IRC log for #asterisk on 20120820

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02:33.21dijibanybody have any experience with an spa2102-r?
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05:25.56diabMatin.
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06:48.27schmidtsgood morning
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06:54.17bulkorokhi
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07:44.21ChannelZoh noes
07:51.13flingbad things happen
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08:24.59zambahow do i build the dahdi_dummy module on ubuntu?
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08:29.30kaldemarzamba: if you have a current version of DAHDI, the dummy functionality is in the core module. you won't get a separate _dummy module.
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09:23.33markithi, I've a 4 ports sip ata that worked fine with older version of asterisk, some time ago I've upgraded to 1.8 and seems that looses it's registration. I've looked at the packets with wireshark, and I've noticed that is trying to register with a nounce, but asterisk replies with the previous one, then the ata generates a new one, and asterisk... they never meet! Is it a known issue or a config problem?
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09:33.25john_____hi all
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09:45.05jacobwwhich configuration files in /etc/asterisk are parsed without being #included from another file?
09:56.19kaldemarjacobw: grep -r -o -h "\w*\.conf" * | sort | uniq
10:07.39jacobwkaldemar: i understand that, my question is where does the parser start?
10:08.24jacobwkaldemar: for example, sip.conf #includes sip_foo.conf, does sip_foo.conf have to be included from sip.conf or can it be included from sip_arbitrary.conf and still be found by the parser?
10:10.00jacobwkaldemar: my question really is, 'are all files in /etc/asterisk (ending with .conf?) read by the parser, or only certain files the files that they include'
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10:11.31kaldemarjacobw: all files are not read.
10:12.08kaldemarjacobw: modules, resources etc. usually read a single file.
10:12.12gopalin opensuse asterisk is hanging while loading modules
10:12.21gopalhave anybody faced this
10:15.51jacobwkaldemar: ok, thanks
10:16.27jacobwkaldemar: do you know how i can determine which file are parsed?
10:17.45ectospasmgopal: try loading asterisk in the foreground ("asterisk -cvvv") and see if you can identify what is causing it to hang.
10:21.55kaldemarjacobw: depends on which modules you have installed/loading.
10:24.23gopalectospasm: i started but its hanging in modules.conf
10:25.35gopalectospasm: when i start asterisk -vvvvc it hangs when it try to load modules.conf "loading modules"
10:28.31ectospasmgopal: what is the last module it loads?  Try putting "noload <module>" in modules.conf
10:30.08gopalectospasm: if i disable autload=no then asterisk is starting... and i am getting CLI prompt...post to that if i try to load chan_sip or whatever modules it hangs
10:30.43ectospasmwhat version of Asterisk? << gopal
10:30.53gopalAsterisk 1.8.15
10:30.58gopalonly in Opensuse I face this error
10:31.05gopalother Ubuntu and CentOS it is fine
10:31.12ectospasmYeah, I avoid any SuSE like the plague
10:31.27ectospasmdid you build Asterisk from source?
10:31.29gopalectospasm: its really weired working in OpenSuse....
10:31.34gopalectospasm: yes
10:31.58ectospasmI haven't a clue then.  Something must be up with the build environment
10:32.02ectospasmwhat arch?
10:32.25gopalectospasm: i tried in both 32bit and 64 bit arch
10:32.31gopalectospasm: failed :(
10:32.57ectospasmdoes SuSE separate the modules in /usr/lib (32bit) or /usr/lib64?
10:33.19gopalits uses default /usr/lib
10:33.27gopalwhether it may be 32bit or 64bit
10:33.48ectospasmwhat if you download and install Asterisk from OpenSuSE's package manager (YaST I think?)
10:34.00gopalhmm that i haven't tried
10:36.04ectospasmgranted, it will probably be outdated, since upstream Asterisk 1.8.15.0 was only released a couple of weeks ago
10:36.48gopalwith yast i tried, not able to find the package
10:40.07gopalectospasm: i will try to install with older version Asterisk 1.6
10:41.32WIMPyBetter try a newer one.
10:42.21ectospasmyeah, as of April 21, 2012, all versions of 1.6 (and prior) are EOL
10:42.57WIMPyAnd 1.6 wasn't that great anyway IMHO.
10:42.57gopalectospasm: so you mean to say that using 1.8 is better?
10:43.10gopalok
10:43.18WIMPyYou could try 10.
10:45.41ectospasm...but 10 won't be around for much longer (relative to 1.8)
10:46.13WIMPyBut it's better.
10:46.42ectospasmnot if you want full support for the longest period
10:46.55ectospasm...regular vs. LTS
10:47.16WIMPyThen use 11 beta.
10:47.38ectospasm...but 11 beta doesn't come with official support *yet*
10:48.51WIMPyIn practice it has the best support. I'm a user not a lawyer.
10:49.28WIMPyOtherwise I would never use Linux.
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11:16.47indika1978hi, How could I get the retrieve extension of parked call when it used in a custom dial plan? Your help expected.
11:54.58ectospasmindika1978: when a user parks a call, the system should report what parking slot the caller is in...
11:55.09ectospasm...as an audible notification
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12:13.14_val_Hey everyone. I know there is a way to block certain incoming numbers but where do I do this and how?
12:13.40[TK]D-Fender_val_, "Core show application gotoif" <-
12:16.17[TK]D-Fender_val_, "core show function CALLERID"
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12:17.42_val_[TK]D-Fender: there is a stalker calling all the time.
12:17.54indika1978ectospasm: thanks .. Is there any way to get that extension no through asterisk application
12:17.58_val_Need just to block 06845741245
12:18.01_val_such number
12:18.02[TK]D-Fender_val_, So go ahead and dump their calls
12:18.28[TK]D-Fender"Core show application gotoif" , "core show function CALLERID" <------
12:20.51_val_[TK]D-Fender: I get some parameters.. but still don't get the idea. @_@
12:22.35WIMPyJust do exten => yourexten/numbertoblock,1,Hangup(21)
12:23.38ectospasmindika1978: see "core show application ParkedCall"
12:23.42_val_WIMPy: what is yourexten in this case?
12:24.03WIMPyYour extension
12:25.17_val_WIMPy: I need it for all extensions
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12:25.51WIMPyUse a pattern like _X..
12:28.26_val_WIMPy: so it would be then  exten => _X..,0616854658,1,Hangup(21)
12:28.31_val_putting this in extension.conf?
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12:29.47WIMPyThe second . was from the sentence.
12:30.11WIMPyAnd the callerid is seperated with a /.
12:30.11dymHey everyone! Is it possible to manipulate a variable which has no obvious seperator ina way to get the last 3 chars?
12:30.16dym<3 WIMPy
12:30.35WIMPy:-3
12:30.48dymdoesnt that remove the last 3?
12:31.06dym"If a negative length value is entered, Asterisk will remove that many characters from the end of the string. "
12:31.15dymits not like it didnt look :)
12:31.27WIMPyOk. ::-3
12:31.40dymtries
12:31.45_val_WIMPy: this then?  exten => _X./0616854658,1,Hangup(21)
12:31.56WIMPyyes
12:32.10_val_WIMPy: this goes into extensions.conf right?
12:32.30WIMPydym: My doc says start n chars from the end. So only one : should do it.
12:32.39WIMPy_val_: Yes
12:33.07dymWIMPy: http://www.asterisk.org/astdocs/node77.html
12:33.38WIMPydym: Yes. What I said.
12:34.12dymerr - no its not. Set(DW=${EXTEN:-3}) would then remove the last 3 digits from the number
12:34.28_val_WIMPy: now need to reload the config.
12:34.32WIMPyThat's not what the page says, you just linked.
12:34.53_val_How would I reload the extensions.conf?
12:35.02WIMPy'dialplan reload'
12:35.12dymWIMPy: sorry, read the bottom line /:
12:35.23_val_Thanks.
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12:35.47WIMPydym: What are you talking about?
12:37.21dymWIMPy: I said: sorry, i read the bottom line. This means i apologize for beeing wrong about my statement. whats not to get?
12:37.44_val_WIMPy: Added extension '_X.' priority 1 (CID match '06168544645') to default (0x7f18ec00a9b0)
12:37.59WIMPyThat needed clarification :-)
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12:38.23WIMPy_val_: You're using teh default context?
12:38.26_val_WIMPy: does this mean anything went well.
12:38.54WIMPyIf your calls come in the defautl context, yes.
12:39.30_val_WIMPy: I don't onderstand what the default context is in this "contexts" :>
12:40.06WIMPy_val_: Who configured them?
12:40.24_val_WIMPy: one of mine colleagues. He is on vacation.
12:41.35[TK]D-Fender_val_, It is whatever you made it to be
12:41.56[TK]D-Fender_val_, [default] is a context name like any other.  That is the one you chose to put that in
12:41.59WIMPyThen look at the config of the channel the call is coming from or at the logs.
12:42.11[TK]D-Fender_val_, so hopefully that is where your incoming call is landing.
12:44.55_val_[TK]D-Fender: it is a  [incoming] clausse
12:45.19[TK]D-Fender<_val_> WIMPy: Added extension '_X.' priority 1 (CID match '06168544645') to default (0x7f18ec00a9b0) <-- then you should ahve made it in [incoming] instead
12:46.29_val_[TK]D-Fender: [TK]D-Fender no there is an [incoming]  and [default]
12:46.40_val_I put the line WIMPy gave me under the [default] clausse.
12:46.47[TK]D-Fender_val_, yes well you need to put that in the context your CALL lands in
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12:47.28_val_[TK]D-Fender: that context is?
12:47.42[TK]D-Fender_val_, Wherever you told them to land
12:47.58carrarAt the Airport!
12:48.01[TK]D-Fender_val_, this is YOUR config.  Go look at where you pointed them to.
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12:50.01_val_[TK]D-Fender: yes. I think it is the [incoming]
12:50.21[TK]D-Fender_val_, then that is where you should be doing your check
12:51.51_val_[TK]D-Fender: http://pastie.org/4556147
12:51.58_val_this is how [incoming] look like
12:53.27[TK]D-Fender_val_, so go make exten => s/0616854658,1,Hangup(21)
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12:55.50_val_[TK]D-Fender: http://pastie.org/4556175  is how  [default] look like
12:56.00_val_look/looks
12:56.24[TK]D-Fender_val_, You said the calls land on [incoming].  [default] does NOT matter then
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12:57.04_val_[TK]D-Fender: I am not sure.. as I just got into this. I thought based on the syntax you might tell which context I have to put the blocking number.
12:58.12[TK]D-Fender_val_, where it lands is up to YOU.  YOU tell it where to go.  There is no "automatic" place all calls will always go to.  This is configurable and no reason to assume one person's config looks anything like anyone elses.
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12:59.12[TK]D-Fender_val_, Your [incoming] looks like I would expect for calls coming from a SIP provider that doesn't pass the DID in the standard way (hence the TO: header parsing we see there)
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12:59.39[TK]D-Fender_val_, So by appearances, [incoming] looks to be where you need to put exactly the line I gave you
13:02.06_val_allright [TK]D-Fender thanks. So does it matter if it stays under [incoming] and [default] ?
13:02.12_val_it won't matter will it?
13:02.18[TK]D-FenderDon't add extra junk where it doesn't belong
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13:13.58_val_[TK]D-Fender: & WIMPy Thank you. I'll have to wait and see if it has had any effect.
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13:28.14dymHey - im trying to escape the ; within a channelID by doing this, which fails: Set(__CIDESC=${CUT(CID,;,1)}\;${CUT(EXTEN,;,2)} any hints?
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13:30.53[TK]D-Fendershow us the failure along with NoOp's dumping the contents prior to the call
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13:33.00dym[TK]D-Fender: http://pastebin.com/uaku9wZ4
13:33.13dymNo NoOp Output, since theres nothing to output
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13:35.40[TK]D-Fender(missing '}')
13:35.44[TK]D-Fenderthat says it all
13:35.56[TK]D-FenderAnd you can output the VARS PRE-CUT
13:36.39[TK]D-Fenderdym, and you aren't showing us complete dialplan or complete output either
13:37.07dym[TK]D-Fender: oh, you want the EXTEN as it is before?
13:37.13[TK]D-Fenderall of it.
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13:47.01indika1978hi , how can we hold the call and dial to extension for choice 1.take call or 2. terminate the call?
13:47.53indika1978<PROTECTED>
13:49.36[TK]D-Fenderindika1978, "core show application waitexten", "core show function TIMEOUT"
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13:54.26eduzimrsHi, i have a situation i cant understand -> http://pastebin.com/ksGef1Z3
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13:55.11eduzimrswhy in my cdr log is recording the Mac Addr in CDR(dst) instead of 411 ?
13:55.48[TK]D-Fendereduzimrs, You should probably be shoing us the call...
13:55.59WIMPyBecause you Goto. That changes the extension.
13:56.27eduzimrsOMG there is such way to outline this issue and mantain 411 ?
13:57.56eduzimrsso why i NoOp it shows 411?
13:58.01eduzimrsso why in NoOp it shows 411?
14:00.08eduzimrsthis CDR field is just RO
14:02.15indika1978[TK]D-Fender: Thanks for the information , I am trying to write a dialplan and it need to hold a call for certain time and at the hold begins i need to dial internal extension.
14:03.17indika1978with the hold I cant dial another extension and always it dial after hold over. how can i achieve that? Pls help me in this. Thanks
14:03.24[TK]D-Fenderindika1978, by "hold" you mean "just sit around and make them wait"?
14:03.49[TK]D-Fenderindika1978, Your description isvery hard to read
14:04.03[TK]D-FenderinIf you put a call on hold, jsut about any phone will let you dial another #
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14:08.50eduzimrsWIMPy; there is such way?
14:09.13indika1978Thats not happening in herer .... http://pastebin.com/hgFMfxvk
14:09.29ipengineerdoes anyone know how to make a queue appear empty and advance to the timeout destination if all agents are busy?
14:10.09[TK]D-Fenderipengineer, lok in the sample config.  the parms for that are all there
14:10.14[TK]D-Fenderlook*
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14:11.08GraidenHas anyone ever gotten a 'reply to e-mail to delete voicemail' function working?
14:11.16[TK]D-Fenderindika1978, Because you told it to wait forever.
14:11.38[TK]D-Fenderexten => s,2,MusicOnHold(10) <-- this says look for Class "10"
14:12.02ipengineermember:identifier:%5Btk%5Dd-fender I have been through there several times. I have set joinempty = no and ringinuse = no
14:12.21indika1978I am tying to dial once the hold the call and dial another extension once the hold start. I am in a real mesh
14:12.26[TK]D-FenderWhen you have only 1 parameter filled in, that is the CLASS.  in order to specify a duration you have to specify the class, and the 2nd paramter is a timeout
14:12.38[TK]D-Fenderindika1978, Yuo did not properly read the instructions for that app
14:12.57[TK]D-Fenderipengineer, there is more than just "no"
14:13.03[TK]D-Fenderipengineer, Read closer
14:13.11indika1978I might .. sorry to bother you .. ok I will go thourh it
14:13.35[TK]D-FenderGraiden, * doesn't READ e-mail... so I'd like to know what function you're referring to...
14:14.54Kattyi feel embarassed to be from missouri this morning.
14:15.04ipengineer[TK]D-Fender Im sorry, joinempty = strict I had to go confirm again.. we are running realtime. ringinuse = no
14:15.13*** join/#asterisk Usernick124422 (~Usernick1@2.29.2.105)
14:15.19*** join/#asterisk navaismo (~navaismo@189.144.236.94)
14:16.40*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)
14:17.05Graiden[TK]D-Fender:  I'm working on giving it that ability, was curious if anyone had made that happen.  I'd like my users to be able to send an e-mail back to the e-mail that contains their voicemail and have it delete the message.
14:17.44indika1978[TK]D-Fender: Actually what is the class
14:17.55indika1978i didnt understand that clearly
14:18.24indika1978pls be kind enogh to explain me
14:18.27*** join/#asterisk Usernick124422 (~Usernick1@2.29.2.105)
14:18.57[TK]D-Fenderindika1978, Read the instructions for it.  there are 2 paramteres to pass
14:19.07[TK]D-Fenderparameters*
14:19.35[TK]D-FenderGraiden, Get programming ... * has nothing to do with the retun part.
14:19.49[TK]D-FenderGraiden, This is up to you and your MTA
14:20.29indika1978yes i ve red it says class and duration. what actually it means class in there? time is the holding time I know that
14:21.14jacobwi'm seeing "'/full/path/to/include' was listed an #include but does not exist" messages for files that are owned by asterisk.asterisk with mode 640
14:21.16indika1978this might be kinder question but pls help me to sort it out
14:21.44[TK]D-Fenderindika1978, Class ... as in the SECTION  of musiconhold that defines what you play
14:21.58jacobwwhat ownership and permissions does the configuration parser require?
14:22.12[TK]D-Fenderindika1978, You can have many DIFFERENT sets of MOH.  You tell it WHICH ONE to play, then for how long.
14:22.16*** join/#asterisk usernick124422 (021d0269@gateway/web/freenode/ip.2.29.2.105)
14:22.33indika1978ok ok thanks for that
14:22.34*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
14:22.34*** mode/#asterisk [+o putnopvut] by ChanServ
14:22.48[TK]D-Fenderjacobw, Whatever the user * runs as can read
14:23.39Graiden[TK]D-Fender:  I appreciate it.  I'll work on it and share when we have it perfected.
14:23.44jacobwyeah, that's what i thought
14:30.53indika1978I have done that but it wont act like musiconhold. It wont dial after put the current call on old. Pls help me on this....------http://pastebin.com/rijHPYyP
14:35.02[TK]D-Fenderindika1978, It looks like it plays music for 10 seconds then dials.....
14:35.28[TK]D-Fenderindika1978, If you want it to play music WHILE dialing, then that is a DIAL option <-
14:35.37[TK]D-Fenderindika1978, "core show application dial" <-------
14:36.03indika1978yes i have used dial there
14:36.29indika1978How could i do both things in same time
14:36.54indika1978i ve searched from top to bottom ... nothing fould
14:36.56indika1978found
14:38.13[TK]D-Fenderindika1978, "core show application dial" <-------
14:38.26[TK]D-FenderYou don't call MusicOnHold at all.  Just use the dial option.
14:39.34*** join/#asterisk luisgrin (bec0c24a@gateway/web/freenode/ip.190.192.194.74)
14:40.10indika1978yes then it connect the incoming caller to dialed extension .. according to my dial plan 4004
14:40.38[TK]D-Fenderindika1978, "core show application dial" <-------
14:40.50[TK]D-Fenderthere is an option to make it play MUSIC instead of RINGING.
14:42.39luisgrinhi, i have installed asterisk on woindows using asteriskwin32, i need some  help for  first steps .. im trying to develop an ivr .. i have a mysql database with data that comes for and php aplication
14:43.08[TK]D-Fender~book
14:43.08infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
14:43.11[TK]D-Fenderluisgrin, ^
14:43.18*** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com)
14:43.48luisgrin:) thanks a lot
14:50.51*** join/#asterisk esrax (hyrax@213.229.64.6)
14:51.11*** join/#asterisk chris_n (~Chris@nc-63-162-197-228.sta.embarqhsd.net)
14:51.38esraxcan someone clarify the c= line in SDP for me? if i have a host handling SIP and another handling media, the end point or proxy will expect that the c= line contains the IP of the media server? or am I wrong?
14:52.46*** join/#asterisk navaismo (~navaismo@189.144.236.94)
14:54.27pabelangeresrax: http://tools.ietf.org/html/rfc4566#page-14
14:57.41esraxpabelanger: thanks, i've read through that part of the rfc a couple times and it's still not specifically clear to me
15:07.49wdoekesesrax: proxies generally don't touch the sdp. the end point will expect the c= to contain the ip of the media server indeed
15:14.22esraxwdoekes: thanks for the confirmation, that's what i thought. been dealing with one way audio from a trunk provider. our sbc receives media from one of their servers but doesn't know what to do with it, and i was thinking i tracked it down to that media server never being defined in sdp in a c= line
15:14.52*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
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15:36.00eduzimrs[TK]D-Fender, there is such way to keep the exten 411 in CDR(dst) filed when i perform a GoTo ? -> http://pastebin.com/ksGef1Z3
15:36.37[TK]D-Fendereduzimrs, If that's what the app does, then apparently no.
15:37.52eduzimrs[TK]D-Fender, i`d like to keep the numbers instead MAC Addr in the CDR logs
15:38.15navaismoeduzimrs, what if you use a gosub or macro and then record in the CDR using the account code or userfield writing the exten?
15:40.13Kattyhello my asterisk does not work at all how to fix plz
15:40.42eduzimrsnavaismo, so i`d have to edit cdr_custom.conf and change the field CDR(dst) to CDR(uerfield) right? to keep the orrangement
15:40.49WIMPyeduzimrs: Or save the exten to another field, like accountcode.
15:41.15navaismonope i mean using the SET(CDR(accountcode) or SET(CDR(userfield))
15:43.06*** join/#asterisk CrazyTux[m] (~Brandon@108-64-56-113.lightspeed.irvnca.sbcglobal.net)
15:43.48navaismoI use this: http://pastebin.com/bRbZn3yh
15:43.55navaismolike WIMPy said
15:45.22eduzimrsWIMPy, navaismo; Ok i got it but i must keep my CDR arrangement > [mappings]
15:45.22eduzimrsMaster.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})}.....
15:46.23eduzimrsthe third field DST
15:46.35WIMPyThen change the mapping to get the data in the position it has to be.
15:47.28eduzimrstks
15:48.42Kattyhello my asterisk does not work at all how to fix plz
15:49.08WIMPyPut it out in the sun and let it melt.
15:49.23Kattythere is no sun today :<
15:50.11*** part/#asterisk esrax (hyrax@213.229.64.6)
15:50.58navaismoin the rain?
15:51.58[TK]D-FenderKatty, You must pound it with a 20lb sledgehammer until the evil spirits are released!
15:52.40*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
15:53.34CrazyTux[m][TK]D-Fender: LOL, good morning, great advice ;)
15:57.47Kattygoes to get sledgehammer
15:57.52Kattychases her boss out of her office with it
15:58.57MrPocketsKatty, can I borrow that when you're done.
15:59.34WIMPyHas your Asterisk been naughty as well?
16:11.10*** join/#asterisk wonderworld (~ww@dsdf-4db54347.pool.mediaWays.net)
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16:20.00kuruptionso i have this grandstream phone that supports 4 extensions
16:20.47kuruptionconected to asterisk of course
16:20.52kuruptionif im using extension 1
16:21.03kuruptionand someone calls it, will it show up as busy?
16:21.35WIMPyTry it.
16:21.43kuruptioni have outbound concurrency set to unlimited, but is there a limit on inbound?
16:21.49kuruptionhah well, i only have 1 phone right now :D
16:22.04navaismooriginate the call from console
16:22.19WIMPyWell, we don't know what your phone will do.
16:23.40kuruptionyeah ill set up a softphone
16:24.27WIMPyYou don't need a phone.
16:24.42WIMPy*CLI> channel originate ...
16:27.11kuruptionit's showing as busy
16:27.42WIMPyWell, there must be a reason for
16:27.49kuruptionso i guess i can set up a second extension... and have the first extension ring the second extension when it's showing up busy?
16:27.51WIMPy~gs
16:27.51infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
16:28.08WIMPySure
16:28.22kuruptionwell
16:28.35kuruptionwhen you're on a corporate budget, it's one thing.... but this is for home use
16:28.51kuruptionand i've heard good things about gs phones from others, soo
16:29.01drmessanoThere's other options
16:29.21drmessanoUsed Polycoms/Linksys/Cisco/Aastra, Yealink, etc
16:29.23[TK]D-Fender~gs
16:29.23infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
16:29.29[TK]D-Fender~grandstream
16:29.29infobotextra, extra, read all about it, grandstream is the Yugo of VoIP hardware.  Run...  Run away now.  Though, therealcircut says that they're not that bad.
16:30.04drmessanoI wouldn't buy a new Grandsuck phone because "this is for home use"... I got some nice Linksys SPA941s for home use for $35 a pop.. and they've been in service for 5 years.  Not the best phone in the world, but not a Grandstream
16:30.21drmessanoThere's always deals to be found
16:31.10WIMPyI haven't had a GS, but I found the Liksys really shitty.
16:31.26[TK]D-FenderWIMPy, Yuo need to lower your bar again ;)
16:31.41WIMPyThat was not my plan.
16:32.02*** join/#asterisk Bipul (~vipul@unaffiliated/bipul/x-4918593)
16:32.35ChannelZHmm haven't seen this one before -- " app_dial.c:901 do_forward: Not accepting call completion offers from call-forward recipient Local/xxx"
16:32.42WIMPyI'm used to some comfort and I donlt want to give up on that.
16:33.05navaismokuruption, what model of GS are you using I used the GXP2000 and the video one, on those you only configure one account and the others thake the same config
16:33.26kuruptiongxp2124
16:33.39kuruptionsupports 4 sip accounts
16:50.30*** join/#asterisk CrazyTux[m] (~Brandon@wsip-68-15-80-34.oc.oc.cox.net)
16:54.32*** join/#asterisk ipengineer (~ipenginee@static-71-252-134-63.dllstx.fios.verizon.net)
16:55.00ipengineerdoes anyone know how to enable "skip busy agents" on a queue? I cannot find any documentation but see it mentioned several times in the forums
16:59.51jpsharpI thought the queue always skipped busy agents.
17:00.05[TK]D-Fenderipengineer, Show us your now supposedly updated configs, queue status and a call attempt to back it up.
17:00.07[TK]D-Fender~pb
17:00.08infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
17:00.10[TK]D-Fender^^^
17:00.18[TK]D-Fenderipengineer, It works.. when you do your job....
17:01.38ipengineerIt does skip busy agents. If an agent is on a call their phone does not ring. My problem is if all agents are on the phone I need the queue to not let the caller join and go to the next destination. We don't want the user to have to sit and wait for the queue to timeout
17:02.47*** join/#asterisk serafie1 (~erin@75.76.38.159)
17:02.57[TK]D-FenderPastebin <-
17:05.22*** join/#asterisk brdude (~brdude@12.155.183.30)
17:06.55*** join/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es)
17:08.30ipengineerHere is the config. I will add the rest momentarily.. http://pastebin.com/CwVPzm70
17:13.52*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
17:13.54[TK]D-FenderWell so far that shows nothing at all....
17:20.20*** join/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es)
17:21.03*** join/#asterisk rgsteele (~rgsteele@apo.aweber.com)
17:22.44*** part/#asterisk jsjc (~Adium@237.157.78.188.dynamic.jazztel.es)
17:23.39ipengineerHere is the output and the queue with one call on and the other being sent to it. The second call was put on hold instead of going to the next destination.. http://pastebin.com/2gimsh07
17:25.28*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)
17:26.53[TK]D-Fender-- Executing [s@macro-call_agent:1] GotoIf("Local/779@bcposqe_queue_Test_agents-d4e1,2", "0?sendcall|1") in new stack
17:26.59[TK]D-FenderAgent isn't considered busy,.
17:27.02[TK]D-FenderIT CALLED HIM
17:27.03[TK]D-Fender^
17:27.29lorsungcuTK snapping off.
17:28.27[TK]D-Fenderipengineer, Now I also didn't see any of the actual queue config that I asked for half an hour ago, and you also haven't told us what version you're running or shown us how you logged that member in.
17:28.51*** join/#asterisk WolWid (WolWid@pD9ED49F8.dip.t-dialin.net)
17:29.25*** join/#asterisk Praise (~Fat@unaffiliated/praise)
17:29.43ChannelZWhy won't this thing just do what I want?  Plz help
17:29.54[TK]D-FenderChange what you want :)
17:30.11ipengineerno it did not send the call to him that was the original call. I can clearly see the response that the agent is busy
17:30.12ipengineer<PROTECTED>
17:30.13ipengineer<PROTECTED>
17:30.14ipengineer<PROTECTED>
17:30.18ChannelZuh oh
17:30.21*** join/#asterisk gusto (~gusto@ppp-93-104-83-136.dynamic.mnet-online.de)
17:31.08[TK]D-FenderResponse is irrelevant
17:31.21[TK]D-FenderIt called in the first place.  How a dial answers back is not a "state"
17:33.42[TK]D-FenderGo look at how you added them in the first place.
17:34.40[TK]D-FenderA phone could reply back "busy" with DND on a phone.  That also isn't a real "state", just a "what the phone is going to do if a call comes to it"./
17:35.03[TK]D-FenderNow if they were on an actual call then that is something you should be able to do something about.
17:35.13[TK]D-FenderAgain, go pay attention to how you added that member.
17:35.45ipengineerI dial *01+agent ID from the phone
17:36.03[TK]D-Fenderthat number doesn't mean anything yet....
17:36.37[TK]D-Fenderthe actual dialplan being called is what matters
17:38.24*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
17:43.54bitglueipengineer: there's an option in queues.conf to kick agents out of the queue if they don't answer.
17:45.40bitglueipengineer: it sounds like the problem is your users are being "away" by setting DND on their phone. This makes the phone reject incomming calls, but it doesn't notify asterisk, so asterisk can't do anything about it. You will need to either configure your phones to notify asterisk on DND, or find an alternate solution.
17:45.46[TK]D-FenderThere is also an option to look at an appropriate device when the one added isn't the best representative of it's state...
17:46.23[TK]D-FenderAlso I did not imply that the phone was on DND.  I gave one case where it might be relevant however
17:46.28[TK]D-FenderWe have no details.
17:46.49[TK]D-FenderNor the configs.  Or the dialplan that added them.
17:47.12[TK]D-FenderSo maybe we'll see what's going on in the bigger picture later...
17:48.54bitglueipengineer: did you know you can do "queue show" to see what the current state of your agents are?
17:49.32[TK]D-FenderDid you know he already pastebinned it?
17:49.39ipengineeryes and when on a call they are showing not in use which I think is prob the root of the problem. Here is the macro that logs agents in.. http://pastebin.com/AacHsHYv
17:49.57bitglue[TK]D-Fender: sorry, no. I haven't exhaustively read the scrollback.
17:50.10[TK]D-Fenderexten=>s,2,AgentCallbackLogin(${ARG1},,${ARG2}@${ARG3}) <- this has no state-based checks
17:50.17[TK]D-FenderAnd looks like * 1.4 junk
17:50.19[TK]D-Fenderor worse
17:51.07[TK]D-FenderAddQueueMember has an option for you to point to a better sample of the status like a related SIP device, etc
17:52.07*** part/#asterisk WolWid (WolWid@pD9ED49F8.dip.t-dialin.net)
17:57.10Kattyvirgina is scary.
17:57.36Kattymissouri is scary.
17:57.51Kattyi can haz safe place to live?! :<
17:57.55GraidenFlorida.
17:58.23lorsungcuick
17:58.25Kattyhurricanes seem safer at this point.
17:58.56GraidenHurricanes are fun, usually.
17:59.09GraidenWe don't have any deathtraps like New Orleans for the most part.
17:59.18lorsungcujust cannibals
17:59.20lorsungcunbd.
17:59.41GraidenOne hobo eats someones face and we're suddenly the cannibal state.  Jebus.
17:59.47ipengineerI inherited this system so I am sure it is pre 1.4.. Is there a way I can add a callback for the agent/device state on that dialplan. BTW we are on 1.4.29
18:00.17[TK]D-Fenderipengineer, No.  That is also outdated even for 1.4
18:00.18lorsungcu1.4.29 is definitely pre 1.4
18:00.19Kattywas that the bath salt incident
18:00.33GraidenKatty: Affirmative.
18:00.35lorsungcuKatty: no turns out he was totally sober
18:00.47Kattywell i knew he didn't actually  /eat/ the guy
18:00.48GraidenWell, okay... alleged bath salts.
18:00.50Kattyjust...maimed.
18:01.13Kattybut i still call it the bath salt incident.
18:01.29Kattyseems as though his toxicology report came back negative
18:01.30GraidenWe call it the Miami Zombie.
18:01.36lorsungcui call it the 'typical florida afternoon' incident.
18:01.41Kattyhee
18:01.47Kattybut yes, i am considering florida.
18:01.52Kattysomeplace with fall.
18:01.55lorsungcu...
18:01.57lorsungcucome to MN
18:01.58GraidenOh, that's not Florida.
18:02.06Kattywas considering perhaps jacksonville
18:02.07GraidenWe have "It's hot." and "OMFG MY SKIN IS ON FIRE."
18:02.14Kattytho their definition of fall is yellowish brown leaves
18:02.25lorsungcuor maybe don't.  not sure we need people who consider florida to be safe here :/
18:02.58[TK]D-FenderKatty, http://zipmeme.com/uploads/generated/g1341549768265453170.jpg
18:03.14Kattyreddit ^_^
18:03.31F2Knightuse to live in FL, (fort lauderdale) let me tell you … Miami, is another country and you live with the AC running 24/7
18:03.52Kattywas florida overly hot this year? like the rest of the usa, just about
18:04.28F2KnightIn an odd twist of events Florida was a tad cooler then the mid west.
18:04.50Kattythat's not saying much
18:04.55F2Knightbut Katty , there is no seasons for fall.. you get maybe 2 nights a year that get below 75
18:04.56Kattyconsidering we have record highs this year
18:05.18F2Knight10 months out of the year you have hot humid love for nature.
18:05.19drmessanoGeorgia has two heat settings.  SAUNA and SAUNA WITH THAT LITTLE BREEZE FROM THE DOOR BEING LEFT OPEN
18:05.28Kattygeorgia is scary.
18:05.32Kattyi'm not living there
18:05.41F2Knightis that banjo music i hear?
18:05.45Kattyrow faster.
18:05.47lorsungcuhttp://en.wikipedia.org/wiki/Climate_of_Minnesota
18:05.49drmessanolol
18:06.05drmessanoMost of Georgia is far scarier than Deliverance
18:06.20lorsungcuhttp://en.wikipedia.org/wiki/File:Minneapolis_Climate_Averages.png
18:06.23lorsungcuTHAT is seasons.
18:06.25Kattythat doesn't look lovely at all to me
18:06.55Kattythat looks like 50 weeks of fridged and 2 weeks of shiver.
18:07.02lorsungcuyeah that page reads like a 'disasters in MN' page..
18:07.13Kattyi would prefer more along the lines of 40 weeks of ahhh
18:07.20Kattyand 12 weeks of...bit nippy today!
18:07.36[TK]D-Fender28 weeks ..... grrrrarrrrrhghhhh
18:07.59Kattyi wonder what winter will be like this year
18:07.59Kattymild, or extreme
18:08.27drmessanoI hear that in Canada, its never summer.  They even have block heaters on their PBXs.  You should be thankful
18:08.53drmessanoActually, the block heater thing is a nasty rumor I started so I could sell PBX block heaters.
18:09.02drmessanoBut the rest is from the internet
18:09.11[TK]D-FenderHere in Quebec we have different seasons : Almost winter.  Winter.  Still winter. and Road construction.
18:09.20drmessanoROFL
18:09.55Kattybrace yourself. road construction is coming.
18:11.34*** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell)
18:13.21pabelangerFYI: We are seeking an integrator, see the following post: http://lists.digium.com/pipermail/asterisk-biz/2012-August/035931.html
18:15.02[TK]D-FenderKatty, TOO LATE
18:19.15*** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain)
18:19.46ujjainIs there much reason not to get a wifi voip phone?
18:20.04[TK]D-Fender~wifivoip
18:20.04infobot[~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended.  Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc
18:20.07[TK]D-FenderYMMV
18:20.28[TK]D-Fendercheck out reviews on your specific prospective model
18:20.45[TK]D-FenderAnd if you found "just some check model", be prepared for the worst
18:20.49[TK]D-Fendercheap*
18:21.36ujjainI have a PAP2T right now.
18:21.45ujjainI see, maybe I should just not change the set-up.
18:23.58drmessanoPAP2 and a $16 DECT phone works great
18:33.00*** join/#asterisk ChanServ (ChanServ@services.)
18:33.00*** mode/#asterisk [+o ChanServ] by zelazny.freenode.net
18:38.03*** join/#asterisk craig3smith (~craig3smi@ip67-90-212-69.z212-90-67.customer.algx.net)
18:47.28*** join/#asterisk GameGamer43 (users.5533@id-5533.hampstead.irccloud.com)
18:47.53KattyAHHHHHHHHHHHHHHHHH.
18:47.55Kattyahem.
18:47.59lorsungcueasy bro
18:48.01Kattygoes back to tickets.
18:48.07KattyI AIN"T YOUR BRO, BOR
18:48.08KattyBRO
18:48.11radenKatty, D
18:48.15Katty!@
18:48.17Kattyraden :>>>>>
18:48.19lorsungcu@#
18:48.22Kattyglomps raden
18:48.28raden:D :D :D :D
18:48.33radenhow have u been ?
18:48.42Kattyfrustrated, but not too bad. you?
18:49.04radenWisp is growing like crazy
18:49.15radenmy electronics business has been insanely busy
18:49.23radenmy daughter keeping me on my toes
18:49.34radenand I have more phone and camera systems to install than I can keep up with
18:49.44radenand I need a client billing solution for VOIP
18:49.54radenKatty, besides that life is peachy
18:49.55Kattysounds like cha-ching
18:50.08Kattyi am glad to hear it!
18:50.46radencha-ching cha-ching cha-ching :D
18:50.55radenbeen working 12+ hour days since thanks giving
18:51.05radennot a day without working yet
18:51.08radengetting a lil burnt
18:51.13radenbut will be worth it :D
18:51.16Graidenraden:  What kind of billing solution do you need?
18:51.32radenGraiden, just need to keep track of outbound minutes from customers phones
18:51.58GraidenDoes your * dump to a MySQL CDR?
18:52.00Kattyi think we just use account codes for that
18:52.19radenGraiden, if i could get asterissk to run off Mysql Id be set
18:52.27Kattyand areski
18:52.28radeni could then just pull a report each month
18:52.36radenaccount codes >
18:52.40radenso whats new with Katty ?
18:52.48Kattyhmm.
18:52.53Kattywhat's the last thing you heard from me?
18:52.58Graidenraden:  That's what we do.  Wasn't sure if you were setup as such, but the setup doesn't seem to be that complex.
18:53.05radenum i havent talked to you in like 4 - 6 months :(
18:53.11Kattynew boy.
18:53.17Kattykicked the old one to the curb.
18:53.18radenget a man :P
18:53.29Kattyi did not appreciate being second in his life
18:53.48Kattythat's about it
18:53.54Kattyoh and i swim now. lots
18:53.57Kattyoh and i got a boat.
18:54.39Kattyso not much new.
18:56.07lorsungcucoffee is done.
18:59.44radenKatty, swimming :D
18:59.47radeni love to swim
18:59.51radenI want to get a sailboat
18:59.54radena big sail boat
18:59.56radenand sail sail sail
19:02.43*** join/#asterisk ChanServ (ChanServ@services.)
19:02.43*** mode/#asterisk [+o ChanServ] by zelazny.freenode.net
19:06.05Katty:>
19:11.23*** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163)
19:14.07[TK]D-FenderSmall sailboat = very affordable
19:14.19*** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-mazgukbbybldndkp)
19:14.23[TK]D-FenderIf I had a garage I'd have a small fleet by now..
19:17.28*** join/#asterisk Sidrov (~sid@AC9E6419.ipt.aol.com)
19:17.47Sidrovhello all
19:18.02Qwellaol.com?  wtf
19:18.12Sidrov?
19:18.23Sidrovwhy not ?
19:18.34*** join/#asterisk Pigglesworth (~kamil@reactos/developer/pigglesworth)
19:18.36Kattybonks Qwell
19:18.38Kattybe nice!
19:18.43Qwellbut...AOL!
19:18.55Sidrovanyone here is using asterisk with gtalk ?
19:20.40QwellSidrov: https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
19:21.14Sidrovthis is not my question :)
19:21.44Qwellokay, so be specific.
19:21.51Sidrovjust curious if gtalk supports multiple channels
19:21.53*** join/#asterisk pepus (~chatzilla@gabcikova.p-h-a.net)
19:22.15QwellWhat does the documentation say?
19:23.25Sidrovi didn't find that answer, it might rely on Google
19:23.38Sidrovi guess they don't allow you dial multiple calls same time :)
19:27.02pepusI'm experiencing a strange Asterisk behaviour - as a phone call between two users starts, a CPU load of Asterisk is about 2%. But it is increasing every minute by 10%, so after about 10 minutes this single call drains all CPU power of the machine. I'm using self-compiled asterisk-1.8.15.0 under Debian, CPU is Intel Atom. Any suggestions what could be wrong? :(
19:27.42Qwellpepus: You didn't tell us what is involved in the "call".
19:28.07pepusSIP call between two soft phones registered to this Asterisk server
19:28.23pepusno encryption, codec G.729
19:30.27Sidrovpepus: you are not recording or using any agi scripting ?
19:30.55Sidrovhow much space left on hardisk ?
19:31.05pepusno
19:31.21pepus26GB free
19:33.13pepusand if I cancel the call the cpu load drops to 0%. but if I dial the call again, the CPU load resumes where it ended. If I restart Asterisk, I start again from 2%.
19:33.18Sidrovif using debian, why haven't installed it with apt-get ?
19:33.28pepusno, from source
19:34.16Sidrovadd this line to your sources.list file
19:34.19Sidrovdeb http://packages.asterisk.org/deb squeeze main
19:34.32pepusim not installing it from apt repo
19:34.41pepusi have downloaded source codes from asterisk
19:35.00Sidrovtry repo version
19:35.19Sidrov99% did something wrong at installation
19:35.19pepusthere is broken TLS support
19:35.38pepus./configure
19:35.39pepusmake menuconfig
19:35.41pepusmake install
19:38.09Jeff-workpepus, those aren't the right commands
19:39.04*** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163)
19:39.15SidrovQwell: you didn't answer me.. what do you think ? do they allow two calls same time going out to same google account ?
19:39.45QwellSidrov: I absolutely did answer.
19:40.05QwellJeff-work: O.o
19:40.23SidrovQwell: documentation say it ?
19:40.31pepussorry, what commands should i use?
19:40.35Jeff-workQwell, it's supposed to be make menuselect isn't it?
19:40.41QwellJeff-work: menuconfig works too
19:40.51Jeff-workoh, i hadn't heard of that
19:41.13Qwellthat's what the Linux kernel uses, so we added it as an alias.
19:41.53Jeff-workoh
19:41.54pepusCan the problem be in some options in config? I havent changed anything, except some G.729 stuff
19:42.04*** join/#asterisk _Corey_ (~chatzilla@64.215.11.114)
19:42.18Jeff-workpepus, what distro are you using?
19:42.28pepusdebian stable
19:42.33Jeff-workoh
19:42.40pepussqueeze
19:43.04Jeff-worki wrote a script that automates the installations for me... but it's for CentOS
19:44.10dijibanybody around to aid in a v10 upgrade?
19:44.31Jeff-workdijib, did something break? what are you upgrading from?
19:44.43dijib1.8.7
19:44.48dijibfax is lame.
19:44.49bitgluepepus: how are you measuring cpu load?
19:44.56dijibtop.
19:45.00pepusby top
19:45.06bitgluepepus: which number in top?
19:45.12dijiball 3.
19:45.15dijib5 10 15
19:45.23*** join/#asterisk bmg505 (~leon@196-209-84-253.dynamic.isadsl.co.za)
19:45.27pabelangerJeff-work: why write scripts when you can use Puppet
19:45.28Sidrovcould anyone suggest me some reliable sip/iax trunking / multiple channels usa provider ?
19:45.33pabelanger<3 puppet
19:45.45pepusbitglue: PU
19:45.47Jeff-workSidrov, for business?
19:45.49pepusPU
19:45.54pepusCPU
19:45.56Jeff-workpabelanger, never heard of that
19:46.06pabelangerhttp://puppetlabs.com
19:46.24pabelangerGreat tool to use for automation
19:46.39pabelangerHere is a bunch of manifests I use to automate my asterisk deployments: https://github.com/kickstandproject/puppet-modules/
19:46.47pabelangerso much so, I'm talking about it for Astricon
19:46.48bitgluepepus: i'm not familiar with a "PU" number in top. Can you pastebin a screenshot?
19:47.08SidrovJeff-work: business or small-business - i find with 1-2 channels same time, need better
19:47.10_Corey_pabelanger: My fiance found a business card from Puppet Labs in my pocket a couple months ago and thought I was buying her an actual puppet...  ;)
19:47.18bitgluepepus: except now i read your correction, and i wonder which "cpu" number you are referencing.
19:47.47Jeff-workSidrov, check out Intelepeer http://intelepeer.com/
19:47.52dijib>>Astriconhttp://puppetlabs.com
19:47.52bitgluepepus: you mean the CPU% column, in the row for the asterisk process?
19:47.59pepusyes
19:48.21Jeff-workSidrov, that's who i use for our  company, they're awesome! Great service, good quality, nice people too
19:48.21pepusand the load starts at 2% and continues to grow until the CPU is loaded at 200%
19:48.34SidrovJeff-work - thanks
19:48.35pepuswith just one call in about 15 minutes
19:48.40bitgluepepus: what do the "Cpu(s)" numbers at the top say?
19:48.57Jeff-workSidrov, go with their usage based billing, not their user based
19:49.06Jeff-workit's more flexible with asterisk
19:49.22dijib00.1
19:49.50pepusbitglue: after 1 minute it is about 10%
19:50.04bitgluepepus: what is 10%?
19:50.24bitgluepepus: be *specific*
19:50.34bitgluethere are like, 20 numbers that could be "cpu load"
19:50.49bitglueeven within top, there are a handful. They all mean something different.
19:51.02pepusthe percent CPU column in top
19:51.30QwellWhich one?
19:51.39bitgluepepus: my top has 8 numbers there.
19:52.05Qwellpaste the line...
19:53.36pepushttp://pastebin.com/uFkwJWNm
19:53.53dijib<PROTECTED>
19:53.55pepusnow it is 4% as i started the call
19:54.15dijibthats good pepus
19:54.18dijibwhats your issue?
19:54.22dijibOI?
19:54.38pepusno thats not good, in 10 minutes it is 200% with this single call
19:54.47bitgluepepus: great. show us.
19:54.55dijibwhat applications are you envoking?
19:55.04dijibMixMon? Monitor? what?
19:55.28pepusnone
19:56.02Qwellshow us
19:56.32Jeff-workis it normal for * to use 3688m of VIRT memory with 3 active calls?
19:57.54bitglueJeff-work: the amount of virtual memory allocated to a process has little to do with RAM usage. It could be, for example, entirely memory-mapped files.
19:58.04bitglueJeff-work: so the answer is, "maybe".
19:58.14Jeff-workok, no worries then
19:58.16Jeff-workthanks
19:58.24dijibwhat codec are you using?
19:58.32dijibis that relevent?
19:58.32Jeff-workeverybody talking about TOP, i had to pull it up and look at my own
19:58.35Jeff-workdijib, ulaw
19:59.29pepusafter 2 minutes http://pastebin.com/Dk3gYP1m
19:59.38pepusand keeps growing
19:59.38dijibhttp://pastebin.com/Dk3gYP1m
20:00.10pepusnow its 15%
20:00.16pepus16%
20:00.20dijibwhats your cpu?
20:00.38pepusIntel(R) Atom(TM) CPU D510   @ 1.66GHz
20:00.48dijiband the how.
20:00.52dijibvmm?
20:00.54Qwellsys?
20:01.17pepus2GB RAM
20:01.41dijibwhats the atoms L1 L2 chache?
20:01.46pepusnow CPU at 21% (still one single call that started with 4% load)
20:01.47bitgluepepus: it's odd that asterisk is using 12% of the cpu, but the cpu is 94% idle.
20:01.56dijibs: it's odd that asterisk is using 12% of the cpu, but the cpu is 94% idle.
20:01.58bitgluepepus: let's see top's output when it's very bad.
20:01.59dijibsry
20:02.08dijib>> load average: 0.00, 0.01, 0.00
20:02.28pepusthe machine has 2 cores and hyperthreading, so 4 virtual cores
20:02.37Qwellpepus: all the time is being spent in the kernel..
20:03.08dijibsame with my Core duo. let me test this
20:04.28pepushttp://nopaste.info/a750d789c0.html
20:04.30wdoekespabelanger: that kickstand project wiki has been raped
20:05.40pepusbut as I waited for 200% load the voice call started to interrupt
20:06.36Qwellpepus: press '1' in top, and paste again
20:06.58dijibhttp://nopaste.info/26f1b67e7b.html
20:07.12pepushttp://nopaste.info/b549db2a01.html
20:07.36Qwellwell there's your problem
20:08.02pepuswhere?
20:08.12QwellWhy would it be spending so much time in the kernel?  You've still not told us anything about your setup.
20:08.51bitgluepepus: all of the cpu usage is in the kernel. So, it's maybe a hardware driver, or something else the kernel is doing.
20:09.02bitgluepepus: ie, it's not anything asterisk is doing directly.
20:09.26bitgluepepus: any errors in dmsg or syslog?
20:09.41pepusmy setup is pretty standard: http://nopaste.info/7aac3b0b05.html
20:10.12bitgluepepus: are you using any telephony hardware?
20:10.19pepusno
20:10.26pepussoftphone client bria
20:10.37sruffellany crazy iptables rules?
20:10.57dijibis this a new instalation?
20:11.08pepusnot at all. only im connected using a USB 3g modem, could it be the problem?
20:11.13pepusnew install
20:11.18QwellDefinitely.
20:11.22sruffellwhat he said
20:11.41pepuswill check it again localy over ethernet, one moment
20:12.05pepusI would understand it if the load would be constant
20:12.17pepusBut why is it still growing
20:13.07bitgluedriver bug? bad hardware?
20:13.22QwellCrap drivers have crap behavior.
20:13.28sruffellsampling errors with how si% is calculated?
20:13.35QwellIf it's a problem, talk to the manuf
20:14.12bitgluesruffell: on each timer interrupt, the kernel increments a counter reflecting what it did for that jiffy.
20:15.10bitgluesruffell: i don't think it's subject to sampling error
20:15.13sruffelltrue..but heavy workloads in hi context, on certain kernels, aren't accounted for properly and can show up like this.
20:15.15bitgluebut i'm not entirely sure.
20:15.25sruffellunless you have IRQ_ACCOUNTING setup in your kernel.
20:15.32sruffellbut that's not available on any distro kernels that I know of.
20:16.19bitglueyou mean CONFIG_IRQ_TIME_ACCOUNTING ?
20:16.25sruffellnods
20:16.41bitgluewell, sounds like you know more about it than i do
20:16.54bitgluethough it's off on my debian squeeze-backports kernel
20:20.24pepusok running the same test over ethernet
20:20.33pepuscpu load 0% so far!
20:21.51pepusok because it does reinvite, must disable it to simulate the same NAT scenario like in 3G case
20:24.55bitgluepepus: i'd maybe look at a packet trace with wireshark on your 3g card
20:25.31bitgluemaybe some combination of small pipe and crappy drivers is leading to a compounding problem with congestion
20:25.40bitglueor something equally fun
20:25.48pepusok I have disabled canreinvite and now its asterisk 1% lets see what will it do
20:25.50Kattyhello my asterisk does not work at all how to fix??? answer plz.
20:27.29*** join/#asterisk Galen_ (~Galen@rrcs-24-43-17-235.west.biz.rr.com)
20:30.03pepusgreat, still 1%!
20:30.34pepusso thank you, im going to kill my voice operator for selling me that crapy USB 3G
20:31.05bitglueKatty: begin by reading this: http://www.catb.org/esr/faqs/smart-questions.html
20:35.01Kattywhat on earth is that.
20:35.11Kattygiggles
20:35.15Kattyoh you thought i was serious
20:35.18Kattypats bitglue
20:35.21Kattythat's adorable.
20:35.22Katty:>
20:36.28*** join/#asterisk jakent (~jakent@c-71-63-6-140.hsd1.va.comcast.net)
20:38.01sruffellpepus: before you kill your voice operator, you may want to confirm that you have the most recent driver for the modem from the manufacturer.
20:39.13pepuslooking for it, it is some T-mobile branded modem and it "worked" with drivers provided by debian
20:43.36bitglueKatty: guess i missed whatever you were mocking :)
20:44.14Kattypfff, i miss myself coming and going
20:44.19Kattyif you see me, tell me to go home!
20:57.18radenKatty, Boooooooooooooooooooooooo
20:58.03Kattyraden: ohai
20:58.09Kattyraden: i was just about ready to go home
20:58.15Kattyraden: and eat bratwurst.
21:04.24*** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com)
21:06.12*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:06.28*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
21:07.25kuruptionhow does asterisk determine if a given extension is busy?
21:07.39Kattydialstatus, i believe
21:07.52WIMPyExtensions are never busy.
21:08.09Kattyand wimpies are never smart asses.
21:08.14WIMPyDevices/peers may be. They descide themselves.
21:08.22Katty<3
21:08.52kuruptionhow does asterisk determine if it can ring an extension or send it to voicemail?
21:09.03kuruptionis that a better question, question nazi?
21:09.05[TK]D-FenderIt tries and sees if it fails
21:09.16*** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez)
21:09.27[TK]D-FenderAnd it doesn't "send" to voicemail... this is your dialplan.. if YOU call voicemail after a Dial... that's up to you
21:11.20kuruption<PROTECTED>
21:11.20kuruption<PROTECTED>
21:11.47kuruptionso how is it determining if it's BUSY exactly?
21:11.52kuruptioni don't see it trying it in the logs
21:12.26kuruptiondo i need to increase verbosity to see that?
21:13.31[TK]D-Fenderkuruption: We can't see what that GotoIf is comparing....
21:13.41[TK]D-FenderOr what else that NoOp is expected to show if anything.
21:13.44[TK]D-FenderPASTEBIN is your friend
21:13.46[TK]D-Fender~pb
21:13.46infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
21:13.48[TK]D-Fender^^^
21:14.10[TK]D-Fenderkurrand that looks very much like FREEPBX dialplan
21:14.16[TK]D-Fenderkuruption: and that looks very much like FREEPBX dialplan
21:14.25radengood cheap SIP phone that not polycom ? any recomendations ?
21:14.34kuruptionit is freepbx
21:14.41tm1000raden:  yealink
21:15.11[TK]D-Fender~freepbx
21:15.11infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
21:15.35[TK]D-Fenderkuruption: First that is only a fraction of the actual call debug.  Second... if there is anything wrong in their dialplan, it isn't supported in this channel
21:16.31kuruptionwell, that's why im asking the questions... im trying to understand how it is figured out
21:16.44kuruption(im trying to understand how asterisk works here, not freepbx)
21:16.57Qwellkuruption: Using FreePBX is the worst way of going about that.
21:17.05kuruptionsighs
21:17.19Qwell~book
21:17.19infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
21:17.21Qwellmuch better way
21:17.44navaismokuruption, freepbx uses custom macros so go to the extensions_additional to find how freepbx handle taht
21:17.51tm1000Qwell is a developer.......kuruption ....listening to him is a good idea
21:17.52Qwellleifmadsen: I want a shiny nickel for every 100 times I link that.  Just FYI.
21:18.23radenHow the F do i change ring tone on these POS polycoms I have ? 321's
21:18.37kuruptioni really don't care who he is
21:18.39*** join/#asterisk longst (~longst@c83-250-121-107.bredband.comhem.se)
21:18.52kuruptioni don't want to dive into the inner workings of asterisk beyond getting what i need done
21:19.03kuruptionso, while i understand freepbx is not supported *here*
21:19.05QwellYou just said you wanted to learn how Asterisk works.
21:19.17longstAnyone has experience about Digium PRI card?
21:19.20[TK]D-Fenderkuruption: Laymans terms : FreePBX is 10 tons of convulted dialplan BEFORE even getting into their AGI's.  That's like trying to learn how to bake a cake .... starting with a book on advanced particle physics... cause you know .. you need like MATTER and stuff to make your cake from.
21:19.31kuruptionid like to understand why AsteriskNow distributes FreePBX but you then tell me it's not supported
21:19.33kuruptionbut whatever
21:19.35kuruptioneither way
21:19.37Qwellkuruption: AsteriskNOW != Asterisk
21:19.41kuruptionim not playing this nazi game
21:19.50tm1000kuruption:  look. NoOp means NO OPERATION. Its like a comment. Comments are throughout freepbx
21:20.03kuruptioni've been done playing this help channel bullshit 15 years ago.
21:20.12WIMPy~ask
21:20.12infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
21:20.16WIMPylongst: What infobot said
21:20.46NuggetIS ANYONE HERE AN INFOBOT EXPERT?  :)
21:20.51QwellNugget: MAYBE, why?
21:21.09WIMPyNugget: infobot self?
21:21.24[TK]D-Fenderkuruption: That ISO comes with a whole pile of CentOS... does that mean that we owe support for every single little binary it comes with?  The fact they threw some stuff in for free on the side doesn't make THIS the place for support on it.  Remember that FreePBX was an OPTION
21:21.36[TK]D-Fender~infobot
21:21.36infoboti guess infobot is me, http://infobot.svn.sourceforge.net/svnroot/infobot/ // The original infobot is no more, it has been replaced by the latest version of blootbot, accompanied by a rename back to infobot // apt/ibot/infobot/purl is now an infobot run by TimRiker
21:21.40QwellNugget: also if it's super complicated, TimRiker is a pretty nice guy, and is usually willing to help.  I'm pretty sure the bot also has a channel
21:21.46tm1000kuruption:  and this states ""1?s-BUSY,1"" go to 1 if 1 is not busy
21:21.48[TK]D-Fender~ibot
21:21.49infobotwell, ibot is not a help system
21:21.51tm1000else go to voicemail. I assume
21:21.55navaismokuruption, you can go to the freepbx channel
21:21.59kuruptionso basically, im trying to understand if this is a problem in freepbx, then i can go that route... but i don't know that, and neither do you... im just trying to figure out why asterisk won't notify my phone of a call waiting event
21:22.11[TK]D-Fenderhrm, what was the old name again?
21:22.15Qwell~jbot
21:22.15infobotit has been said that jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch, or suck, or a pain in the ass, or ibot's stupid cousin
21:22.23kuruptionis that because of a bad dialplan?
21:22.24[TK]D-Fender!
21:22.30[TK]D-FenderQwell++
21:22.34[TK]D-FenderNugget: MAH BITCH!
21:22.46Qwellkuruption: Possibly - they can answer your questions in #freepbx.
21:22.47[TK]D-Fenderkuruption: PASTEBIN the COMPLETE call
21:22.50tm1000kuruption:  you gave us two lines of dialplan, then stated you use freepbx. you won't find your answers in here
21:22.53Nuggetheh
21:23.05tm1000or post more
21:23.10[TK]D-Fender~pb
21:23.10infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
21:23.11[TK]D-Fender^^
21:23.16kuruptionsighs
21:23.44[TK]D-Fenderkuruption: for reference, there is no such thing as "notifying your phone of call waiting".
21:23.51Qwellkuruption: You are not entitled to help.  Your sighs do nothing but annoy us further.
21:23.56[TK]D-Fenderkuruption: The call either gets sent to the phone or it doesn't
21:24.02kuruptionokay well
21:24.04QwellAsk your question in the correct place, and maybe you'll get help.
21:24.04kuruptiondo me a favor
21:24.18[TK]D-Fenderkuruption: If the phone is capable of handling multiple calls it will deal with it in its own way
21:24.18kuruptionand stop addressing me for a few minutes while i figure out why my inbound has stopped working
21:24.29kuruptiononce i figure that out, it may fix my other issues
21:24.40kuruptionbut until i figure that out, it's worthless diving any deeper into this
21:24.59[TK]D-Fendersure
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21:26.21luisgrindo you know what happens with the site asterisknow.org??? chrome and firefox told me this is a very danger site
21:26.40Qwellluisgrin: asterisk.org/asterisknow/
21:27.46luisgrini think i got there looking for a distro of centos and asterisk all in one.. anyway i got it from other site
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22:20.36sjenningshas anyone gotten chan_motif working with google voice?
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22:22.37[TK]D-FenderYou mean the brand new one that hasn't hit full release and isn't supported here yet?
22:22.47sjenningsyes that one :P
22:23.08[TK]D-Fender#asterisk-dev
22:23.10[TK]D-Fender^
22:23.17sjenningsthat was my next question
22:23.19sjenningsthanks
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22:31.31volga629Hello Everyone,  I see this messages ignoring 'video' media offer because port number is zero
22:31.39volga629what is mean ?
22:33.11ChannelZIt is ignoring an offer to provide video media because the port number given was 0.
22:33.13ChannelZWhat do I win!?
22:38.01lorsungcuwhere'd kuruption go?  that shit was great.
22:38.08volga629RTP port  I am just trying use video, on payied version working no problem with h264 on free version with h263 nothing
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22:38.53lorsungcudude has BEEN been done with help channel bulllllshit.
22:47.10volga629what can cause it usually ?
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22:49.39SeRiload nicklist
22:49.44SeRilol
22:55.03*** join/#asterisk leocassarani (~leocassar@unaffiliated/cassarani)
22:55.04ChannelZshrugs - are the codecs limited in the config? what phone/client are you even using?
22:56.00leocassaranihi, so I've kind of inherited this Asterisk system which works just fine, except for one annoying thing I'm hoping to fix -- basically, we have about 20 ("landline") numbers that we redirect to employees' mobile numbers
22:56.14leocassaranithe way we do it is exten => 020blahblah,1,Macro(inbound_user,user.name)
22:56.36leocassaraniand the macro does essentially Goto(users,${ARG1},1)
22:57.06leocassaraniand then under [users] we have exten => user.name,1,Dial(${OUTBOUNDTRUNK}/+447blah)
22:57.32leocassaraniso far so good, except I'd like the caller ID to be kept the same when dialling the employees' mobile phone
22:57.46leocassaranibut I can't see how to do it as a Set(CALLERID) call
22:57.59ChannelZSet(CALLERID(num)=whatever) and then Dial
22:58.18leocassaranibut how do I get "whatever"? I mean I'd like the caller ID to be the one that called our 020 number
22:58.34leocassaraniobviously that changes every time, it's not a static value I can just put in
22:58.47ChannelZoh, passthru..
22:59.02leocassaranidid I mention I'm quite new to Asterisk? :(
22:59.17WIMPyAsterisk won;t change the number on its own.
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22:59.36WIMPyBut ypur provider might not allow you to send the original callers number.
22:59.44ChannelZyeah
22:59.49leocassaranioh right
22:59.57leocassaraniwhy would they do that exactly?
23:00.07WIMPyIs that a line or an ITSP or what?
23:00.17WIMPySo that people don't fake caller IDs?
23:00.33ChannelZLots of providers prevent you from changing your CID
23:00.36WIMPy(unless they pay for that service, off course)
23:00.38ChannelZ(or rather, just ignore you)
23:00.44leocassaraniI believe we're using Gradwell
23:01.00ChannelZWhat is ${OUTBOUNDTRUNK}?
23:01.18leocassaranigood question
23:01.24ChannelZSIP/something ?
23:01.29leocassaraniwhich configuration file could I find that in?
23:01.32leocassaraniah, wait
23:01.36leocassaraniIAX2/gradwell-out
23:02.06ChannelZok so Voip anyway.  Well you'd have to ask them.
23:02.12leocassaranioh right, I see
23:02.20leocassaraniI was hoping it was a simple misconfiguration issue
23:02.37leocassaraniI'm sorry, my questions must sound so bizarre, as I said I inherited a system I don't fully understand
23:02.46leocassaranithis is me trying to learn by changing things :)
23:02.49WIMPyIf you don't change the caller ID, no.
23:03.28leocassaraninot in the case of that inbound_user macro, no
23:03.32ChannelZI assume right now the caller ID shows up as your own DID number?
23:04.12leocassaraniyeah, unless I overwrite it of course, it's a US number
23:04.33ChannelZWht do you mean 'unless I overwrite it'
23:04.45kuruptionso i discovered that my DND was set on the extension... but asterisk wasn't saying that in the call log
23:05.01leocassaranisorry, that was poorly phrased -- rather than have it show up as our DID number, we have it set up so the extension that was called is actually set as the caller id
23:05.09ChannelZkuruption: It doesn't necessarily know
23:05.15leocassaranibut of course that's quite sub-optimal, ideally I'd like the originating caller ID
23:05.20leocassaranihence my questions
23:05.28kuruptionasterisk doesn't necessarily know i set DND on asterisk by doing *97?
23:05.39WIMPyleocassarani: Talk to your provider.
23:05.48lorsungcukuruption: is that how it was set?
23:05.49leocassaraniWIMPy: got it, thank you
23:05.52leocassaraniand thank you ChannelZ, too
23:05.53kuruptionyes it was
23:06.13ChannelZ*97 doesn't mean anything
23:06.17ChannelZYou must be talking of FreePBX again
23:06.22leocassaraniWIMPy: on that note, is there a name for what I want from my provider exactly?
23:06.28ChannelZor *97 is a feature code of the device its self
23:06.45kuruptionsorry, freepbx set dnd toggle to *76
23:06.50kuruptionthat's how it was set
23:06.52ChannelZstill
23:06.56kuruptionis that something specific to freepbx?
23:07.06WIMPyleocassarani: No idea, what ITSPs call it. For classic telcos it's called "CLIP (no screening)".
23:07.23lorsungcukuruption: you answered your own question
23:07.25ChannelZAsterisk has no built-in DND function I'm aware of.
23:07.38leocassaraniWIMPy: ok that's fine, I'm sure they'll understand
23:07.40kuruptionso freepbx creates a a dialplan for DND?
23:07.44leocassaranithanks again, good night
23:07.52ChannelZFreePBX creates lots of dialplan for a lot of crap
23:08.05kuruptionah
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23:08.39kuruptionokay i found it
23:09.13kuruptionwell that solved my problems
23:09.14kuruptionthx
23:09.15ChannelZThat's part of what FreePBX is, besides the GUI it's tons of macros and prebuilt dialplan mess
23:09.24kuruptionyeah
23:09.33kuruptionTHAT is what im starting to understand
23:09.33kuruption:D
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23:16.26Kattymmm chocolate chip cookie and pumpkin ale
23:16.47radengah freaking file system is full :(
23:16.48radenFML
23:16.53radenblaaqaaaaa
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23:27.59carrarKatty!
23:28.01carrarPics!!
23:28.29joobiehey guys.. i want to setup so that a user can input in a code into the phone, and that allows calls to come into the office.. then if another code is dialed it makes calls go to voicemail.. i dont wnat to use a queue as I wanted to only dial particualr phones around the office for 30 seconds and then go to voicemail if it's not answered.. any ideas on how ot best set this up at a high level?
23:28.51joobiei was thinking to use the asterisk-db to write out a variable which is looked up on each inbound call.. if it's set go to one context if it's not set go to the other.. but not sure if it's the best way
23:30.00carrars/code/extension/ ?
23:30.52navaismolike using DND button in the phones?
23:31.19F2Knightjoobie:  the quickest way is to just do a DIAL(SIP/ext,25) followed by the next and the next and the next in the order you want ending with VoiceMail(u,extension)
23:31.35carrarjoobie, you mean like a night mode switch
23:31.41carrarthat can be set at any time
23:31.47carrarby dialing a extension
23:32.42carrarI'd use a db to store that value
23:32.53carrarthen you could also set via a web page if you wanted
23:33.26carrarunlimited options!
23:33.44carrar... hello?
23:33.50joobiewhen the office is open, it will just do something like DIAL(SIP/ext&SIP/ext&SIP/ext,30) for a few phones.. if no answer, then it goes to voicemail
23:33.56joobiewhen the office is closed, it would just go to voicemail
23:34.07carrareasy as pie
23:34.10joobiewhat determines if the office is closed / open is controlled by a code dialed into the phone
23:34.23joobieso when staff arrive, they have to switch it to open.. when they leave, they switch to closed
23:34.32joobiereacon the asteriks-db is the best way to do it?
23:34.33carraryup
23:34.41joobieand just reference the variable in the dialplan to send it to a different context?
23:34.51carrarI, personally would use a external db as I would do other things
23:35.03joobieother things like what?
23:35.05carrarthen just check that for every incoming call
23:35.17carrarlike I said, if you wanted to also make it controlled via a web page
23:35.23carraror something source
23:35.28carrarsome other
23:36.06carrarcould use the astdb also
23:36.42joobiemight just go with astdb
23:36.53joobieno real need to make it web based
23:37.21joobieany idea if it would be possible to use the WMI on the polycom 321 to turn on and off in line with the queue being open / closed ?
23:37.41carrarnever tried
23:37.55joobiewould be ideal to sorta see that it's open / closed
23:38.11carrarWhat if that phone got a actual vm?
23:38.17joobievoicemail to email setup
23:38.44ChannelZI made a simple nightmode/daymode switch a couple of years ago and then have never used it
23:38.54carrarheh
23:39.21carrarwe have a few customers who had to have it too
23:39.24carrarand never use it
23:39.28joobieChannelZ, with WMI on the polycoms?
23:39.38joobiethese guys want ot use it for some reaosn
23:39.47joobiei reacon it'll be more of a PITA .. but oh well
23:39.48ChannelZno
23:40.01ChannelZno blinkenlights sorry
23:40.11joobieit's all about the blink blink
23:40.16carrarconnect it to flashing strobe lights
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23:40.21joobiehaha
23:40.23carrarand BUZZERS
23:40.26carrarwith flames
23:40.27joobienow ur talken
23:40.39joobieand naked chicks who get prompted to come out and dance on the desks
23:40.49carrarpopup cake dancers
23:40.57joobie:)
23:41.05ChannelZFor whatever weird reason though the Linksys SPAs have a switch in the web interface to arbitrarily turn the MWI on and off.  Dunno if you can do it with secret SIP messages, short of an actual MWI notification
23:41.07joobiebetter yet, that chick from that stephen segal movie
23:41.11joobiewho jumped outa the cake
23:41.29joobieshould be possible
23:41.30carrarhaha ChannelZ, yeah I never understood that
23:41.37joobiei mean voicemail sends the requedst to the phone
23:41.46joobiejust gotta see what it's sending and hopefully asterisk can send it manually
23:41.59joobieit's done with SIP OPTIONS packet
23:42.06joobielooked into this a year or so ago..
23:42.12[TK]D-Fenderjoobieany idea if it would be possible to use the WMI on the polycom 321 to turn on and off in line with the queue being open / closed ? <- No, use up a line-key for a BLF for it
23:42.26[TK]D-Fenderjoobie: Or better yet, a MicroBrowser idle page.
23:42.38joobiehey TK
23:42.39carrarMicroBrew!
23:42.45[TK]D-FenderYou have 2x17 to work with for that
23:43.19joobieTK, what do you mean by use up a line-key for a BLF?
23:43.31joobiecurrently i have the single registration assigned to both line keys
23:43.59[TK]D-Fenderuse just 1 line key and you can use the other for a speed-dial / BLF ansd use a DEVICE_STATE flag to light it up
23:44.08joobieMicrobrowser idle page sounds interesting too..
23:44.22[TK]D-FenderOr better still, keep the key as is and us a MicroBrowser page to display the status
23:44.31joobieyea
23:44.34joobiesounds like a good option
23:44.48joobiebut TK, this WMI light - can it not be done on the polycom 321?
23:45.08joobiemicrobrowser is a bit more work than WMI - and i think both are equally as good a solutin
23:45.14[TK]D-FenderNot AFAIK.  I don't believe it's assignable
23:45.15joobie.. given the users dont use voicemail on the phones
23:46.20joobiebrain is going off in a tangent, thinking that i could create a dummy voicemail and assign that to each sip reg, then when the office is open, it creates a viocemail in that dummy voicemail which will get asterisk to light the WMI on all phones.. then when it's closed, flush all voicemails to remote the WMI
23:46.24joobiesounds messy tho..
23:48.39joobieis v10 considered stable these days?
23:48.44[TK]D-Fenderyes
23:48.46joobieim still running 1.4..
23:48.58joobieneed ot update
23:49.15joobiei think the biggest feature im going to love is the ability to adjust call volume via the dialplan
23:49.20joobiesuch a pita at the moment wiht 1.4
23:49.21ChannelZcould you fudge MWI's with the console 'sip notify' command?
23:50.10joobienot sure ChannelZ but i'll look into it
23:50.18joobiethanks btw, i was thinking it was sip options
23:50.20ChannelZI mean the sample sip_notify.conf has a 'clear MWI' message already. Assuming the phone can be tricked.
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23:50.29ChannelZtries
23:52.23ChannelZhah worked for me
23:52.49ChannelZ[set-mwi]
23:52.49ChannelZEvent=>message-summary
23:52.49ChannelZContent-type=>application/simple-message-summary
23:52.49ChannelZContent=>Messages-waiting: yes
23:53.32ChannelZis all it took.. then 'sip notify set-mwi Bob' on the console
23:54.05joobieBob being your SIP identifier?
23:54.14joobienever played with sip_notify.conf before
23:54.57ChannelZyes
23:55.07joobie.. can you trigger the sip notify mesasges via the dialplan, without an external script?
23:55.11ChannelZThough I guess you'd have to set it off via AMI...
23:55.51joobiemay be able to just create a [set-wmi-off] and put Messages-waiting: no for an off state
23:56.14ChannelZor you could do System(asterisk -rx sip notify blahblah) from the dialplan perhaps
23:56.29joobieyer
23:56.45ChannelZYeah depends on how picky your phone is
23:56.49joobiejust googling into it and that looks like how people achieve it from the dialplan
23:57.21joobiewell, thanks a heap
23:57.35joobiemore than enough info to wrap this one up.. ill feed back how i go :)
23:58.13ChannelZsure good luck

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