it?
23:43.04blitzragecaller is a cell phone or something?
23:43.12Alriccaller is an SBC POTs line
23:43.21blitzragegotcha
23:43.27Alricbut a Verizon cell also doesn't work
23:43.35Alricthis is if they call me.
23:43.39Alricif I call them, all works.
23:43.44blitzragedo you have a second provider?
23:43.49Alricno
23:44.07blitzragewant me to forward a 416 test number to your asterisk so you can verify its not you?
23:44.11AlricThis provider = best rates in the world :)
23:44.29blitzrageyah well... you're single homed :)
23:44.41AlricHaven't had issues before :/
23:44.59AlricLet me finish this svn update
23:45.07AlricThen if its still not working we can try the 416 test number.
23:45.29Alricadditionally, I have a full sip/rtp debug
23:45.31Alricall looks normal though
23:45.44Alricto my untrained SIP analyzing eye :)
23:45.50blitzragesounds good
23:46.06blitzrageI need to generate a polycom creation script
23:46.18AlricHmmm
23:46.22AlricI have a sloppy one I think.
23:46.34Alricactually, wouldn't work.
23:46.45Alricmine pulls info from a DB about its sip name and all that jazz.
23:47.13AlricI want polycoms to autoprovision :)
23:47.16blitzrageyah, mine is gonna do that too
23:47.19Alricand add themselves into my GUI.
23:47.22blitzrageI understand :)
23:47.35Alricand then have the GUI email me and tell me that a new phone was added
23:47.37blitzrageI wish I was better with GUI work
23:47.41blitzragenice
23:47.47Alricand I can say "MUHAHA! Another license!"  and bill them $700!
23:47.50AlricBrillant!
23:48.02blitzragenice moves
23:48.10Alricand I wonder why I don't have customers...
23:48.33Alricsvn is being slow.
23:48.36blitzrageI'm still building infrastructure.... oh so slowly
23:48.45blitzrageI can't even touch 1.4 yet...
23:48.50blitzrageI need to get a couple of apps...
23:49.13blitzrageI've got so much patched into 1.2 anyways. I have features that aren't even in trunk
23:49.19Alricbuilding infrastructure?
23:49.30Alricyou mean infrastructure tools?
23:49.37blitzragebuilding a large set of features into a complete rewrite of my Asterisk backend
23:50.10blitzrageI got write of the weird PHP script generation and built the DB connection directly into the dialplan
23:50.13AlricHmmm
23:50.19blitzrageinstead of using PHP to generate the dialplan
23:50.29blitzragewhich was a stupid amount of overhead for no gain
23:50.30Alricah
23:50.43Alricusing PHP to generate the dialplan was stupid for no gain
23:50.43Alric?
23:51.32AlricFrack.
23:51.37AlricDamn torisa.c
23:51.39AlricWon't compile.
23:51.40blitzrageall it did was slow down development. I finally have all the buddy watch stuff, recording, mp3 conversion, and DB lookups directly from extensions.conf. No realtime issues either, but I can pull in rows of info via variables.
23:52.16Alricrecording, huh?
23:52.20Alriclike dynamic call recording?
23:53.18blitzrageevery vPBX can either record all PBX calls, all device calls, or one-touch recording with recording saved to mp3 and assigned to a CDR record in the DB.
23:53.48AlricOne-touch recording.  Hmmm
23:53.52AlricHow did you accomplish that?
23:54.16blitzrageit was a bitch of a bitch :)
23:54.29AlricYeah
23:54.33AlricI can't think of any way...
23:54.38blitzrageI made my first DeadAGI. Want me to send you the code?
23:54.45blitzrageits pretty simple now
23:54.52AlricCool, I'd love to see it.
23:54.57blitzragemsg email
23:56.01AlricWhat do you think about the * Gui framework?
23:56.08AlricFeel like you wasted development on nothing yet? :P

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