IRC log for #asterisk on 20091229

00:08.19Gio__nobody ?
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00:30.31Get_The_Fishdoes anyone know of a web frontend to the AstDB?
00:31.13bcrispgasps
00:31.21bcrisp~roulette
00:31.21infobotACTION watches bcrisp pull the trigger:  Click!
00:31.33Get_The_Fishlol
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00:31.55Get_The_FishI want users to be able to look at the entries in the blacklist, add/change/delete
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00:40.28sahafeezanyone have a recommendation for someone that can provide dailtone via iax in the uk and the usa
00:40.38hardwireteliax
00:40.39hardwire:P
00:41.12sahafeeznumbers in the uk?
00:41.36sahafeezyah no
00:41.52hardwireyou may have to split the load
00:43.20drfreezeI've got phones in two offices connected by a Pt2Pt (T1).
00:43.35drfreezeThe phones in the main office have been setup for a month now and work fine
00:43.50voipmonkok...
00:43.51drfreezeThe pt2pt was connected today and the offices can call each other fine
00:44.11drfreezeBut, the phones in the terminal office can't call out. I get the error: Channel 0/2, span 1 got hangup request, cause 41
00:44.41drfreezeThe phones in the far office use the same dialplan too
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00:46.32gwav8orneurosys you on?
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00:47.52hardwireheh
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00:58.50ManxPower-workCause 41 is "Temporary Failure: The call was disconnected due to a network failure.   The network is not functioning correctly and that the condition is not likely to last a long period of time;  e.g.  the user may wish to try another call attempt almost immediately. "
01:00.55hardwirewe should train infobot on those
01:01.14hardwireinfobot: hug
01:01.14infobotACTION hugs hardwire
01:01.17hardwire!!!
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01:14.55bahjonsdoes anyone know where I can find documentation on asterisk realtime? namely database structures
01:24.20[TK]D-Fenderbahjons: in the tarball
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01:25.10drmessanoIN TEH TARPITS
01:25.12drmessanoMR BOND
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01:26.54bahjons[TK]D-Fender: yea the only thing that references the realtime is docs/realtime.txt, docs/extconfig.txt, configs/sip.conf, configs/extconfig.conf, but nothing has the database structure.
01:28.28bahjons[TK]D-Fender: I'm still working with this state_interface garbage, but can't get call-limit field in realtime to work. I've determined that is the cause of the problem. It's the only different in settings between realtime and conf.
01:29.36[TK]D-Fenderbahjons: Should map fine if the name matches in your DB AFAICT
01:30.49bahjons<PROTECTED>
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01:52.06steve745can anyone help me with getting sql data like agent status available or unavailable???
01:52.20steve745what db would i query and what field
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02:20.23ManxPower-worksteve745: Are you running agents in Realtime?
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02:48.35ChannelZThis again?
02:49.32jblackI dare you to watch this: http://www.getonmyhorse.com/
02:49.46jblackNSFW, NSFC, NSFS, NSFL
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02:51.29jblackShut up woman, get on my horse!
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02:54.53ChannelZHmm.
02:55.15ChannelZI like weebl and bob but that's just kind of too random to be funny
02:56.55jblackThis is fantastic. http://www.weebls-stuff.com/toons/Meow/
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03:01.02kerxHi, I'm receiving lots of messages that look like this:  "WARNING[20903]: chan_sip.c:1804 __sip_xmit: sip_xmit of 0xd3ee790 (len 959) to 192.168.1.141:5060 returned -1: Operation not permitted"
03:01.50kerxI'm not sure what to make of this, however I know the phones are not able to get called
03:03.14loathsomekerx: are you running asterisk as root?
03:03.23kerxloathsome, Yes
03:03.28[TK]D-Fenderjblack: That was..... retarded
03:03.30loathsomeok, do you have selinux enabled?
03:03.40ChannelZsexlinux
03:03.59jblackWhich one? The really retarded one, or the incredibly retarded one?
03:04.02kerxselinuxenabled ; echo $?
03:04.04kerxI receive a 1
03:04.18kerxi have my SELINUX=disabled
03:05.09kerxman page tells me selinuxenabled returns a 1 if disabled
03:05.25kerx-rw-r--r-- 1 root root 24576 Dec 28 18:57 astdb
03:09.58kerxAny suggestions?
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03:27.44jesse098765Hi Everyone.  Anyone know why my MP3Player won't play?  The .wavs works fine, but there seems to be a problem.  I'm new.... http://pastebin.ca/1730211
03:28.33Tech_TravisDoes * read the extensions.conf file sequentially?  I have an existing dialplan with 714-555-1212 coming in with a couple of IVR options, 1 for tech-support going to queue1, 2 for sales etc. Now I need to add a second number 619-555-1212 but keep the same option numbers such as 1 for techs, however it needs to go to tech-support queue2.  Will this approach work if I just add the new number and menu options after all of the origina
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03:31.17ChannelZTech_Travis: it's not sequential in so much as it will 'fall through' to a totally different extension
03:32.23ChannelZbut depending on your setup you could just set a channel variable with the name of the queue you want for each number, and centralize the main IVR into a different different extension that the other two jump to for instance
03:32.51[TK]D-FenderTech_Travis: Thats what CONTEXTS are for.
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03:36.48Tech_TravisChannelZ: I'm okay with changing around existing dialplan to centralize things, just having a hard time wrapping my mind around how best to set this up so I won't need to redo it each time a new number needs to be created.
03:37.47Tech_Travis[TK]D-Fender: I wasn't sure that contexts could be used before the call actually comes into the box, I mistakenly believed that the call needed to be picked up before I could use contexts for routing.
03:38.22[TK]D-FenderTech_Travis: Every line in extensions.conf is an opportunity to go somewhere ELSE
03:38.32[TK]D-FenderTech_Travis: and there is no such thing as "routing".
03:38.50[TK]D-FenderTech_Travis: Another term you can trown in the dumpster
03:38.54[TK]D-Fenderthrow*
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03:40.03Tech_Travis[TK]D-Fender: dumpster noted. what is the correct term be for sending things elsewhere in the dialplan?
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03:41.35[TK]D-FenderTech_Travis: No magic term.... however Goto() as an app is reference enough.  its always a question of what command you use.
03:41.55[TK]D-FenderTech_Travis: Then there is the concept of INCLUDE-ing contexts in another.
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04:05.47drfreezeAnyone have any luck with creating a call file that can call to a channel other than Sip?
04:06.22drfreezeIf I send it to a Dahdi channel, I get: pbx_spool.c:356 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy)
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04:10.06[TK]D-Fenderdrfreeze: It works with anything you'd Dial()
04:10.17[TK]D-Fenderdrfreeze: So show us everything relevent and we'll tell you why
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04:14.48hardwiremeh
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04:18.10drfreeze[TK]D-Fender: http://pastie.textmate.org/private/agyhfgmdsfiftnros8cqfg
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04:23.51pagecin ael is there a way to dial a number in another context with using include?
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04:32.57[TK]D-Fenderdrfreeze: Show my your configs, and your failed attempt
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04:38.17[TK]D-Fenderpagec: There is this mircale app called Goto() you could try....
04:49.40drfreeze[TK]D-Fender: http://pastie.textmate.org/private/ompn19p9mwgawxi81fvrg
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04:57.45[TK]D-Fenderdrfreeze: Cause No. 21 - call rejected.  This cause indicates that the equipment sending this cause does not wish to accept this call. although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible. This cause may also be generated by the network, indicating that the call was cleared due to a supplementary service constraint. The diagnostic...
04:57.47[TK]D-Fender...field may contain additional information about the supplementary service and reason for rejection.
04:57.59[TK]D-Fenderdrfreeze: Seems pretty clear to me.  Call file is fine
05:01.43drfreeze[TK]D-Fender: seems as unclear as ever to me
05:01.55drfreezethere is no reason that I can see why the call should be rejected
05:02.03drfreezethe internal context dials out all the time
05:02.09[TK]D-Fenderdrfreeze: Where are you that a 7 digit number is legal to dial?
05:02.20drfreezeDAHDI/g1/number work just fine, no rejection
05:02.38[TK]D-Fenderdrfreeze: Show me you calling that same number normally.
05:02.49drfreezework for 5551212, 5125551212 and 15125551212
05:02.58drfreezebut, all those cases fail in the call file
05:03.14[TK]D-Fenderdrfreeze: SHOW ME
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05:05.26drfreezeexten => _NXXXXXX,n,Dial(DAHDI/g2/${EXTEN})
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05:07.40[TK]D-Fendermoves on to more productive matters
05:11.19drfreeze[TK]D-Fender: not on site right now to make that call locally
05:11.54drfreezenot sure how to make a call from a sip extension
05:12.13[TK]D-Fenderdrfreeze: ..... You don't know how to dial a friggen number on a soft phone?
05:12.24[TK]D-Fenderdrfreeze: How long have you been using * now>?
05:14.54drfreezedon't have a softphone installed at the moment
05:17.17[TK]D-Fenderdrfreeze: Eitehr way, you're getting an ISDN rejection and aren't looking at PRI debug.  Your number doesn't look legit in that jsut about ever NA PRI I've ever seen tend to demand 10-digit dialing.
05:17.45[TK]D-Fenderdrfreeze: Enable PRI debug and see if it tells you anything more
05:19.31loathsomein all my 7-digit dial sections I have to prefix a local area code.
05:20.04loathsomein fact, i have to prefix the 1 as well. It needs 11-digit dialling.
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05:56.36drfreeze[TK]D-Fender: I installed a sip phone
05:57.14drfreezeIt is behaving the same as the call file. Somehow, the sip phone and the polycom phones are different, even tho both use the same dialplan
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05:59.55[TK]D-Fenderdrfreeze: Perhaps its the CALLERID that it doesn't like.  Perhaps you should try actually providing one with your call file.
06:00.09[TK]D-Fenderdrfreeze: PRI's tend to expect you to announce yourself properly
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06:06.14drfreeze[TK]D-Fender: seems like adding the caller id fixed the problem
06:06.54Get_The_Fishcan anyone point me to some documentation on the AMI MXML interface?  Havent been able to find much on this...
06:07.56[TK]D-Fenderdrfreeze: Good to hear... PRI debug may have alluded to this if you looked at it when you started all of this, and is a factor that should never be overlooked
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06:11.02drfreezeok, now I need to get that callerid set in the call file
06:13.55Get_The_Fishanyone know why I would get a "permission denied" immediately following a successful login using the mxml interface
06:20.05drfreeze[TK]D-Fender: added callerid to the call file
06:20.14drfreezeit makes the call but my phone never rings
06:20.14drfreezehttp://pastie.textmate.org/private/jo5kic1fdcezrysjirjvq
06:21.11Get_The_Fishnevermind on that, I apparently cant read
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06:22.34[TK]D-Fenderdrfreeze: And All I see is you running out of dialplan to execute
06:22.58drfreeze[TK]D-Fender: ok, duh, figured it out
06:23.09drfreezehad the '1' in the number from a previous trial
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06:39.03BugKhaMWhich variable in asterisk holds the client(either SIP or IAX)'s IP address?
06:48.00Get_The_FishBugKham, function SIPPEER(peername,ip)
06:48.36Get_The_Fishdoes anyone know what the sipregs realtime family does?
06:52.16BugKhaMGet_The_Fish, does it work for agi?
06:52.42Get_The_Fishshould, but I dont know...
06:52.42Get_The_Fishhttp://www.asterisk.org/docs/asterisk/trunk/functions/sippeer?type=functions&value=SIPPEER
06:52.49Get_The_Fish(I love that page)
06:56.01BugKhaMGet_The_Fish, thanks
06:56.41Get_The_Fishnp
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07:03.32shamelessn00bchannelz
07:03.42shamelessn00bHi guys
07:04.10shamelessn00bhey regarding that asterisk server stress testing, thiis server is pissing me off, its handling about 300 SIP calls
07:04.35shamelessn00band it shows 20 percent CPU utilization at upto 100 dhadi calls
07:04.51shamelessn00bbut just as I reach 101-103 calls CPU shoots up to 100 percent
07:05.02shamelessn00band all hell brakes lose
07:05.16shamelessn00busing asterisk 1.6.2.0
07:05.29shamelessn00bppl saying I should switch to 1.6.1.12
07:05.54shamelessn00bdunno if its asterisk to blame or DAHDI/wanpipe
07:11.25aiksa[LV]morning
07:11.26aiksa[LV]:)
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07:13.13shamelessn00bmorning :)
07:14.31aiksa[LV]shamelessn00b: could be related to some other bottleneck
07:14.53aiksa[LV]lets say - recording for example
07:15.03shamelessn00bIm landing on the same context
07:15.11shamelessn00bin SIP calls
07:15.12aiksa[LV]if at 100 calls you were at 80% of your disk max write speed
07:15.13aiksa[LV]....
07:15.15shamelessn00band DAHDI calls
07:15.26shamelessn00bso I doubt it
07:15.40shamelessn00bthe system can manage around 250-300 SIP calls
07:15.51shamelessn00blemme try that again
07:16.03shamelessn00bgimme like 4-5 mins
07:16.26BugKhaMshamelessn00b, I'm using asterisk 1.2 + zaptel on my P3 box and can handle around 80 calls thru ISDN PRI
07:17.27aiksa[LV]shamelessn00b: do you have sw echo canc on those dahdi callls?
07:17.36aiksa[LV]this could be another issue
07:17.50aiksa[LV]oslec is rather resource hungry if that is your choice
07:17.52shamelessn00bI was testing on P4 dual core 3 ghz with 1 gb ram
07:18.01shamelessn00band xeon quad core 2 ghz
07:18.05shamelessn00b2 gb  ram
07:18.15shamelessn00bboth crashing after exactly 100 calls
07:18.33shamelessn00bbefore 100 they show minimal loads
07:18.39shamelessn00blike 20 30 percent CPU
07:18.52shamelessn00bexcept for when the calls are actually being set up
07:18.52aiksa[LV]and 15 min load AVG of what?
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07:19.08shamelessn00baround the same
07:19.16shamelessn00bthey are constant at 20 percent of CPU
07:19.17aiksa[LV]0.2 that woudl be
07:19.21shamelessn00byeah
07:19.26aiksa[LV]ok
07:19.47shamelessn00bno idea why this is happening just as I reach above 100 calls
07:21.41shamelessn00b100 DAHDI calls
07:21.43shamelessn00b**
07:25.21tzafrir_laptopshamelessn00b, system crashes? or just hangs?
07:25.29shamelessn00bhangs
07:25.33shamelessn00bindefinately
07:25.38shamelessn00bdoesnt process any call
07:25.42shamelessn00bSIP or DAHDI
07:25.47tzafrir_laptophangs: makes sense. No more CPU cycles available
07:25.54shamelessn00band the DAHDI card start showing error messages
07:26.03shamelessn00bI frame out of sync
07:26.03tzafrir_laptopwhat DAHDI device? What echo canceller?
07:26.11shamelessn00bhardware
07:26.15shamelessn00bsangoma
07:26.18ChannelZor your PCI bus is going titsup
07:27.02aiksa[LV]PCI bus should be able to handle more than 100 calls
07:27.18aiksa[LV]if its not maxed out by other things
07:28.06tzafrir_laptopshamelessn00b, start adding calls steadily, and keep an eye on 'top' while you do that
07:28.17shamelessn00byeah I did that
07:28.23shamelessn00btried with 1 call a sec
07:28.32shamelessn00bthat waits for 600 secs
07:28.38shamelessn00blands in the remote context
07:28.46shamelessn00bthat runs some AGIs
07:28.59shamelessn00b1 call/sec 100 times
07:29.07shamelessn00b(did it using a script and a call file)
07:29.13tzafrir_laptopAlso note that the setup of the call typically takes more cpu time than an actual "steady state"
07:29.20shamelessn00byeah
07:29.22shamelessn00bI mentioned
07:29.23shamelessn00bthat
07:29.31tzafrir_laptoptry leaving e.g. 3 seconds or more between new calls
07:29.43shamelessn00bits handling 1 call/sec
07:29.48shamelessn00band 100 calls/sec
07:29.58shamelessn00bits not handling 101-103
07:30.01shamelessn00bcallsin total
07:30.12shamelessn00bwhich pisses me off
07:30.39tzafrir_laptopyou typically hit a wall somewhere when you run out of free cpu cycles
07:30.48shamelessn00bwell I didnt really try 100 calls /sec
07:30.57shamelessn00bI tried 10 calls/sec 10 times
07:31.08shamelessn00bsomewhere around 12 calls/sec
07:31.13shamelessn00bit gets messy
07:31.21shamelessn00bbut recovers
07:31.31shamelessn00bit doesnt recover if I hit more than 100 calls
07:45.36aiksa[LV]tzafrir_laptop: btw there is a nice alternative for top - htop
07:46.13tzafrir_laptopaiksa[LV], but htop takes more cpu cycles :-)
07:47.47aiksa[LV]tzafrir_laptop: yeah it does, but gives more insight
07:48.09aiksa[LV]whats happening across the cores etc.
07:48.10tzafrir_laptopfor the global load? not really
07:48.49tzafrir_laptoppress '1' in top to toggle multi-cpu view
07:48.58aiksa[LV]wow. thanks
07:48.59ChannelZand 'I'
07:49.40aiksa[LV]I learened something new today
07:50.07aiksa[LV]and perhaps top also knows how to cascade processes and scroll?
07:50.33shamelessn00bim using htop :P
07:51.10shamelessn00bhey can anyone tell me how to set up sip calls on local system using call files
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07:52.08aiksa[LV]just as DAHDI calls, just change channel specification to SIP/something
07:52.44shamelessn00bChannel: SIP/3002
07:52.46shamelessn00bExtension: 1001
07:52.48shamelessn00bContext: default
07:52.49shamelessn00bPriority: 1
07:53.08shamelessn00bim on the same system and originating calls that land on the same system
07:53.24shamelessn00b3002 is a SIP account I registered on that sysem
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07:53.34shamelessn00bI just want to eliminate dahdi for a while
07:55.13shamelessn00bbut not working
07:55.15shamelessn00b:F
07:55.42aiksa[LV]where does calls to SIP/3002 go?
07:56.11aiksa[LV]3002 is an entry in sip.conf right?
07:56.28aiksa[LV]Does it have registered peer associated with it?
07:57.04benngardany knows who "may213" is, if he can be reached here or at mail
07:58.12shamelessn00byes
07:58.23shamelessn00bhow do I connect it to the default context
07:58.25shamelessn00b?
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08:02.55aiksa[LV]well I would make two entries in my sip conf
08:03.03aiksa[LV]one for peer and one for user
08:03.12aiksa[LV]with the same credentials
08:04.02aiksa[LV]so when routing a call to peer it would be picked up user
08:04.20aiksa[LV]then you have to define a context for a receiving party
08:04.37aiksa[LV]and within that context simply add an s extension
08:04.57aiksa[LV]with a) Answer and b) Wait(THE_LENGHT_OF_CALL)
08:05.53aiksa[LV]c) Hangup
08:06.18aiksa[LV]smth. like that
08:06.36aiksa[LV]a loopback call to describe in more simple terms
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09:08.21ChannelZhmm anyone know of any free faxback services off the top of their head so I can test a fax receive?
09:12.18sawgoodI just saw one tonight ... give me a second
09:14.25sawgoodI searched on voip-info.org for ENUM stuff, and in the list was a URL for free faxes to US and Canada
09:14.32sawgoodI can't recall it right off the top of my head ... sorry
09:14.38ChannelZI think I found one, http://faxzero.com/
09:16.12sawgoodYeah, I have an account with them too ...
09:16.27sawgoodvery good for testing SIP and PSTN faxes at customers sites
09:18.39ChannelZthis is PSTN, I just got a bunch of fax voicemails and something wierd is happening in my fax detection in 1.6
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09:20.17ChannelZwell wtf
09:22.34ChannelZ-- Redirecting DAHDI/3-1 to fax extension
09:22.40ChannelZ-- Sent into invalid extension '?' in context 'incoming' on DAHDI/3-1
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09:47.41ChannelZhmm ok anyone know what the frequency of the initiate tone on a sending fax machine uses?  one article I found suggests 2100hz but that sounds too high
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09:49.43shamelessn00bChannelZ: whats chan_woomera
09:49.45shamelessn00b??
09:49.59shamelessn00balso, Im suck withthis shitsux stress testing
09:50.28shamelessn00bmy system is able to land `250 SIP calls on a context
09:50.53shamelessn00bbut if I make dahdi calls to taht context load averages at 20 percent till 100 calls
09:51.06shamelessn00bthen as I increase number of calls beyond 100 the cpu goes crazy
09:51.19shamelessn00ball cores on 100 and no calls are entertained
09:57.09ChannelZI have no idea, chan _woomera ??
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10:03.39benngardi have some aastra rfp32 running h.323, any1 knows if they can be converted to sip?
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10:16.04shamelessn00byeah chan_woomera
10:16.11shamelessn00bI read its some alternate for dahdi
10:17.13ChannelZ"The chan_woomera channel driver allows the Asterisk IP-PBX to act as a Woomera client."
10:17.17ChannelZI've never heard of Woomera
10:18.21ChainsawChannelZ: Initiating tone can mean two things. "CallED" is 2100hz, "CalliNG" is 1100hz.
10:18.35ChannelZChainsaw: yeah it appears to be 1100hz
10:18.56ChannelZfax detection is borked in >1.6.0.5
10:19.20ChainsawChannelZ: Seems to work for me on a TDM400P in 1.6.1.12
10:20.07ChannelZhmm try it a few times.  I got like 6 fax voicemails, and looking at the console get what I pasted awhile back, sending it to extension '?' (where ? is actually a strange non-ascii character)
10:20.35ChannelZSo I used one of those faxback things to test, and that actually went through.  But all subsequent tests are failing
10:21.43ChannelZcutting out the middleman just calling in and playing a 1100hz tone to trigger, and it keeps coming up with this wierd extension
10:23.17ChainsawChannelZ: Provided I patch the Asterisk core so the BT automated linetest doesn't knock my line out every night... faxing works forever.
10:23.53ChainsawChannelZ: You may find this document helpful, at any rate: http://telecom.tbi.net/fax-call.htm
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10:24.17ChannelZI think I just discovered the trigger
10:25.03ChannelZIf the tone comes in during a Background() it's barfing the extension.  If the tone comes in before or after (in my case 'after' is a WaitExten()) it works
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10:26.12ChainsawChannelZ: Yes, don't try to listen for DTMF when detecting a fax.
10:26.59ChannelZwell if I knew I was listening for a fax I wouldn't need fax detection
10:27.59*** join/#asterisk DND (n=arabia@94.200.7.26)
10:28.09DNDhi guys, what codec does x-lite use?
10:28.15ChainsawDND: Whichever one you tell it to.
10:28.56DNDhmm i just thought that x-lite only uses ulaw
10:28.58ChainsawChannelZ: The documentation I linked you to gives you the time you need for efficient fax detection. I'd say roughly 9 seconds to be safe.
10:29.30ChainsawChannelZ: You can do whatever DTMF-related thing you want to do afterwards, but you need to get the faxes shunted away to your fax backend first.
10:30.15DNDChainsaw, i dont see any dropdown to choose codec on x-lite
10:30.20DNDthis is the free version
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10:30.45ChainsawDND: The cheapskate version is ulaw-only, yes.
10:30.50ChainsawDND: The proper one is not.
10:31.47DNDhmm so eyebeam is the one that has most codecs
10:31.47DNDx-lite free is ulaw
10:32.12ChainsawMy interest in proprietary software is low. But yes, that sounds correct.
10:32.14ChannelZChainsaw: this is a bug.. the point is, * is detecting the 1100Hz tone just fine, even says it's redirecting, but then does so to an invalid extension
10:32.38ChainsawChannelZ: Because you're listening for DTMF at the same time, or too soon afterwards.
10:32.46ChainsawChannelZ: Shunt your fax *away* before you listen for DTMF.
10:32.49ChannelZDND: Zoiper has a few more; GSM, ulaw, alaw, speex, something else..
10:33.15shamelessn00bchannelz do you have some time, I need help on something very basic
10:33.37ChannelZChainsaw: nevermind you're not getting what is going on
10:33.54ChainsawChannelZ: That must be it, yes. Good luck.
10:34.18ChannelZshamelessn00b: I really need to get to bed but based on what you've asked earlier I can't help you anyway with this system overload issue
10:34.43ChannelZChainsaw: https://issues.asterisk.org/view.php?id=16050
10:35.41DNDChainsaw, can you suggest a free or open source softphone that requires less bandwidth? i read that g711 requires 80kbit both ways?
10:35.48shamelessn00bwell, I want to take dahdi out of the equation
10:35.56shamelessn00band make SIP calls to taht context
10:35.56ChainsawDND: Isn't there a Windows port of Ekiga these days?
10:36.10shamelessn00bIm using call files to generate the calls
10:36.25shamelessn00bI have 2 SIP extentions defined on the target system
10:36.35shamelessn00band the context answers on extention 1001
10:36.42shamelessn00bthe SIP accounts are 3001 and 3002
10:37.08shamelessn00bso I want to make a call from 3001 to 3002 that plays the context defined onextension 1001
10:38.20tzafrir_laptopshamelessn00b, the simplest way to originate calls is with the 'originate' CLI command
10:38.33shamelessn00b100calls
10:38.35shamelessn00bor more
10:38.43shamelessn00bthats why im going for a call file
10:38.49tzafrir_laptopIt's a bit limited (vs. call files and the manager interface), but it's way simpler to script
10:38.58drmessanoX-Lite has GSM, Speex, iLBC, and ULAW
10:38.59shamelessn00bI made a simple script
10:39.07drmessanoGSM or Speex work fine
10:39.21shamelessn00bthat renames copies the call file to /var/spool/asterisk/outgoing/
10:39.23tzafrir_laptoporiginate SIP/whatever extension 123456@test-context
10:40.02shamelessn00busing call files
10:40.04shamelessn00b:/
10:40.10DNDdrmessano, how will i define GSM on x-lite? or i'll just retrict it in extensions page?
10:40.20shamelessn00bcall files have thier own syntax
10:40.22drmessanoGo into the options and look man
10:40.33drmessanoEnabled Codecs
10:40.37shamelessn00bhttp://www.the-asterisk-book.com/unstable/call-file.html
10:40.38drmessanoAudio
10:41.06DNDyup got it. i didnt know there was an advanced button underneath
10:41.08DNDthanks
10:42.21shamelessn00bChannelZ: I made this call file
10:42.24shamelessn00bChannel: SIP/3002
10:42.26shamelessn00bExtension: 1001
10:42.28shamelessn00bContext: default
10:42.30shamelessn00bPriority: 1
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10:43.07ChannelZand?
10:43.59shamelessn00bit gives me error
10:44.06shamelessn00berrors
10:44.09ChannelZwhich says...
10:44.10shamelessn00bdoesnt make any calls
10:44.13shamelessn00bwait
10:45.40shamelessn00b[Dec 29 10:44:29] NOTICE[10007]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
10:45.42shamelessn00b[Dec 29 10:44:29] ERROR[9850]: app_queue.c:1101 device_state_cb: Received invalid event that had no device IE
10:46.11shamelessn00bim sitting on the same machine calling the samemachine
10:46.55ChannelZwell if I had to guess SIP/3002 isn't configged right or something
10:47.17ChannelZor it's a single call device and you have it calling it's self and it's throwing an error or I-dont-know-what
10:47.57shamelessn00b<PROTECTED>
10:47.58shamelessn00b<PROTECTED>
10:48.00shamelessn00b[Dec 29 10:46:51] NOTICE[10028]: channel.c:3834 __ast_request_and_dial: Unable to request channel SIP/3001
10:48.02shamelessn00b[Dec 29 10:46:51] ERROR[9850]: pbx.c:9264 device_state_cb: Received invalid event that had no device IE
10:48.04shamelessn00b[Dec 29 10:46:51] ERROR[9850]: app_queue.c:1101 device_state_cb: Received invalid event that had no device IE
10:48.06shamelessn00b[Dec 29 10:46:51] NOTICE[10028]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
10:48.07shamelessn00bthats the whole thing
10:48.25shamelessn00bwhen I dial from a sip client like x lite
10:48.32shamelessn00bauthenticated with 3001
10:48.36shamelessn00bit works fine
10:49.01ChannelZis x-lite still running when you put the call file in?
10:49.06shamelessn00bno
10:49.17ChannelZwell then SIP/3001 doesn't really 'exist' then does it
10:49.35shamelessn00bthen I start getting calls on 3001
10:49.38shamelessn00bon my ip phone
10:49.40shamelessn00b:/
10:49.49shamelessn00bsoftphone rather
10:49.53ChannelZwell yea that's what dialing SIP/xxxx usually does....
10:49.58ChannelZcalls things...
10:50.20ChannelZYou were expecting what to happen instead exactly?
10:50.49shamelessn00bI want 100 calls to be generated
10:50.59shamelessn00bdont really want to listen to them
10:51.01ChannelZWell then you need 100 things to call.
10:51.14shamelessn00b100 instances tof xlite
10:51.16shamelessn00b:/
10:51.44shamelessn00bis there a workaround
10:51.45shamelessn00b??
10:51.46ChannelZYou can't call "nothing".  You're not really testing anything at that point.
10:52.13shamelessn00bI can put the call on some extension instead
10:52.15shamelessn00bthat just waits
10:52.17shamelessn00bcan I do that
10:52.19shamelessn00b??
10:52.30shamelessn00bdefine some context xyz
10:52.34shamelessn00bwait(6000)
10:52.39shamelessn00bsomethingsimilar
10:52.39ChannelZYes but you have to be calling FROM SOMETHING!
10:53.53ChannelZmaybe you can hack up something with Local channels, I dunno, you can go read about it: http://www.voip-info.org/wiki/view/Asterisk+local+channels
10:54.03ChannelZMe I have to go to bed so I can wake back up in 4 hours and go to work :/
10:55.40shamelessn00bI was up since 1 AM last night
10:55.49shamelessn00bwent to sleep at 2 am today
10:56.01shamelessn00bback at work at about 9 am
10:56.03shamelessn00bepic
10:57.56shamelessn00bgnite
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11:08.59lesouvageI'm calling an extension that is mend to list the status of all the devices in the cli with ",NoOP(The status of ${NR1002} attached to 1002 is ${DEVSTATE(SIP/${NR1002})})". The output for the device I use to call this extension is "NOT_INUSE" and that  is definitly wrong. What do I have to do to get the proper dev status as output?
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11:58.58ManxPower-workI hate mornings
11:59.42mort_gibManxPower-work: Yes, but working late into the night is just much worse!
12:00.05ManxPower-workmort_gib: Not for me.
12:00.32ManxPower-work<--- nocturnal
12:00.50mort_gib:-
12:00.55mort_gib:-)
12:01.08mort_gibI get enough of computer eventually!
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12:57.55benngardi must be stupid
12:58.18benngardi have a "test" line:
12:58.20benngardExecIf($[${LEN(${BLINDTRANSFER})}>0]?NoOp(${BLINDTRANSFER}))
12:58.46benngardtrying to use "CUT" so i just have the SIP/number left
12:59.11benngard...?NoOp(SIP/0317998975-000001c0)") in new stack
12:59.46ManxPower-workI'm not seeing your CUT statement
13:00.51ManxPower-work${CUT(${BLINDTRANSFER},-,1)}
13:01.14ManxPower-worksorry...
13:01.30ManxPower-work${CUT(BLINDTRANSFER,-,1)}  That should be correct.
13:03.32benngardlets try
13:03.46ManxPower-worknext time paste the broken statement
13:04.16benngardExecIf($[${LEN(${BLINDTRANSFER})}>0]?NoOp(${CUT(${BLINDTRANSFER},-,1)})) that didnt work so minimized it
13:04.38ManxPower-workas you can see that's not what I pasted.
13:04.46benngardno i see that
13:05.08ManxPower-workCut wants a variables name BLINDTRANSFER not a string ${BLINDTRANSFER}  It's an easy mistake to make
13:07.10benngardthx that did the trick
13:07.33benngarddidnt know that cut wanted a variable
13:07.43benngardExecIf($[${LEN(${BLINDTRANSFER})}>0]?NoOp(${CUT(BLINDTRANSFER,-,1)})) gave
13:07.47ManxPower-workbenngard: *nod*  It's sort of odd, but I'm sure there's a reason
13:08.43benngardthe correct answer
13:09.00benngardjust se whats happen when i replace noop with a dial then
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13:11.37Thornhello
13:11.42*** part/#asterisk alrs (n=lars@46.sub-70-213-201.myvzw.com)
13:12.34Thornhow do I write an interval from 22:00 dec. 31st to 08:00 jan. 1st in GotoIfTime()?
13:12.55ManxPower-workThorn: I don't think you can do that in one statemnet.
13:13.25ManxPower-workyou would want 22:00 to 23:59 and 0:00 to 08:00
13:14.06ThornManxPower-work: that was my initial idea too, thanks
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13:16.42benngardhmm, strange i did call 0317998975 from my cell phone, 0317998985 did transfer it to 0317998985 (didnt answer) got the call back and my cell phone "was talking" to 0317998975 again, perfect, but 0317998985 did continue to ring
13:16.56benngardexten => 0317998985-NOT_INUSE,1,Dial(SIP/0317998985,20,t)
13:16.57benngardexten => 0317998985-NOT_INUSE,2,ExecIf($[${LEN(${BLINDTRANSFER})}>0]?Dial(${CUT(BLINDTRANSFER,-,1)}))
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13:20.44benngardspli
13:20.44benngardsplit
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13:30.41DakonI've just upgraded from 1.4 to 1.6
13:31.06Dakonwhich is the supposed way to get app_rxfax and app_txfax again?
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13:35.09ManxPower-workDakon: why not use app_fax?
13:35.15ChainsawDakon: Enabling spandsp support.
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13:36.49ManxPower-workDakon: have you read the UPGRADE*.txt files, which contain all important changes to Asterisk?
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13:43.17voipmonkyawns
13:47.35ManxPower-workI guess Dakon didn't really want help.
13:48.33benngardweird, 985 calls 976 who blind transfer to 975, 975 starts to ring but dont answer call gets back to 976, but no voice between 985 and 976 and 975 continues to ring
13:49.00benngard<PROTECTED>
13:49.00benngard<PROTECTED>
13:49.23ManxPower-workcall doesn't "get back" to 976 unless you tell it to.
13:49.37benngardi do tell
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13:49.58benngardthats was what i struggled with before
13:50.18Dakonsorry, was distracted by work
13:50.21benngardexten => 0317998975-NOT_INUSE,2,ExecIf($[${LEN(${BLINDTRANSFER})}>0]?Dial(${CUT(BLINDTRANSFER,-,1)}))
13:51.18ManxPower-workdialplan snippets are pretty useless to us
13:51.34ManxPower-workDakon: What are the answers to my two questions?
13:51.38benngardthe get back works
13:52.33ManxPower-workDakon: come back when you have time to focus on your question
13:52.35Dakon"was not aware of", yes
13:52.56ManxPower-workRead those UPGRADE*.txt files.
13:57.02benngard"Asked to transmit frame type slin" but i have hardcodec alaw everywhere...
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14:00.03Dakonwhere do I find app_fax.c? core asterisk?
14:00.32ManxPower-workDakon: yes.  but if you don't have spandsp installed you won't be able to select it.  Use menuselect.
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14:00.37*** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk) [NETSPLIT VICTIM]
14:00.37*** mode/#asterisk [+o angler] by irc.freenode.net
14:03.36ManxPower-workNow go read the UPGRADE.txt files
14:04.09Dakonspandsp 0.0.6_pre12
14:04.13*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)
14:04.15*** join/#asterisk roo (n=w7@freenode/staff/wikimedia.werdan7)
14:04.21Dakonjust to note that the UPGRADE files say basically nothing about fax
14:04.25ManxPower-workread them all since something in, for example, in the UPGRADE12 file may say an application was depricated.  It won't me mentioned again when it's removed in 1.6, for example
14:04.51*** join/#asterisk sebbl (n=Momofu@109.192.162.148)
14:05.39*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:06.36benngard${BLINDTRANSFER} just works for sip, correct?
14:07.35Dakonhow intuitive
14:09.57ManxPower-workbenngard: what does channelvariables.tex tell you about BLINDTRANDFER?
14:10.17ManxPower-workI think the asterisk.pdf that's usually built with Asterisk will also have that information in it.
14:12.53ManxPower-workDakon: I think the entire documentation system in Asterisk is crap.
14:14.55coppiceO'Reilly might disagree :-)
14:14.58*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:14.58*** mode/#asterisk [+o leifmadsen] by ChanServ
14:15.04*** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk)
14:15.11_cgchi everyone
14:15.37Dakonis there any documentation on app_fax? Or can I just switch any app_[rt]xfax to app_fax and expect that working?
14:16.16_cgcdoes anyone know when you record a call in asterisk 1.6.1.11 it records it as 2 different files, 1 for each side of the conversation?
14:16.52*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:16.52*** mode/#asterisk [+o leifmadsen] by ChanServ
14:17.05_cgci want the recording to be as 1 file ${UNIQUEID}.wav, but instead i get ${UNIQUEID}-in.wav and ${UNIQUEID}-out.wav
14:20.24ManxPower-work_cgc: Use MixMonitor
14:21.18_cgc<ManxPower-work> Is this new then because I just used Monitor before?
14:21.30ManxPower-work_cgc: It was new in 1.2
14:21.41_cgcahh lol, ok cool, thanks
14:21.48ManxPower-work"core show application monitor" and "core show application mixmonitor"
14:22.25*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com)
14:22.32Kattyguten morgan
14:22.54benngardmust be blind :( i am reading channelvariables.tex...
14:23.43benngardAsterisk standard channel variables
14:23.55benngard${BLINDTRANSFER}         The name of the channel on the other side of a blind transfer
14:24.10ManxPower-workdoes it mention anything about being SIP specific?
14:24.53Kattyvie heist DU
14:24.54benngardno but when i did try to transfer a call that came in on a h.323 channel, that parameter was empty
14:25.18benngardso i thought it was sip only
14:25.33ManxPower-workChances are H323 is a special case.
14:25.43Kattyhat jemand deutsch sprechen?
14:25.46benngardhow should i know that? ;)
14:26.03ManxPower-workbecause H323 support in Asterisk sucks and nobody uses it if they have any choice in the matter?
14:26.10Kattymein telefon funktionert nicht
14:26.43benngardi know that h323 sucks, did got a patch for ooh323 gonna test and recompile later
14:26.53Kattywas kannich tun?
14:27.22DakonStecker rein
14:27.43Kattyhehehehe
14:27.50Kattyplugggggggg it in plug it in!
14:31.10_cgcManxPower-work> thanks, that worked :)
14:31.58*** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca)
14:31.58tzafrir_laptop_cgc, moved to sip?
14:31.58ManxPower-workLooks like we have our usual "at least 5 clients are down because of Verizon problems" again today
14:32.02*** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk)
14:32.11ManxPower-workHow many fsckin' cable cuts can they have in a fsckin' week?
14:32.11KattyDakon: es nun, was soll ich tun?
14:33.29ManxPower-work_cgc: "core show applications" and "core show FUNCTIONS" are your friend.
14:33.49shamelessn00bHURRRR
14:33.53shamelessn00bfound the fkin error
14:33.55ManxPower-workBTW, functions themselves are uppercase
14:34.25DakonKatty: keine Ahnung
14:34.44ManxPower-workSometimes I think Verizon dispatches a tech with an axe a couple of times a week to cause "massive outage"
14:35.03KattyDakon: uber die lippen gebracht glanz
14:35.30shamelessn00bthe database server had the connection limit set to 100
14:35.40shamelessn00bthats why my processorfkinshot up to 100 percent
14:35.49shamelessn00bright after it reached the 100 call mark
14:35.50*** join/#asterisk jackal (n=jackal@pool-96-247-205-46.clppva.fios.verizon.net)
14:35.52KattyDakon: how do you pronounce Ahnung?
14:36.16ManxPower-workAhnung?  Sounds like a porn star name.
14:36.23x86lol
14:36.58x86Katty: du spreche zie Deutsch?
14:37.00KattyManxPower-work: http://gangstaname.com/porn_name.php
14:37.06Kattyx86: just a little (=
14:37.14x86Katty: nifty :)
14:37.26Kattyprobably not enough to get by without a proper translator
14:37.39x86Katty: I've learned (a little) german from the guys over in #theplanet (here on this network)
14:37.42Kattyi'd be telling everyone to slowwwwww down
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14:37.55x86it's a native german-speaking channel..
14:38.02x86yeah
14:38.21x86I can barely read it proper and less than barely write it
14:38.34x86like, how the hell am I supposed to know that a tree is male? :)
14:38.34KattyManxPower-work: hilarious, i put in Angela and it says my name is Kitty Jam
14:38.44x86hahahaha
14:39.11KattyAngie is Asia Jizz ^_-
14:39.20Kattyx86 is Sweat Darkholer LOL
14:39.27x86ROFL
14:39.34x86darkholer... wow
14:39.40Kattymanxpower is corporal fuegobutt
14:39.49KattyManxPower-work: i didn't know you were a corporal!
14:42.08Kattyi really hate one of my senators.
14:42.17Kattyhe keeps voting no on all the important bits.
14:43.05drmessano<--- BJ Jiggler
14:43.12drmessano:(
14:43.21Kattyah ahahhaaha
14:43.21drmessanoThat hits too close to home for me\
14:43.55DakonManxPower-work: I replaces rxfax(...) by ReceiveFAX(...) and txfax(...) by SendFAX(...) and it sort of works
14:44.29Dakonit crashes when receiving faxes and has strange problem calling my script, but at least the tiff file looks correct
14:44.40drmessanoThe PDFs for the Digium apps document the differences well
14:44.45jayteeI hate words and phrases like "sort of" and "kinda"
14:45.00Kattyi sort of kinda do too
14:45.05jaytee:-)
14:45.10Kattycept, sorta kinda
14:45.14drmessanoI'm kinda... slow on that one, thanks Katty
14:45.55Dakoncu
14:45.55Kattyjaytee's porn star name...wow.
14:46.00Kattyi'm not even gonna type that one here
14:46.21drmessanoLemme guess...
14:46.30drmessanoDolph Lundgren
14:46.44benngard;)
14:47.07drmessanoThat would be jaytee's porn name
14:47.44benngard== Dick Maxim :)
14:48.30ManxPower-workManx Power IS my porn name.
14:49.53KattyMinx Power.
14:50.07drmessanowow
14:50.12drmessanowhat a difference a space makes:
14:50.18drmessanoleifmadsen -> Corporal Muffmuncher ..... leif madsen -> Humpy Jizz
14:51.06drmessanoProving once again that real life sucks worse than IRC
14:52.35x86heh
14:52.44x86corporal muffmuncher... rofl
14:53.26Kattyso many critters in my yard this morning.
14:53.34Kattyyou'd think the apocolypse happened
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14:54.26Katty5 squirrels so far
14:54.31Kattyand it's only 9!
14:54.54x86Katty: they are probably just gather food and stuff for when 12-21-12 rolls around... you know, then end of the world
14:55.21Kattythey probably ARE gathering food for 12/21, but not because it's the end of the world :P
14:55.30ManxPower-workx86: I'm pretty sure they just know there's free food.
14:55.42Kattyand who doesn't like free food?
14:55.57Kattyinfobot: forget Critter Cam
14:55.57infoboti forgot critter cam, Katty
14:56.02Kattyinfobot: forget crittercam
14:56.02infobotKatty: i forgot crittercam
14:57.55Kattyinfobot: critter cam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d
14:57.56infobotKatty: okay
14:58.06Kattyinfobot: crittercam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d
14:58.07infobotokay, Katty
14:58.53*** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie)
14:59.14Kattya 6th just showed up
15:00.41*** join/#asterisk gme30066 (n=gme@173.160.69.30)
15:00.43Katty8
15:01.17Kattyit's like i'm supporting squirrely welfare.
15:01.30Kattyfood stamps for the children.
15:02.22ManxPower-workDon't you mean "rats with bushy tails welfare"?
15:02.36Kattyrats are smarter than squirrels.
15:02.57Kattyrats are cute.
15:04.38*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
15:04.42Kattyhi gr0mit
15:04.53*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
15:04.59Kattyhi _ShrikE
15:05.06gr0mithi Katty
15:05.10gr0mityou ok?
15:05.18Kattywell that's debatable.
15:05.20*** part/#asterisk gme30066 (n=gme@173.160.69.30)
15:05.24gr0mitoh dear!
15:05.31Katty:P
15:05.44gr0mitlooks at the pouring rain
15:05.47*** join/#asterisk Akiraa (n=Akiraa@79.112.17.211)
15:05.59gr0mitand wishes he was somewhere other than .uk
15:06.01Kattygr0mit: :<
15:06.13Kattygr0mit: there's some sunshine here. i will bottle it up and ship some to you!
15:06.21Kattygr0mit: first sunshine in what feels like weeks.
15:06.21gr0mitwhere is .here?
15:06.25leifmadsendrmessano: o.O
15:06.27Kattycentral usa
15:06.35gr0mitany particular bit?
15:06.39Kattyhi mister madsen
15:07.02*** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl)
15:07.09leifmadsenms. katty
15:07.28gr0mitcentral as in mid-west?
15:08.04Kattygr0mit: 36 degrees 18'33" N, 89 degree 32'47" W
15:08.18x86heh
15:08.18Kattygr0mit: yes, midwest. southern missouri
15:08.23x86http://icanhascheezburger.files.wordpress.com/2008/06/funny-pictures-dueling-lobsters.jpg
15:08.27x86rofl
15:08.30gr0mitaah ok.
15:08.42gr0mithas never been there. spent a lot of time in chicago
15:08.42Kattyx86: that's awesome.
15:08.55Kattygr0mit: there's mostly trees and cattle here.
15:08.59Kattygr0mit: oh, and corn.
15:09.05Kattygr0mit: fields and fields of corn. everywhere.
15:09.19gr0mitonce you leave the suburbs of northern chicago, thats the same!
15:09.23leifmadsenKatty: but not corn syrup funny enough
15:09.34Kattyleifmadsen: they probably turn it into corn syrup
15:09.41Kattyleifmadsen: it's mostly field corn, afterall.
15:09.42leifmadsen:)
15:09.51leifmadsenguess I'm not funny this early in the morning
15:09.54Kattyor livestock feed
15:10.09x86Katty: http://icanhascheezburger.files.wordpress.com/2008/05/funny-pictures-technical-support-cat.jpg
15:10.30Kattyahahaa
15:10.34x86yeah ;)
15:10.43Kattyi'm totally goign to answer my phone like that today
15:10.50gr0mitwishes he was wearing his "No, I will not fix your daughter's laptop" T-shirt
15:10.54x86i'm gonna print that out and tape it up somewhere in the helpdesk area at work
15:11.25ManxPower-workoffers to let gr0mit borrow his "No I will not fix your computer" t-shirt
15:11.51gr0mitputs it on, realising it is too late. Vista re-install nearly complete
15:12.14gr0mithumph. 96 important updates.
15:12.26Kattyi used to have some thinkgeek shirts.
15:12.40Kattyhad one that had PI on the front, and then there's no place like 127.0.0.1
15:12.47gr0mitlol!
15:12.53ManxPower-workI have the Pi one.
15:13.02*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
15:13.15Kattythe st. louis science center has a shirt that says PI on the front, and has a scoop of icecream on top of it
15:13.32Kattydoes it come in girl sizes tho :<
15:13.45Kattys/does/doesn't/
15:14.28gr0mitstruggles wuth a 900 metre wifi link
15:14.34ManxPower-workI used to have this one: http://www.t-shirthumor.com/Merchant2/graphics/fullsize/flam_lg2.gif
15:14.58Kattyomg
15:15.00Kattyeww lol
15:15.03x86gr0mit: why struggle? get an amplifier ;)
15:15.04ManxPower-workLOL!
15:15.17Kattythat's just WRONG!
15:15.23Kattyretweets it
15:15.36gr0mitwell, our telco is installing FTTC here
15:15.43ManxPower-workgr0mit: Have you looked at defactowireless.com  I get all my long range gear
15:15.46ManxPower-workfrom them
15:15.50gr0mitbut the rollout stops 900 metres from my house
15:16.00gr0mitso have installed a link to a friends house
15:16.09ManxPower-workThey sell 600mw APs
15:16.15gr0mitand will get the VDSL installed there
15:16.21ManxPower-work14dBi gain antennas, enclosures, etc
15:17.13gr0mithttp://www.compex.com.sg/DownLoads/Manual/UM-MMJ543.pdf is wot i got
15:17.16x86gr0mit: http://www.l-com.com/item.aspx?id=22137
15:17.50gr0mitfears the RF police!
15:17.59ManxPower-workx86: lcom has some pretty cool too.
15:18.00gr0miti am running at 5.5ish GHz
15:18.12ManxPower-workHello Rain fade!
15:18.14Kattywho was it that recommened i watch Marine 2 last night?
15:18.32gr0mithmm its all ulaw-ish
15:18.53gr0mit5.5 GHZ is not so prone to raing, right?
15:18.58gr0mitor wrong?
15:19.29ManxPower-workgr0mit: I'm not an expert.  I think most all frequencies in the microwave range will have rain fade.
15:20.36gr0mithttp://www.flickr.com/photos/13418468@N07/
15:20.48x86ManxPower-work: used to be hyperlink technologies back when I used them
15:21.10Kattyis that a solar panel?
15:21.13x86ManxPower-work: I used to manage a small WISP and we used all hyperlink antennas / amps / cabling / everything... loved them
15:21.14Kattyor just a window
15:21.27gr0mitVelux roof window
15:21.32Kattyk
15:21.40gr0mitwith my 5Gz antenna mounted outside
15:22.07Kattysounds pretty schnazzy
15:22.31gr0mitit was really cheap, like £10 for the pair
15:22.35ManxPower-workI have something like this that I need to install http://shop.defactowireless.com/core/media/media.nl?id=18793&c=300197&h=140ccd8debf949f6e937&resizeid=-2&resizeh=340&resizew=240
15:22.35gr0miti mean £120
15:22.51shamelessn00boh yeah
15:23.01shamelessn00bhandling 500+ calls
15:23.03shamelessn00bjust awesome
15:23.05ManxPower-workx86: I think I bought some ethernet surge supressors from them.
15:23.08shamelessn00ball calls executing agis
15:23.15shamelessn00bherp derp
15:23.44gr0mitam planning to operate a small wisp
15:23.45ManxPower-workWhen you run 1000 ft of underground cable thru iron saturated soil on a mountain you need ethernet surge supressors.
15:23.51gr0mitto our street
15:24.13gr0mitgot pppoe logins to a mikrotik box
15:26.24AkiraaWhat does a cable with a ring capacitor do?
15:26.50gr0mitin UK?
15:27.14Akiraaanywhere, but I find them on a UK distribuitor
15:27.22gr0mitok, it is a UK thing
15:27.41Akiraawhat is it used for?
15:27.53gr0mithttp://www.wppltd.demon.co.uk/WPP/Wiring/UK_telephone/uk_telephone.html
15:29.47gr0mitwas used on old pulse dial phones to stop bells on other extensions rattling when you dialled
15:29.53x86ManxPower-work: we never worried about lightning on the ethernet side... we'd run fiber from our switches to the AP... we'd just use a converter on the AP side that didn't have a fiber connection
15:30.19x86ManxPower-work: so if the AP got hit, that's as far as it could go (well, and the media converter)
15:30.34gr0mitAkiraa, you probably don't need whatever it is you are looking at!
15:33.26Akiraagr0mit: thanks, just wondered what it was
15:39.03coppiceThe UK tried to made it hard for foreign suppliers by making their phone system incompatible with the rest of the world - a dumb and unsafe plug/socket and a stupid third wire arrangement
15:39.40gr0mitunsafe?
15:40.39coppiceyep. it fails international safety standards
15:40.55coppicea child's finger can go in
15:41.18coppicethe RJ11 is a little smaller, and passes that test
15:41.40gr0mitaah yes i recall when i worked at BABT
15:41.48gr0mitit was all a mess
15:41.56coppicefor a long time, until they cooked the standards in the UK, new sockets couldn't get approval :-)
15:42.30gr0mitindeed so.  They needed to be recessed by approx 3mm
15:42.31Corydon76-digcoppice: Dunno, maybe it's Darwin at work?
15:42.45gr0mitin order for a finger not to get access to the contacts
15:45.08coppiceHK used that stupid system, too, which is odd, because HK mostly followed US telephony practice
15:45.16*** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
15:46.18ManxPower-workcoppice: I thought the 3rd wire was a common ground so your tx path and rx path don't go on the same wire.
15:46.36ManxPower-workWhich honestly sounds like an awesome idea from the standpoint of echo
15:46.39gr0mitManxPower-work, nope
15:47.02gr0mitit just separates the 100v ring current
15:47.03coppiceManxPower-work: it might do, until you understand some engineering :-)
15:47.11ManxPower-workToo logical for the Brits?
15:47.25gr0mit3rd wire does not go back to the exchange
15:47.37gr0mitonly appears in the Master socket
15:47.40coppicethe third wire in the UK only passes around the phones within a house
15:48.01Kattythis tea tastes attrocious
15:48.13gr0mithands Katty some Earl Gray
15:48.24*** join/#asterisk sebbl (n=Momofu@109.192.162.148)
15:48.26Kattygr0mit: iced tea.
15:48.32benngardprefer the old swedish system
15:48.39Kattyi'm not sure how earl gray would taste cold ^_-
15:48.59gr0miticed tea?
15:49.02gr0mitshudders
15:49.14coppiceKatty: he has been dead for years so he's probably pretty cold
15:49.44Kattyhehehe
15:49.45Kattynice.
15:49.47Kattyapplauds coppice
15:50.06benngardany1 know if u can find may213 on any irc channel?
15:50.36ManxPower-workbenngard: type /whois may213
15:50.51Kattyyou can also ask nickserv for info
15:50.58Kattyit'll tell you the last time they were online
15:51.04Kattysomething like /msg nickserv info Katty
15:53.07eppigyschooches closer to Katty
15:53.53Kattyhello deary!
15:53.55Kattyhugs eppigy
15:59.41eppigyhiya
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16:06.01*** join/#asterisk bpgoldsb (n=bpgoldsb@ip24-250-198-162.ga.at.cox.net)
16:06.33bpgoldsbHas anyone run Asterisk under Xen and can comment on the performance of it?
16:10.41bmoracabpgoldsb, i prefer VMware ESXi
16:11.04*** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com)
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16:12.31murraytmcan someone tell me what compile time options were used for the 1.6.0.20 asterisk that's in the asterisk-current yum repo?
16:13.46Chainsawmurraytm: The .spec file, hopefully?
16:14.34bmoracawhy worry about it?  just download the source and compile it yourself.  it's not exactly rocket surgery
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16:15.35Kattyeppigy: you would not believe how difficult it is to find boots :<
16:15.44Kattyeppigy: i think it's a conspiracy
16:16.29jayteerocket surgery? like a gastric bypass on a Saturn 5?
16:16.42murraytmi'm basically trying to get a couple of modules to load in the one installed from yum and would rather not have to reinstall all of asterisk to do it
16:17.02murraytmwhere could i find the .spec file?
16:19.17bmoracawhat modules?  if they're not statically linked, you'll need to compile those modules anyway
16:19.45murraytmcustom volume function and modified res_agi
16:19.54ManxPower-workRemmember almost nobody will help you if you compile from packages.
16:20.34ManxPower-workKatty: your local army/navy surplus may have a good selection of boots.
16:20.34[TK]D-FenderManxPower-work: I don't recall mention of RPMS
16:20.52ManxPower-work[TK]D-Fender: sorry, I thought a yum-repo had RPMs.
16:21.10murraytmi'm trying to come up with a fast and reliable way to migrate a system running 1.4 and installing by yum, then dropping in some binaries seems like a quicker way than recompiling all of asterisk
16:21.21bmoracamurraytm, if they're statically linked, just drop the files in the modules directory and restart asterisk (or use the module load command)
16:21.30*** join/#asterisk jmacz (n=jmacz@186.80.77.231)
16:21.36[TK]D-FenderManxPower-work: (source) RPM's....
16:22.06[TK]D-Fendermurraytm: Compiling * takes what... 5 minutes maybe?
16:22.14bmoracamurraytm, compiling really doesn't take that long.  and, like i said, unless those binaries you have were statically linked, you won't be able to use them anyway
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16:22.41murraytmmaybe i'm just being paranoid.  seems like a lot can go wrong during compilation compared to just installing from the repo.
16:23.50bmoraca./configure will tell you if you're missing anything.  make menuselect lets you choose exactly what you want and won't let you choose what you don't have prereqs for...it's a fairly brainless process anymore
16:24.13murraytmit's not the compiling, it's the making sure the install works after it's compiled that i'm worried about
16:24.18bmoracawell, it will let you choose ilbc if you don't have prereqs...so, i'd advise caution around that one
16:24.19KattyManxPower-work: no, i mean dress boots.
16:24.23KattyManxPower-work: just plain NORMAL dress boots.
16:24.31KattyManxPower-work: without all this extra buckle frill fur stuff
16:24.32bmoracamurraytm, make install puts everything where it needs to go
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16:24.57murraytmi did 1.4 from source so i'm familiar with the process, just was thinking there might be a better way
16:25.21bmoracamurraytm, if you need custom modules, there is no other way
16:25.25eppigyKatty: :<
16:25.33murraytmok, well that settles it then.  thanks. :)
16:25.33eppigyi have faith that you will find your boots
16:25.38HolisterI was having problems with my DAHDI card (it made very wierd noises when a call was connected), so I rebooted. Now I can't get the card to initialize, and port 1 doesn't have a green light. Is the card done for?
16:26.23Kattyeppigy: i will :>
16:26.24Kattyeppigy: but i'm going to complain in the meantime!
16:26.24bmoracamurraytm, installing on CentOS?
16:26.24ManxPower-workHolister: can you power cycle the machine?
16:26.24murraytmyes
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16:26.45HolisterManxPower-work: would be difficult, but I could try. The problems started when I power cycled
16:27.07benngardavaya - ooh323 - asterisk is not the working :(
16:27.10HolisterManxPower-work: well, not all of the problems, but not being able to initialize the card, and the light goinbg off
16:27.10bmoracamurraytm, there's a fairly replete guide on voip-info for installing 1.6 on CentOS...at the very least, it'll give you all the prereqs you need
16:27.14ManxPower-workHolister: you can also contact tech support.  This should be somethnig they will support.
16:27.39HolisterManxPower-work: the manufacturer of the card?
16:27.42murraytmok, thanks, i'll have a look through that
16:28.03ManxPower-workHolister: correct.  I assume it's Digium since you didn't tell use the make or model of the card.
16:28.16HolisterManxPower-work: it is TDM400
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16:28.52ManxPower-workContact Digium then.
16:29.10ManxPower-workHolister: also remember that if you upgraded your kernel then you need to recompile and reinstall DAHDI
16:29.12Holisterthx
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16:29.22Holisterhmmmm
16:29.28Holisterwell modprobe appears to work
16:29.31torrancewwould anyone have a recommend a good, affordable, cordless IP phone?
16:29.41ManxPower-worktorrancew: There are none.
16:29.59ManxPower-worktorrancew: But you can easily use a SIP ATA + your favorite cordless phone.
16:30.45torrancewwould the Linksys PAP2 suffice for that?
16:30.50AkiraaAre there licensing limitations in deploying IAX2 devices?
16:31.01ManxPower-worktorrancew: yes
16:31.23torrancewand what if i were willing to compromise on the affordable end of cordless IP phone?
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16:31.42ManxPower-workAkiraa: no, but there are copyright issues with using things like Asterisk, Digium, etc in marketing.  Digium has a policy on this.
16:31.42torrancewdoes that change the situation at all?
16:31.47bmoracatorrancew, i recommend the Philips CD1 phones...they support CID and have a MWI that works really well with the Linksys PAP2T and the SPA8000
16:31.52ManxPower-worktorrancew: Define "affordable"
16:32.04torrancewthat's negotiable - it's for a business
16:32.18ManxPower-work~phones
16:32.19infobotWhile personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones.  Do not consider Cisco phones.  Ever.
16:32.21torrancewreliable and relatively portable would be the concern
16:32.28ManxPower-workYou'll notice there is no cordless model listed.
16:32.38bmoracatorrancew, the Polycom cordless phones are the only ones I'd trust and they're NOT cheap.  Snom makes one that's more affordable, but doesn't work as well
16:32.39torrancewtouche'
16:32.48torrancewthanks all
16:32.52bcrispi want a wrist watch phone 007 style
16:33.01torrancewbcrisp: agreed
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16:33.16Kattybcrisp: LASER PEWPEWPEW
16:33.16anonymouz666what's the problem with Cisco phones?
16:33.19Chainsaw*G* Neat. Cisco & Grandstream swapped?
16:33.34Kattyanonymouz666: they're just very expensive
16:33.43Kattyanonymouz666: but they do have a lot of very nice features.
16:33.44Chainsawanonymouz666: Firmware hidden behind a paywall, bleak XML support on the SIP firmware...
16:33.45anonymouz666Katty: indeed but works
16:33.53Kattyanonymouz666: i'm particularly fond of cisco's blackberry call precense app
16:33.57Kattyprescense
16:33.59Kattywhatever
16:34.02Kattyi can't spell that word
16:34.05Chainsawpre-sense?
16:34.08KattyPRESENTS
16:34.15Chainsawpresence :)
16:34.16ChannelZyay!
16:34.26Kattyi also can't spell convience
16:34.30Kattycon viene ence
16:34.36torrancewKatty: convenience
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16:34.39Kattywhatever.
16:34.40torrancew:-)
16:34.49coppiceKatty: I'm particularly fond of Blackberry and Apple Pie
16:34.54Kattyi should make them a password for something.
16:35.02Corydon76-digKatty: phonetics FTW
16:35.03Kattycoppice: blackberry cobbler's some mean stuff.
16:35.23Kattypie FTW.
16:35.45Kattythere is something terribly wrong with spring fashion trends.
16:36.01Kattyit all looks /retarded/
16:36.19Kattyit's all tunics, and fluffy frills.
16:36.43ChannelZfashion is retarded
16:37.32Kattyhave you seen the shoes?
16:37.33Corydon76-digUh, you're talking to an audience of geeks.  Fashion is not really something that any of us take into account.  Jeans and tshirts, remember?
16:38.05[TK]D-Fender.. and the occasions Storm Trooper uniform
16:38.05Kattyhttp://images.bakersshoes.com/images/products/1_102526_FS.JPG <- seriously? come on now.
16:38.18KattyCorydon76-dig: oh right. yeah.
16:38.38Kattythere's an idea, i will just get a ton of star trek uniforms.
16:38.40Corydon76-digBonus if the Tshirt was free, because a vendor gave it to us
16:38.55torrancewManxPower-work: how feasible would it be to run my own STUN server to help with using SIP extensions from off-site locations?
16:38.59Corydon76-digExtra bonus if the Tshirt is clean
16:39.06torrancew(have an in-house asterisk server)
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16:40.44SuPrSluGtorrancew: there are plenty for free use. Why spin your own?
16:40.54ManxPower-worktorrancew: seems like quite a bit of work for almost no gain.
16:41.10torrancewcompany doesn't want to use an external server
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16:41.29torrancewthough if someone here can put up a better argument i can take to The Man for why it's no big deal, i'm game
16:42.05torrancewthe company in question is an IT consulting firm, specializes in Macs.
16:42.33ManxPower-workWhy do you need a STUN server?
16:42.49torrancewthey want to make SIP calls from off-site with their iPhones
16:43.39ManxPower-worktorrancew: OK.  You DO NOT NEED STUN WHEN USING NAT
16:43.57bcrispATT is supposed to open up for SIP on their network
16:44.03torrancewah, i misunderstood then
16:44.18torrancewdon't you need open ports for RTP though?
16:45.13torrancewat both networks, that is
16:46.06ManxPower-worktorrancew: You only need to port forward on the NAT for the Asterisk box.  No port forwarding needed (and in fact may break stuff) on the Phone NAT router.
16:46.17torrancewah i see
16:46.22SuPrSluGthe client shouldn't have to. you will need to port forward SIP 5060 &  RTP 10000-20000 to the * box
16:46.34torrancewright
16:46.47torrancewok
16:46.52torrancewi'll explore that a bit
16:46.53torrancewthanks
16:48.57torrancewone last question - going through a PAP, would I lose any of the key asterisk features? i'd assume most of them will be fine, since alot of that is handled by extensions and dial plans, but would like to hear from experienced minds
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16:51.37SuPrSluGnot asterisk features, but phone features
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16:52.34torrancewSuPrSluG: such as?
16:53.00thieums93Hi, do you know how to transmit voicemailuser to callback / dialout voicemail context ?
16:53.11SuPrSluGone touch dialing for features like voicemail or extensions
16:53.40SuPrSluGnot deal breakers or anything
16:54.09[TK]D-Fenderthieums93: huh?
16:54.31thieums93not clear ?
16:54.47[TK]D-Fenderthieums93: No.
16:55.33torrancewSuPrSluG: thanks
16:55.47SuPrSluGbetter codecs like hd voice too, that's nice
16:57.00Kattyjeebus, the lighthouse feeder has gone down 50% since this morning
16:57.08Kattyand i believe the suet block is officially GONE
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17:00.53thieums93a user is authenticated on voicemailmain, press 3(advanced option) then 4(dialout), jump to the context defined by dialout= in [general] (voicemail.conf) . I want to get in my dialout context the authenticated mailbox account, is that possible ?
17:01.28Kattydid you check to see if that's a variable?
17:01.43thieums93yep, nothing on wiki
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17:02.52[TK]D-Fenderthieums93: set the "dialout=" in the box's definition.  Most parms can be used in [general], and in the box definition itself
17:03.18cosmicwombatAnyone know of a common reason callers on hold get dropped after 30 seconds... everytime ?
17:06.37Kobazcosmicwombat: misconfiguration
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17:07.20Kobazcosmicwombat: it could be the calling party, it could be the phone, it could be dialplan
17:07.21cosmicwombatKobaz: where should I look ?
17:07.35Kobazare you using sip phones?
17:07.36[TK]D-FenderKobaz: Umm... nope
17:07.42Kobazpaste the sip debug
17:07.46Kobaz[TK]D-Fender: nope what?
17:07.51[TK]D-FenderKobaz: Not dialplan.
17:08.08cosmicwombatI think firewall is at play
17:08.10Kobaz[TK]D-Fender: if he's got some dialplan code in the background killing calls after 30 seconds, that could be it
17:08.11[TK]D-Fendercosmicwombat: Common cause for this is a bithy device complaining about not getting RTP while on hold
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17:08.25ManxPower-workthieums93: don't look at the Wiki, look at /path/to/src/asterisk/doc/tex/channelvariables.tex
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17:08.48[TK]D-FenderManxPower-work:  .... he's not talking about channel vars...
17:08.49Kobaz[TK]D-Fender: you're always the one that encourages people to not overlook stuff :P
17:09.02[TK]D-FenderKobaz: Hey look... its ELVIS!
17:09.08Kobazheh
17:09.13[TK]D-Fenderpoints the other way then runs while Kobaz is distracted
17:09.27cosmicwombatThank you very much , uh huh
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17:09.36[TK]D-Fendercurls his lip
17:10.22benngardDec 29 18:04:52 sip kernel: [2146494.253819] asterisk[14647]: segfault at 94 ip b7cc49a0 sp b62d3204 error 4 in libpthread-2.7.so[b7cbd000+15000] :(
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17:11.44jblackbenngard: Not good at all.
17:11.52benngardnup
17:11.55jblackat least you have a good hint of where to start.
17:12.09benngardyupp
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17:12.25benngarddid know what i did so i think i do a "rollback"
17:12.40jblackactually, maybe not. being pthread, it could be a lot of things.
17:12.54jblackpthread may just be catching it.
17:13.15benngardyes but i know what i changed so i "hope" i can go back
17:13.29ManxPower-workWhat did you change?
17:13.57benngardi went from h323 channel to ooh323 channel
17:14.25benngardi did some tests with may213, gave hime some logs
17:15.24benngardand ofc some dumb asshole at my office put some live traffic on my lab box so i have to do some reversed work later
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17:17.53benngardi have may213 in a private chat, gonna give him access to the box if he wants to have it
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17:22.14cuscohi...
17:22.55cusco[Dec 29 17:19:25] WARNING[13670] /home/murf/asterisk/1.6.1/main/ast_expr2.y: non-numeric argument
17:23.41cuscoI can't find the why
17:23.45cuscowhat argument is it referring to
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17:24.40ManxPower-workcusco: make sure you have bison and yacc installed
17:25.11_cgcdoes anyone have any experience with liz for sugar?
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17:25.59_cgcbut on the asterisk side obviously
17:26.03ChannelZanyone bored and want to try a test for me?
17:26.06ManxPower-work_cgc: did you try asking on #SugarCRM?
17:26.09cuscoManxPower-work: bison is installed. yacc calls bison
17:26.42TheDavidFactor-HChannelZ what do you need?
17:26.50_cgcManxPower-work: its asterisk i need help with though, not sugar
17:27.36ChannelZTheDavidFactor-H: Try playing a .gsm file while on a ulaw transport and see if it sounds right
17:27.36_cgcManxPower-work: i had it working before, but for some reason it's stopped working with the configuration I had, and not too sure why...
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17:28.02TheDavidFactor-Hwhat version of *?
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17:28.29ChannelZTheDavidFactor-H: Any - what are you running?
17:29.00TheDavidFactor-Htrunk, 1.6.1.6, 1.4.20, 1.4.23.1, and one or two others
17:29.00_cgcits related to dialing queues, when I change the queue from member => SIP/phone1   to  member => Local/1@phones it says all the phones are invalid
17:29.27ManxPower-workDo you have an exten => 1 in the [phones] context in extensions.conf?
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17:29.30ChannelZTheDavidFactor-H: I'd be interested in 1.6.1.6 and 1.4.20
17:30.04_cgcyep exten => 1,1,Macro(callphone,phone1,${EXTEN},SIP)
17:30.11_cgcand in there is:
17:30.14ChannelZTheDavidFactor-H: Something odd is happening to me on 1.6.1.12 and I reverted to 1.4.27.2 to test and it was doing it then too
17:30.31ManxPower-work_cgc: "dialplan show" and confirm that
17:30.39ChannelZTheDavidFactor-H: But I know it worked once upon a time when I originally built the system (but I don't even remember what version of 1.4 that was under, couple years ago now)
17:31.11_cgcchatterbox*CLI> dialplan show phones
17:31.11_cgc[ Context 'phones' created by 'pbx_config' ]
17:31.11_cgc<PROTECTED>
17:31.11_cgc<PROTECTED>
17:31.11_cgc<PROTECTED>
17:31.12_cgc<PROTECTED>
17:31.14_cgc<PROTECTED>
17:31.16_cgc<PROTECTED>
17:31.18_cgc<PROTECTED>
17:31.20bcrispstop flooding
17:31.22_cgc<PROTECTED>
17:31.23ChannelZ_cgc: USE PASTEBIN.CA
17:31.24_cgc<PROTECTED>
17:31.26_cgc<PROTECTED>
17:31.28_cgc<PROTECTED>
17:31.30_cgc<PROTECTED>
17:31.31TheDavidFactor-H~pastebin
17:31.32infobot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
17:31.32_cgc<PROTECTED>
17:31.34_cgc<PROTECTED>
17:31.34bcrisp...
17:31.36_cgc<PROTECTED>
17:31.38_cgc<PROTECTED>
17:31.40_cgc<PROTECTED>
17:31.42_cgcsorry
17:31.43ChannelZgrrrrrr
17:32.09_cgchttp://pastebin.ca/1730876
17:33.02sektorNBAahh
17:33.18_cgchttp://pastebin.ca/1730877
17:33.31QwellChannelZ: https://issues.asterisk.org/view.php?id=16516
17:33.38QwellChannelZ: if you have any more information to add, please do
17:33.46Qwell(unless that's your bug..)
17:34.05ChannelZQwell: That's me ;)
17:34.18Qwellyeah, kinda thought so
17:34.23Qwellhard to remember some of thenick mappings :p
17:34.31ChannelZJust trying to figure out if it really _is_ a bug
17:34.32ManxPower-work_cgc: you could be banned from the channel for flooding
17:34.33*** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman)
17:34.39ChannelZor just something screwy with me
17:34.44_cgcyes, sorry about that
17:35.37QwellChannelZ: would it be possible to record the bad audio from the phones perspective?
17:36.00_cgcit will not happen again :)
17:36.19Qwelloh.  8.04.  what gcc version?
17:36.23Qwell~gsmbug
17:36.23infobot[~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read :  http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39
17:36.33ChannelZQwell: Yeah I can do it through my audio card on a softphone for sure when I get home
17:37.07ChannelZQwell: ahh interesting.. (reading)
17:37.08ManxPower-workA few phones default to 30ms packets instead of 20ms (which is what Asterisk expects).
17:37.48ChannelZlooks like gcc 4.2.4
17:38.26Qwellwell, there's your problem.  upgraded recently?
17:38.37SuPrSluG_cgc: are you trying to call 2 technologies at once? SIP and Local
17:39.56TheDavidFactor-HChannelZ, I didn't have a problem with 1.6.1.6, but it looks like you found your problem?
17:40.26QwellChannelZ: post on that issue once you've recompiled with something besides 4.2, so we can close it or whatever
17:40.44ChannelZQwell: Yes and no, upgraded a long time ago but only recently found this as a problem when I upgraded to 1.6 (I don't use gsm much).  I'll try a recompile with optimizations off but it looks like this is probably what is happening for me
17:41.03ChannelZTheDavidFactor-H: Thanks for testing that!
17:41.07ChannelZQwell: I will
17:41.11_cgcSuPrSluG: ultimately its meant to dial some SIP phones, but liz requires you to use Local/${EXTEN}@phones so from the macro that gets called in the phones context i'm dialling the sip phone, is this wrong then?
17:42.22*** join/#asterisk KaneHau (n=KaneHau@133.40.166.155)
17:43.24_cgcthe other alternative is rewriting liz but i'm not that good with php
17:44.07voipmonklocal is not a bad solution
17:44.13voipmonkesp if it works
17:44.23*** join/#asterisk gme30066_ (n=gme@173.160.69.30)
17:44.23voipmonkget er done - move on to the next problem
17:45.11_cgcwell it should according to the documentation but it seems to say all of the queue members are 'Invalid' when using Local
17:46.22*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
17:46.22*** mode/#asterisk [+o Deeewayne] by ChanServ
17:47.02*** join/#asterisk xuser (n=xuser@unaffiliated/xuser)
17:50.25KaneHauAloha... asterisk newbie here with a problem.  When I do a 'ztcfg -vv' it identifies the channel properly but then says "1 channels to configure" rather than "1 channel configured" - any hints?
17:50.50[TK]D-FenderKaneHau: I don't see a problem there.
17:51.04[TK]D-FenderKaneHau: because ASTERISK configures it... not ztcfg
17:51.26KaneHauah, ok... hmmm.   does zapata.conf go in /etc or in /etc/asterisk?
17:51.38[TK]D-FenderKaneHau: Feel free to move on and show us where an actual problem presents itself :)
17:51.47[TK]D-FenderKaneHau: /etc/asterisk
17:52.02SuPrSluG_cgc: try turning up the verbose to see how it dials,  for it to be invalid it must try to do something
17:52.08[TK]D-FenderKaneHau: You really should migrate to DAHDI...
17:52.25KaneHauwell, I'm trying to test a very simple script - to answer the phone in Echo() mode.  I have a Viking advanced line simulator which provides the dialtone.  It rings, but the FXO is not picking up the line
17:53.55[TK]D-FenderKaneHau: Show us your actual attempt....
17:53.59[TK]D-Fender~pb
17:53.59infobot[~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://bin.cakephp.org/ , or apt-get install pastebinit
17:54.01[TK]D-Fender^^^^^^^^^^^^^^^^
17:54.11[TK]D-FenderKaneHau: and PASTEBIN it.... do not flood the channel
17:54.48KaneHauzaptel.conf is very simple:   fxsks=1    loadzone=us    defaultzone=us
17:55.20KaneHauzapata.conf is also simple:   context = incoming    signaling = fxs_ks    channel=>1
17:55.54KaneHauextensions.conf has, under [incoming]  exten => s,1,Answer()   exten=>s,n,Echo()
17:56.22[TK]D-FenderKaneHau: You said it provides dialtone.  What is there for * to ANSWER?
17:56.38_cgcSuPrSluG: http://pastebin.ca/1730907
17:57.14KaneHauno, you misunderstand... I have a line simulator that actually gives the dial tone to the FXO device.  Plugged into the other end of the simulator I have a real phone.  When I dial a number on the real phone, the simulator rings the FXO line which I'm expecting to pick up, but it doesn't
17:57.30*** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
17:57.44ManxPower-workKaneHau: Less talk, More pastebin.
17:58.17ManxPower-workCLI output, specifically
17:58.31[TK]D-FenderKaneHau: And * has to be started once Zaptel is actually ready.. and must have been compiled AFTER Zaptel is installed to even have support for it
17:59.01[TK]D-FenderKaneHau: So go prove a few things and pastebin "zap show status" "zap show channels" and "zap show channel 1" along with your failed attempt
18:00.28*** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net)
18:01.38raden_workNaikrovek, heya bro
18:03.11_cgchttp://pastebin.ca/1730907
18:04.20[TK]D-Fender_cgc: Should be using /n at the end, and I don't see dialplan to match
18:04.46ChannelZQwell: Note added to https://issues.asterisk.org/view.php?id=16516 - this is the gcc bug!
18:05.07ChannelZ(or gcc 4.2-ism anyway)
18:05.45ChannelZThanks for your help - Dunno why I didn't find that one, but I was doing a few searches and guess I missed looking for closed bugs on one of them..
18:06.06KaneHaubtw, I'm on SUSE and can't seem to find libnewt so that I can create zttool, any suggestions?
18:06.39KaneHauall the libnewts seem to be debian distributions
18:08.38_cgchttp://pastebin.ca/1730923
18:08.55tzafrir_laptopKaneHau, newt-devel ?
18:09.15KaneHauI'll look, thanks
18:09.21*** join/#asterisk thehar (i=thehar@thehar.xmission.com)
18:09.27tzafrir_laptopKaneHau, also note that the latest version of Zaptel is DAHDI
18:09.48KaneHauoh, I see.  So rather than doing the zaptel install, I should do the DAHDI install?
18:10.13KaneHauI'm following teh "Asterisk, the future of telephony" pdf (2nd edition) - is there a better reference I should be reading?
18:10.35*** join/#asterisk vally (i=vally@ip-95-222-217-180.unitymediagroup.de)
18:10.59ManxPower-workKaneHau: it's the best Asterisk Book out there, but it is becoming outdated.  Good thing there are lots of docs in the doc/ directory of the Asterisk souce code.
18:11.00torrancewin the same vein as KaneHau, is there a 3rd edition in the works?
18:11.19KaneHauthanks manx
18:11.35_cgc[TK]D-Fender: http://pastebin.ca/1730923
18:11.44torrancewi found the book quite helpful when i was starting, but the DAHDI/Zaptel bit was frustrating, as well as some 2.6 stuff in general
18:12.24[TK]D-FenderKaneHau: And I'm not seeing any of what I asked for...
18:12.35_cgcthe bottom bit that starts [levelone] is straight out of the queues.conf file
18:12.42[TK]D-Fender_cgc: add the /n and retry
18:12.53_cgcin queues.conf?
18:13.08KaneHauTK.. correct - I think I need to get rid of zaptel and go to DAHDI - so I want to reconfigure first
18:13.25ManxPower-work_cgc: read localchannel.tex in the Asterisk source.  /n for for Local/ channels.
18:13.39KaneHauno sense having you help if my basic setup is wrong
18:13.53_cgcok thanks, ill give it a try :)
18:14.31[TK]D-FenderKaneHau: What * are you on?
18:14.52KaneHau1.6.2.0-rc7
18:15.18ManxPower-workKaneHau: newbies should not be using unreleased code.
18:15.20[TK]D-FenderKaneHau: 1.6.x does not work with zaptel PERIOD.
18:15.26ManxPower-workI recommend you start with 1.6.1.x
18:15.26*** join/#asterisk bjifas (n=chatzill@r200-40-61-230.ae-static.anteldata.net.uy)
18:15.31[TK]D-FenderKaneHau: It requires DAHDI
18:15.31KaneHaumahalo nui loa...
18:15.32*** join/#asterisk angryuser_laptop (n=angryuse@93.180.241.181)
18:16.19*** join/#asterisk errotan (n=errotan@81.0.115.3)
18:16.22[TK]D-FenderKaneHau: And 1.6.2.0 release is out... you shouldn't be on an RC at all if you even really want to be on that new a branch
18:16.48KaneHauI'm downloading 1.6.1.12
18:17.06angryuser_laptopgood day, i have some strange cpu lock's with dahdi dummy, i have removed it from asterisk use but still the system is hanging sometimes, maybe it is CPU problem, can someone look at it ? http://www.pastebin.ca/1730930
18:17.19KaneHaumy application is extremely simple... the device just needs to place outgoing calls - so I don't need 99% of the bangs and whistles
18:19.22cuscoerr... http://paste.debian.net/55193/
18:19.24bjifasHola, tengo una duda de compatibilidad
18:19.28cuscocompiling asterisk addons errors out
18:19.32bjifasHello, I have a question of compatibility
18:19.41bjifaslibpri-1.2.8 and zaptel-1.2.27 are compatible with Asterisk 1.2.23?
18:19.45Qwellcusco: Are you using Asterisk 1.6.2.0?
18:19.46*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
18:19.53Qwell(hint: you aren't)
18:20.03cuscoI will be compiling it
18:20.12cuscodo I need to compile asterisk first?
18:20.18cuscook..
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18:25.43ariel_Hello everyone
18:29.44*** join/#asterisk rare1980_ (n=rare@119.152.35.15)
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18:35.43*** part/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
18:36.00hardwirebookawikawakachika
18:36.05hardwiredoes a little dance
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18:39.37*** join/#asterisk axelilly (n=jfenner@66.181.75.57)
18:40.15axelillyCan someone one point me in the direction of a document that can should me how to test what day and time it is in extensions.ael?
18:44.45[TK]D-Fenderaxelilly: "core show application GotoIfTime"
18:44.48*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
18:44.58[TK]D-Fenderaxelilly: "core show function IFTIME"
18:45.32axelilly[TK]D-Fender: thanks!  Looks like what I needed.  cheers
18:46.12Tim_Toadyand core show function STRFTIME or STRPTIME if u want to have a date string
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18:58.09axelilly[TK]D-Fender: when I tried the gotoiftime I got this message:  application call to GotoIfTime needs to be re-written using AEL if, while, goto, etc. keywords instead!
18:58.27axelillyCan I not use that application in AEL?
18:58.39[TK]D-Fenderaxelilly: I don't see how you TRIED to use it.
18:59.18axelilly[TK]D-Fender: ok, what I mean is that I did a ael reload and I got that message.
19:00.12[TK]D-Fenderaxelilly: What I mean is where's the damn code for me to look at? :)
19:00.15[TK]D-Fender~pb
19:00.15infobot[~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://bin.cakephp.org/ , or apt-get install pastebinit
19:01.35axelilly[TK]D-Fender: yea, just realized, that'd help   http://pastebin.com/d291f2123
19:02.19[TK]D-Fenderaxelilly: try usinf IF() and IFTIME()
19:02.23Tim_Toadyi think in ael its better to use iftime
19:02.32axelillyok, will do
19:02.47*** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk)
19:03.21benngardis happy! may213 did solve the "bug" by "borrow" my lab box
19:03.30Qwellbenngard: woot
19:03.58Qwellit's unfortunate, but sometimes borrowing hardware really is the only way to fix a problem
19:04.00benngardpstn - avaya -ooh323 - asterisk is fucking working
19:05.37benngardhe just "borrowed" some accounts ;)
19:06.00benngardand i did what he wanted me to do
19:07.13Qwellwell, either way..  glad it's working for you now
19:07.25benngardQwell: why ist it unfortunate, we are in  the same boat, if we can help each other why not?
19:07.44Qwellbenngard: I mean that it's unfortunate that some things can't be fixed without going that far
19:08.13benngardit was an avaya involved, what do u excpect ;)
19:08.17Qwellindeed
19:08.25*** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com)
19:09.16*** join/#asterisk asterisk1 (n=Jim@82-169-242-190.ip.telfort.nl)
19:09.18benngardbut may213 have to finish my bugreport! i have not a clue what he did with my "source-tree"!
19:09.21asterisk1hello all !!
19:09.59asterisk1i have a very difficult question. When i forward a call to my cell phone i cant see the orginal CALLER ID
19:10.09asterisk1using asterisk 1.4
19:10.21asterisk1i spend hours on this
19:10.33asterisk1but always see the caller id of my asterisk box
19:10.43asterisk1in stead of the orginal caller
19:10.50asterisk1Can someone help me please ?
19:10.57TheDavidFactor-Hhow is your * connected to the POTS?
19:11.20asterisk1i
19:11.27asterisk1'm using SIP
19:11.30asterisk1dont have pots
19:11.35asterisk1only a VOIP provider
19:11.45asterisk1and a Grandstream GXP2000 phone
19:12.50asterisk1when someone calls my and the call is being forwarded to my cell phone,
19:12.58asterisk1i can't see who is calling
19:13.04[TK]D-FenderastMaybe your provider doesn't allow you to set your caller id
19:13.08asterisk1and can't call back ..
19:13.12[TK]D-Fenderasterisk1: Maybe your provider doesn't allow you to set your caller id
19:13.26asterisk1that might be possible
19:13.44TheDavidFactor-Hyea, double check the outbound callerid right before you dial your sip provider then call them
19:13.51[TK]D-Fenderasterisk1: Then have your call out to your cell announce the callerID number before bridging.
19:14.16asterisk1i'm going to check my extensions.conf
19:14.26asterisk1and try your advise
19:14.26ManxPower-workasterisk1: find a provider that DOES let you set the callerid
19:14.33asterisk1ok
19:14.38asterisk1Thank you !
19:15.15benngardis still happy, have done a lot of test calls all worked
19:15.56benngardgonna celebrate with a BIG BEER
19:16.09benngardis gone for a while
19:17.57ManxPower-workasterisk1: Remember callerid number NEVER EVER has a leading 1, quotes, dashes or other non-number chars.  (the leading 1 is the toll prefix, not part of the callerid)
19:18.02asterisk1it must be my asterisk box... When i connect my Grandstream phone directy
19:18.16asterisk1to my provider, it works perfectly !
19:18.26[TK]D-Fenderasterisk1: huh?
19:18.39asterisk1i did something wrong in extensions.conf
19:18.45ManxPower-workA Noop(CALLERID(num) is ${CALLERID(num)}) as the priority before your dial
19:18.56ManxPower-workasterisk1: what version of Asterisk?
19:19.02asterisk11.4.24.1
19:19.19Kattywelp, i'm 100 bucks more broke
19:19.23ManxPower-workSee the "o" option to Dial in "core show application dial"
19:19.47asterisk1ok will have a look right now
19:20.03asterisk1thank you for your advise !!!
19:21.10asterisk1<PROTECTED>
19:21.11asterisk1<PROTECTED>
19:21.11asterisk1<PROTECTED>
19:21.44ManxPower-workUgh!  Apparently my new script is not actually smarter than a salesperson.
19:22.05ManxPower-workasterisk1: "core show applications"  "core show functions"  (function names are all UPPERCASE)
19:22.14Superbarttis it that dumb ManxPower-work?
19:23.34ManxPower-workSuperbartt: I didn't think so, but I guess I was wrong.
19:24.26asterisk1CALLERID(datatype[,<optional-CID>])  Gets or sets Caller*ID data on the channel.
19:24.29axelillyHow do you do multiple time checks, like 8-8 M-F and 9-5 Sat?
19:25.24[TK]D-Fenderaxelilly: Do multiple IF's
19:29.55ChannelZKatty: What'd you buy?
19:32.00ManxPower-workDoes anyone know of a way to force Asterisk to ignore inband indications and just make Dial hangup?
19:32.49ManxPower-work(this is on a PRI, Asterisk 1.4.recent
19:33.36KattyChannelZ: girly stuffs.
19:33.41KattyChannelZ: unmentionables.
19:34.12*** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com)
19:35.24KaneHauguys... ok, I installed DAHDI and reinstalled the non-beta asterisk - and THANK YOU VERY VERY MUCH - the FXO module now answers the ring and Echo() works for me
19:35.29KaneHaua BIG mahalo nui loa!
19:36.50Geminizerhello all.  question -- when Dial(...) is called from a dialplan, it blocks.  When Dial(...) is done blocking, the DIALSTATUS variable is immediately available for reading, correct?
19:37.51ChannelZshould be
19:38.09ManxPower-workGeminizer: Correct.
19:38.12KattyChannelZ: black polka dots, and cherries, if you must know.
19:38.32ManxPower-workIn fact DIALSTATUS is set even before the call terminates, you just can't usually access it since Dial blocks.
19:38.45KattyChannelZ: oddly enough, i can't even find it on their website :<
19:39.08Geminizerbut it is always guaranteed to store some value, regardless if the Dial(...) succeeded or not?
19:39.14ChannelZThat's OK I have a pretty good imagination
19:39.19Kattyk
19:40.45ChannelZGeminizer: I don't use words like 'guarantee' but it's at least supposed to always contain something interesting
19:41.07KattyChannelZ: they were on sale, buy 1 get 1 50% off.
19:41.12KattyChannelZ: regular price is 45 each
19:41.19Geminizergot it... thanks
19:41.41KattyChannelZ: but then i got side tracked at american eagle.
19:42.14*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
19:42.19*** join/#asterisk ocnarf (n=chatzill@114.108.194.153)
19:42.24ChannelZmeh.  Why don't gun shops do 'buy 1 get 1 50% off' sales?
19:42.27ChannelZpouts
19:42.53Geminizerthey may offer that deal at blow outs
19:43.35ocnarfhi everyone, need help. i want callers not be able to join the queue when all agents are enage. is it possible?
19:43.46ChannelZAnyone have OSX 10.6?
19:43.48benngard"hardcore" testing of ooh323 channel, my vife is using it atm ;)
19:44.08[TK]D-Fenderocnarf: Go read the sample queues.conf.  there are rather clear looking options in there
19:44.12benngardcalling mother iin law and vice versa
19:45.20ocnarfD-Fender: i tried joinempty=no but it doesnt work
19:46.13ChannelZthat's the opposite
19:46.19KattyChannelZ: well.
19:46.27KattyChannelZ: some have ammo discounts when you get a gun.
19:46.39KattyChannelZ: i know a couple places around here ryan has gotten a gift card for 20 bucks when he purchased one.
19:46.46Kattycourse that was a rather pricey rifle
19:46.53Kattyit's just sitting in the closet, collecting dust.
19:47.12ChannelZocnarf: look at maxlen maybe
19:47.14Kattythere's ammo in it, but it's not the word....
19:47.18Kattyuhmmm
19:47.23Kattynot loaded?
19:47.27Kattyit's loaded
19:47.37Kattybut..you'd have to pull that one thing back to get the ammo into the other thing
19:47.46*** join/#asterisk mrbnet (n=mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net)
19:47.48Kattymy description must be hilarious :P
19:47.55ChannelZso it's loaded, but without one in the chamber
19:48.17Kattyyeah.
19:48.35Deeewayne~roulette
19:48.36infobotACTION watches deeewayne pull the trigger:  Click!
19:48.38mrbnetCan anyone recommend a sip provider in the UK?
19:48.41[TK]D-Fenderocnarf: there is another option and value for there...
19:49.46ocnarfD-FEnder: tried both joinempty =no and leavewhempty =yes, still doesnt work.
19:50.03[TK]D-Fenderocnarf: And you still have managed to skip things...
19:50.24ocnarfD-Fender: any thoughts?
19:50.28gr0mitmrbnet. i do sip in UK...
19:50.52gr0mitwhat are you lookkingfor?
19:51.51[TK]D-Fenderocnarf: READ IT AGAIN
19:52.11raden_workNaikrovek, heya bro
19:52.53ChannelZwispers "hint: strict"
19:53.44mrbnetgr0mit: I have a customer opening an office there an my current provider does not offer numbers outside the US. I am looking for some UK numbers
19:54.07gr0mitno probs - pm me.  i can do most UK areas
19:59.17*** join/#asterisk batphone (n=will@rrcs-24-153-211-180.sw.biz.rr.com)
19:59.31batphoneim looking at these INVITES from a customer
19:59.50batphonethey show the To: field to be completely different than the INVITE URI
20:00.16batphonein fact, it says To: <sip:5551212@customer.ip.address>
20:00.22batphonerather than to my address
20:01.01batphoneis this right?
20:06.40*** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202)
20:07.13*** join/#asterisk ticoit (n=ticoit@201.205.153.166)
20:09.09ChannelZguess everyone is at lunch
20:11.52jblackNah. I'm thinking about lunch.
20:12.09jblackI want a big burger.
20:12.16ChannelZme too, I'm hungry
20:12.21ChannelZBurrito, methinks
20:12.42jblackEver been to panchero's ?
20:12.49ChannelZnever heard of it
20:13.17jblackIt's teh r0x0rs
20:13.50ChannelZhmm Grand Junction or Montrose.  Not my neck of the woods
20:14.37Kobazgrand junction is fun
20:14.39Kobazgood mountain biking
20:14.41ChannelZlooks like another Chipotle but with more interesting plates
20:14.46jblackThey use fresh stuff, and even make their tortillas on the fly
20:14.53cuscomrbnet: voipuser.org
20:22.17*** part/#asterisk torrancew (n=torrance@ip70-186-186-21.br.br.cox.net)
20:23.12shamelessn00bChannelZ: its working now
20:23.14shamelessn00b:D
20:23.22shamelessn00bdahdi calls
20:23.25ChannelZit?
20:23.29shamelessn00bI tweaked 2 parameters
20:23.33shamelessn00b240 calls
20:23.36shamelessn00b240 agis
20:24.01shamelessn00bincreasedthe chunk size to 80
20:24.20shamelessn00band changed the max connection limit on my DB server xD
20:24.34shamelessn00bwanpipe chunk size
20:25.46ChannelZI don't know what that is but glad it worked
20:29.16[TK]D-FenderChannelZ: Changes the interrupt load drastically
20:29.50ChannelZwanpipe == sangoma's hardware?
20:31.38ChannelZso it was a throwup of the PCI bus like I thought last night
20:32.32[TK]D-FenderChannelZ: nothing says " I love you" like projectile vomit....
20:33.18ChannelZor being gagged while trying to, really..
20:33.20ChannelZ:)
20:35.03*** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com)
20:37.14bmoracaautoeroticasphyxiation...i think there's a support group for that
20:45.09bmoracacrazy german guy cracked GSM's encryption
20:45.21voipmonkcrazy?
20:45.39TSMstill need a lil bit of power todo it in realtime
20:46.06TSMbut give 6months and with the next gen crop of cuda cards etc it will be much easier
20:46.06KaneHauOk... I'm reading the asterisk pdf book... the examples work fine for me.  However, my applciation needs to PLACE calls, never answer them.  I don't see in the book how to actually initiate a brand new phone call to an outside number programtically (I'm writing in 'C')
20:46.14ChannelZdamnit, NOW how will I order my explosive underwear?
20:46.39KaneHauchannel:  don't bother - TSA's next decision will be to make us all "FLY NAKED"
20:46.46KaneHaushould cut down on obesity :)
20:47.40bmoracaif you're having conversations that you don't want anyone else to overhear, you probably shouldn't have them over a phone anyway.  i don't get what the fuss is
20:48.23bmoracaKaneHau, you can do that three ways:  AMI, call files, and the "Originate" CLI command
20:48.42KaneHaubmoraca: thank you for the pointers
20:48.43ChannelZI think he's actually writing an * application
20:48.59ChannelZor no?
20:48.59KaneHauwell, I have a C application that needs to make phone calls
20:48.59shamelessn00bbmoraca: A5 was cracked ages back
20:49.08KaneHau(needs to call scientists to report on alarm conditions)
20:49.23ChannelZoh.  I thought you meant you were writing an interneral * app.
20:49.30bmoracaKaneHau, your C application...is that WITHIN asterisk or does it just need to INTERFACE with asterisk?
20:49.42shamelessn00bthere are embedded systems that can sniff GSM calls in realtime
20:50.03shamelessn00bk gais, me off
20:50.05shamelessn00btc gnite
20:50.05KaneHaujust needs to INTERFACE to asterisk to place the call, and monitor for DTMF tones
20:50.34KaneHauit places a call, speaks a custom message (created on the fly) and then awaits for DTMF tones to make a few decisions
20:50.48ChannelZRobodialer spam?!
20:50.49bmoracaKaneHau, you won't be able to monitor for DTMF tones unless the audio path goes through your application.
20:51.10KaneHauthe number it calls is unknown until the moment the call is placed
20:51.10KaneHauok
20:51.10KaneHauso that means I'm writing a * application, right?
20:51.32bmoracaKaneHau, why bother with that when said applications already exist?
20:51.58[TK]D-Fender[15:46]<KaneHau>channel: don't bother - TSA's next decision will be to make us all "FLY NAKED" <- Not so...
20:52.11[TK]D-Fenderspins up Pat Benatar's "Sex As A Weapon"
20:52.24KaneHaubmoraca: this has to interface into one of our huge telemetry systems - much easier if I custom  design it
20:52.24*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
20:52.33KaneHauwe alraedy have it for TAPI - just replacing TAPI with *
20:52.41bmoracaKaneHau, use AGI for that.
20:52.50*** join/#asterisk wam (i=wam@unaffiliated/wam)
20:53.36bmoracaKaneHau, unless you want to create a custom SIP UA...but I would see stabbing myself in the eye with a marlinspike as more entertaining and a better use of time :)
20:54.50KaneHauyes, I looked at AGI, but the examples are all from a standpoint of answering a call... how does the AGI actually initiate a call?
20:55.03bmoracaKaneHau, the two are not related at all
20:56.14bmoracaKaneHau, you HAVE to have an external application initiate the calls...whether by sending SIP messages or by using Asterisk's built-in methods (spool, AMI, originate).  once the call is setup, everything else happens within Asterisk...which is where AGI comes in if you need to integrate with another application (or func_odbc if just a database)
20:56.46KaneHauthank you for the pointers
20:56.50KaneHaumuch reading ahead
20:56.55ChannelZit is definately the time of lunch.
20:56.57bmoracaKaneHau, the audio path, however, does NOT go through the application that initiates the call unless that application was created as a SIP UA
20:57.13bcrispum
20:57.22KaneHauactually, the manual said audio was available on file descriptor 3
20:57.24bmoracaKaneHau, in that case, Asterisk is nothing more than a SIP gateway.
20:57.26KaneHauis that not the case?
20:58.09KaneHauwell, atcually I don't need the audio, I just need to know what DTMF tones were hit by the user
20:58.51bmoracaKaneHau, it may be, but that's only going to be available within Asterisk (if you plan to write this as an Asterisk module), but you still need an external program to initiate the calls
20:59.10KaneHauok, thanks
20:59.12bmoracaKaneHau, DTMF is sent with the audio...or out of band, but still follows the audio path
20:59.27bmoracagenerally
21:01.24bmoracaKaneHau, there are two ways, really, to do what you want...you can create a SIP UA that does all processing and call setup itself (including playing messages and capturing audio (DTMF)), or you can create an application that tells asterisk to setup a call and supplies certain variables over AMI and use Asterisk itself to actually play the audio and capture DTMF (then you can do anything you want with it via AGI or func_odbc or anything e
21:01.24bmoracalse)
21:01.31*** join/#asterisk alfa202 (n=svelluto@dhcp-0-9-e8-4a-96-80.cpe.quickclic.net)
21:01.43bmoracaoption 2 would most likely be a lot easier to put together
21:01.50KaneHauI think the 2nd solution you gave is probably what I want
21:02.16KaneHaumy program needs to create the script on the fly, then initiae the call and have astrisk then deal with everything (for the most part)
21:03.11bmoracaKaneHau, that shouldn't be too difficult.  AMI is how you'll initiate the call, and AGI is likely where you'll do your processing (though you don't have to use AGI)
21:03.25KaneHaubtw, unrelated... the VIKING Advanced Line Simulator I'm using rocks!  Very nice for testing out this hardware
21:03.26*** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be)
21:03.40bcrispi like viking stoves
21:03.43KaneHaugot it, thanks
21:04.08KaneHauthis is the Model DLE-300 - has two phone ports on it and handles creating a dial tone on both ports.  Can simulate 911, etc
21:04.47bmoracai use an Adtran TA900 with another asterisk server (virtualized) to simulate the PSTN, generally
21:05.17*** join/#asterisk Tim_Toady (n=moi@77.49.136.12.dsl.dyn.forthnet.gr)
21:05.24KaneHauprobably would have done that if I was already * savy - this got me up and running without having to be * smart
21:06.02[TK]D-FenderKaneHau: I find a < $50 Linksys PAP2T-NA to be a more than adequate line simulator....
21:06.05*** join/#asterisk akira2014 (n=chatzill@220.Red-88-6-197.staticIP.rima-tde.net)
21:06.16KaneHaureads the AGI chapter
21:06.38KaneHauI had money to burn
21:06.57*** join/#asterisk fofware (n=chatzill@190.7.25.160)
21:07.44*** join/#asterisk chazzm (n=chazz@173-24-238-25.client.mchsi.com)
21:09.13akira2014can some one tell how to migrate a sip conf from asterisk 1.4 to asterisk 1.6
21:09.15akira2014http://pastebin.com/d16b4effc
21:09.26akira2014this is part of my sip.conf
21:09.31akira2014thk's in advance
21:09.37Defrazbmoraca: Do you have any configures for an Adtran 924. Trying to use it as a SIP gateway
21:09.57DefrazI have a PRI and an adtran 924 and trying to use that to sip it to my * server.
21:10.13DefrazCan do it on a cisco but can't figure out this Adtran.
21:12.09*** join/#asterisk ticoit (n=ticoit@201.205.153.166)
21:14.15*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
21:14.59KaneHaubmorace: AMI looks to be what I'm looking for, thank you greatly
21:20.42*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
21:23.24*** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110)
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21:28.01ManxPower-workDefraz: standby
21:29.13[TK]D-Fender\o/
21:29.29[TK]D-FenderFinally got asterisk.org 's downloads page looking right....
21:31.05ManxPower-workDefraz: We use massive amounts of Adtran gear.  I'm asked about the configs, but my boss is AFK
21:32.04Qwell[TK]D-Fender: what was wrong with it?
21:32.38AkiraaWhat's the latest edition of "Asterisk: The Future of Telephony" ? I have a pdf of Edition 2 (2007)
21:32.39[TK]D-FenderQwell: Lack of direct downloads link.  Section for sounds with no link to them either (which is highly desirable for non-internet site installs), etc
21:33.05[TK]D-FenderQwell: Qwell And the earlier correctio for complete lack of link to Addons.
21:33.38[TK]D-FenderAkiraa: thats it
21:34.02Akiraaso no significant changes since 2007
21:35.02[TK]D-FenderAkiraa: to the BOOK?  Its paper :)
21:35.24[TK]D-FenderAkiraa: It isn't an "official" doc anyway.  Thats what the tarballs are for
21:35.39Akiraa[TK]D-Fender: no, to Asterisk itself, 2 years is a long time for software
21:36.09luckyabaWhat is the best router in your opinion for VOIP?
21:36.22[TK]D-FenderAkiraa: the book was circa 1.4.  We are at 1.6.2 branch now.  Feel free to activate all those dormant neurons...
21:36.30[TK]D-Fenderluckyaba: iptables
21:37.06luckyaba[TK]D-Fender, haha, fair enough. How about something that we can put in front of a clients network that is a bit easier to setup and manage?
21:37.53[TK]D-Fenderluckyaba: * doesn't really care, but PIX & D-Link's = trouble
21:38.09[TK]D-Fenderluckyaba: Your typical Linksys home router tends to work just fine
21:38.16voipmonki can smack a dlink into submission
21:38.19ManxPower-workluckyaba: Cisco 2621XM
21:38.20voipmonkbut why?
21:38.21voipmonk:)
21:38.40ChannelZmmmm burritoooooo
21:39.00luckyabayou guys have good feedback on maybe using a Linksys or Buffalo with DD-WRT?
21:39.11ManxPower-workluckyaba: You said good router.
21:39.22luckyabalol
21:39.34luckyabatouche
21:39.41ManxPower-work"best consumer grade" is what it looks like you are looking for.
21:39.59ChainsawYou're going to end up with a Billion or something similar.
21:40.06ManxPower-workI personally use an old Cisco 175x as my router.
21:40.08ChainsawZyxel used to be decent, but lost their way around the Zywall 2 plus. Avoid.
21:40.21ChainsawManxPower-work: Overpriced for what you get, but yes, reliable.
21:40.25luckyabaThis is for a solution provided to clients ManxPower-work
21:40.32luckyabaso it has to be a product readily available
21:40.35ManxPower-workChainsaw: like $50 on ebay
21:40.52luckyabaused?
21:40.55luckyabawarranty?
21:40.59luckyabasupported?
21:41.05ManxPower-workYes.  No warrenty.  Buy two of them if you want that.
21:41.15ManxPower-workluckyaba: like you'd ever get an support from Linksys support.
21:41.17luckyabaclient(s)
21:41.25TSMi like sonicwall units, the total secure units are fairly good TZ210 i think arement to be good, ive got PRO2040 HA units
21:41.25luckyabawe will need to be buying a lot more than 2
21:41.37luckyabaSonicwall is what we currently support
21:41.43luckyabaand they are put simply.... Garbage
21:41.48ChainsawTSM: I don't know, it sounds like an air conditioning unit.
21:42.09TSMthey are good, ive had a load of them over the years
21:42.13TSMno problems
21:42.40TSMthe Total Secure units have full IPS/GAV etc protection plus warranty etc..
21:43.21TSMmost of them now come with HA function built in and you can get the second unit a a fraction of the cost of a normal one
21:43.28*** join/#asterisk clyrrad (n=IceChat7@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com)
21:43.31ChainsawTSM: High Airflow?
21:43.46luckyabaTSM, They handle VOIP horribly
21:43.55luckyabaSonicwall themselves will admit it
21:44.00TSMive had no issues, but then im running my VOIP in DMZ
21:44.37luckyabaSonicwall is making an effort to resolve the problems but its going to be a long time before that happens
21:44.39TSMtransparent DMZ is easy to use
21:44.44luckyabaand we need a reliable solution for our clients
21:45.03TSMits reliable, i dont know why people have had issues, mabey yr talking about old software
21:45.05luckyabaThat is a lot of money in extra setup time on the clients dime
21:45.18ariel_firewall I like taking old pc with 2 nic's and putting Endian on it.
21:45.19luckyababecause then your talking about configuring iptables
21:45.28luckyabaso at that point we might as well throw the Sonicwall away
21:46.09luckyabaareay, personally I am with you on that but from a company's perspective I won't get them on board
21:57.06*** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147)
22:10.09*** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net)
22:13.55Geminizerwhy does Playback(exit_status_6-eng) work but not Playback(exit_status_6-${lang}) ?
22:14.09Geminizerwhere $lang = "eng"
22:14.29QwellHow did you set lang?
22:15.23Geminizerit's in a .call file as:   Set: lang=eng
22:15.36ManxPower-worktry setting lang to en not eng
22:15.49ManxPower-workAh, you said it works.
22:16.40Geminizerright... it works when I use the full name.  When I introduce a variable as part of the name, it fails..
22:16.52QwellAre you setting the var on the right channel?
22:18.05ManxPower-workAsterisk's language is "en", which is what I was thinking.  Perhaps I'm caffeine deprived. 8-)  (no
22:18.14Geminizeryes, along with all the other channel variables which work
22:18.26Qwell(tip: you need a channel in order to use a variable.  calling Application: Playback from the callfile won't work)
22:19.18ManxPower-workGeminize: You set it in the call file as Set: lang=eng and not Set: lang="eng"?
22:19.43Geminizercorrect.. the first way you mentioned is how I have it
22:19.51ManxPower-workRemember, most times quotes are literal in Asterisk
22:20.32ManxPower-workGeminizer: Put a Noop in the dialplan to show you the values of the variables you set.
22:20.45ManxPower-workIf you're running Playback, I'm assuming you're in the dialplan.
22:20.51Geminizerahh, I think I know what the problem is... one moment
22:24.13*** join/#asterisk lanning (n=lanning@208.87.235.224)
22:24.23*** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com)
22:25.45Geminizergot it... I hadn't realized the filenames were named differently...
22:30.49*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
22:32.55*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
22:36.32ChannelZSpeaking of filenames
22:37.02ChannelZWhy does * say that it's playing "blahblah.slin" when really it's playing a file called "blahblah.wav"?
22:38.56bmoracaDefraz, if you're still here and interested, the TA900s were not really meant to be used in that direction.  if the problem is that the PRI is not coming up (remember, it has to be in port 3 or 4), it could be that you need to use a T1 crossover cable...depending on how your jack was wired.
22:45.24*** join/#asterisk sbrath (n=sbrath@unaffiliated/sbrath)
22:46.40*** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147)
22:54.09*** join/#asterisk duckz (n=duckz@86.107.84.186)
22:58.23citywokChannelZ: my guess is it decided to use the wav because it required no/less transcoding
22:58.28citywokand it made that decision after announcing it?
22:59.02ChannelZwhat I mean is that physically on disk the file is called whatever.wav but when voicemail plays it it calls it whatever.slin
22:59.27citywokodd
22:59.43*** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com)
22:59.45citywoksounds like a trivial bug, lol
22:59.58*** join/#asterisk ManxPower (n=ewieling@216.186.151.147)
23:00.31[TK]D-FenderChannelZ: perhaps you could pastebin the complete call from the call to Playback through the end, along with the dump of your sounds folder...
23:02.55fileit's actually <name>.<format being fed to the channel>
23:05.39*** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
23:09.57ChannelZwell it's not that it doesn't work, it's just that it looks funny
23:10.04ChannelZ-- <DAHDI/1-1> Playing '/var/spool/asterisk/voicemail/default/200/unavail.slin' (language 'en')
23:25.08*** join/#asterisk joako (n=joako@opensuse/member/joak0)
23:43.39*** join/#asterisk johnyjj2 (n=mainacco@p9g117.traco.pl)
23:44.14johnyjj2hello :)
23:45.19johnyjj2can somebody help me with configuring Asterisk Win32?
23:46.12[TK]D-Fenderjohnyjj2: asteriskwin is not supported here.  for general configuration, the common docs should still apply
23:49.54johnyjj2[TK]D-Fender: thank you, anyway if somebody would be willing to have a look, I made some print-screens from my configuration here http://www.speedyshare.com/data/401343461/20015137/72160770/foto.rar i'm available either here or by mail johnyjj2@gmail.com I just wanted to check Asterisk configuration with X-lite, thanks
23:52.52[TK]D-Fenderjohnyjj2: Want to check it... USE IT
23:52.58[TK]D-Fenderjohnyjj2: And watch * CLI
23:55.20johnyjj2That's the difficulty. I would prefer using Asterisk on Linux but I need to use in on Windows, this is why I downloaded Asterisk Win32. It created icon on desktop for WillVoice PBX Manager but this CLI all the time says "Unable to connect to remote asterisk".
23:57.11[TK]D-Fenderjohnyjj2: "need to use it on windows"?  What insanity is the basis of this?
23:57.59[TK]D-Fenderjohnyjj2: And now you are bringing 3rd party management tools into this... lovely
23:58.07johnyjj2this is not my server and the owner/admin of the server asked me to install asterisk on it and configure it for ivr/asr
23:58.45TSMtell him to foff
23:58.55TSMi mean the owner/admin
23:58.56[TK]D-Fenderjohnyjj2: Well this 3rd party tool seems to want to connect via AMI.  Go set up your manager.conf
23:59.51johnyjj2this asterisk 32win is porting asterisk to windows with cygwin, it installed this willvoice pbx manager (3rd party tool) so i guess that's the only available cli for asterisk win32 but i may be wrong

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