IRC log for #asterisk on 20090102

00:06.29*** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my)
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00:27.57Jabessexist some module that control the noise?
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01:21.28drmessanoNoise?
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01:28.13Hanif08hi team... can someone torch me a light.. where a good place i can check on asterisk call features (ie cfu (call forwarding) park and so on)
01:28.28Hanif08maybe an example on how to do ...
01:30.05eppigyFEATURES.CONF DOG
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01:34.02Hanif08ep
01:34.17Hanif08eppigy.. thanks
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01:37.22eppigynp
01:38.09spyder12345quick question for the experts. Is there a way to remove a sip header that has been added? I see there is sipaddheader, sipgetheader, but is there a remove or delete?
01:39.19drmessanoIt is dark.  You are likely to be eaten by a grue.
01:43.25spyder12345apparently the experts are missing....
01:44.15jayteespyder12345, nope, no remove or delete
01:45.36drmessanospyder12345: Thats not insulting at all
01:45.47spyder12345Jaytee.. Any suggestions then on how to stop a header from being passed onto another system?
01:45.59drmessanospyder12345: Have the other end not send it
01:46.03drmessanospyder12345: That works
01:46.55jayteelet's all mail someone a letter but leave off the address!!! I've got a few extra stamps laying around here somewhere.
01:47.52drmessanoWhy the hell do you want to REMOVE a sip header?
01:48.03drmessanoIts bad enough that apps dont send half the damn headers they need
01:48.09drmessanoNow you want to remove some
01:49.22spyder12345well I have a diversion header that is added in order to communicate with exchange um but then when I redirect a call back through to a cisco platform I need it removed because the cisco box doesn't interpret it right.
01:49.41jayteeand you won't find many "experts" in here. Those people make huge wads of cash and are off golfing in Cabo San Lucas or St. Kitt's
01:50.07Techie-MichealYou weren't supposed to tell them where I am. :( ;)
01:50.08jayteecisco platform?
01:50.10drmessanoSo create another peer for the Cisco
01:50.43jayteeI use Exchange UM with sipX and all our switches are Cisco and I have no problem with redirecting calls
01:51.20drmessanoI think he means a CCM or UC500
01:51.26spyder12345yah its a callmanager box. I have two seperate peers. but for the header still remains.
01:51.28jayteemust be
01:51.46spyder12345I have it working currently without sipX just this one problem.
01:51.48drmessanospyder12345: It doesnt remain unless you are using the wrong peer
01:52.00drmessanospyder12345: or the dialplan is hosed
01:52.31drmessanoIf you are using the common UM setup, only calls directed to VM would add that header
01:55.14*** join/#asterisk dssman (n=no@c-98-203-109-90.hsd1.fl.comcast.net)
01:55.30spyder12345hmm... I have it so a call comes in through CM to asterisk and then asterisk adds the rdnis info in the diversion header to identify the correct mailbox. However if you use the exchange directory and attempt to call back from exchange through asterisk to CM that header still remains and I get a error response from CM..
01:56.11spyder12345obviously a mistake on my part probably but I have separate peers for the cm and exchange.
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01:56.33drmessanoWelllll
01:56.52drmessanoExchange > Asterisk > CM is likely going to add that header if it doesnt exist
01:58.52drmessanoYou can always add another box with SipX on it, strip the headers out to Cisco
01:59.47spyder12345well it apparently seems to be the header that I am adding manually and not anything put in by any of the systems because if I don't add it initially it works just fine.. Only problem is when the call is passed through asterisk to exchange it doesn't know which voicemail box to go to.
02:00.53spyder12345so then I added in the header to attempt to fix that and then it created this problem.. I thought it was probably something simple that I wasn't doing or doing wrong.. I was attempting to get it all working without adding in another piece.
02:08.32drmessanoshoots spyder
02:08.40drmessanoyells "dance"
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02:10.38jayteelol
02:10.54drmessanoGotta love that movie
02:11.05jayteewhich movie?
02:15.17jayteewow, I think it's been almost 48 hours since someone came in here with the old, "Help! I need to setup Asterisk with 2 phones and voicemail in 30 minutes or my boss will fire me. Help me now, dammit! And don't give me any shit!!!"
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02:37.53joakoAnyone want to call 786-522-1172 (or sip:17865221172@sip02.netjdn.com) and tell me what they think of the sounds?
02:38.15jayteesure, hang on. Cell phone ok?
02:38.35PanGoatHelp! I need to setup Asterisk with 2 phones and voicemail in 30  minutes or my boss will fire me. Help me now, dammit! And don't give me any shit!!!
02:38.42PanGoatplease2help
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02:38.45PanGoat;)
02:38.47jayteehehehee
02:39.34joakojaytee: I could care less where you call from
02:39.37jayteehttp://www.osburn.com/asterisk-sign.png
02:40.53drmessanojaytee: I think joako said he could care less
02:40.53drmessanoTell him to Eff OFF
02:41.42jayteejoako, I get a message saying the call is forwarded to a voicemail system and then what sounds like you saying your name.
02:42.08joakojaytee: Better or worse than Alison?
02:42.31beekjoako: Nothing could be better than Allison.    Good evening gang...
02:42.36jayteeabout the same as Allison but your voice saying your name sounds like it's coming from a basement stairwell
02:42.50joakojaytee: I was in a Ford....
02:42.51jayteeevening beek
02:43.19beekjaytee: Have a good New Year's Day?
02:43.34jayteejoako, aha!!! that makes perfect sense now. If you'd been in a Chevy it would have sounded like a large attic with screeching bats in the background
02:44.22jayteebeek, mine was ok. I didn't drink so no hangover, I slept in so I didn't feel tired so all in all it was another dull, boring evening and day
02:44.32drmessanobeek: Theres nothing GOOD about it.  Wanna FIGHT?
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02:45.28beekdrmessano: It'd be a great way to start the new year -- a good ol' IRC fight.   Wait, didn't we have one a few weeks ago?
02:46.29beekjaytee: I'm visiting my college roommate and his wife -- she's a great cook.
02:46.41jayteecool
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02:50.01beekHeading home Saturday.   Monday I get to start fighting the telco again.   They're regressing... today we had four failures.
02:50.26drmessanoROFL
02:50.31drmessanoOh god
02:50.37drmessanoI need to not speed read
02:50.52beekdrmessano: what did you mentally translate that too?
02:50.54drmessanoI read College Roomate's wife and .. not "cook".
02:51.07beekLOL
02:51.53beekdrmessano: At least we know where your mind is.
02:52.41drmessanoYes, no question
02:53.14eppigyhello
02:53.24*** join/#asterisk Jabess (n=ircap@c-98-219-81-172.hsd1.fl.comcast.net)
02:54.21JabessI got this message: [Jan 1 22:54:57] WARNING[1096]: pbx_spool.c:451 scan_thread: Unable to stat /var/spool/asterisk/outgoing
02:54.24beekdrmessano: I checked out that app_rpt URL that you told me about.   You said that you're deploying one soon.   Is this equipment that you mentioned acting as the repeater controller or is the Asterisk part just for linking repeaters?
02:54.46JabessI need information about that
02:55.00eppigyare you running asterisk as root?
02:55.09Jabessyep
02:55.12beekJabess: 1) the directory doesn't exist.   2) you don't have privs to read the existing directory.
02:55.15eppigydoes that directoy exist?
02:55.37Jabessno, does not exist... Can I creaty empty?
02:55.45Jabess*create
02:55.48eppigyLETS FIND OUT
02:55.59eppigy8[]
02:56.01JabessI need information about this
02:56.12eppigymkdir /var/spool/asterisk/outgoing
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02:56.37JabessOK, I did
02:56.42eppigyAUSOME
02:56.42Jabessthat's all?
02:56.46eppigybro
02:56.57JabessI don't need to create anything inside
02:57.01eppigybro
02:57.09eppigylet me tell you a story
02:57.18eppigyabout how i learned to use computers
02:57.22drmessanocranks up the beat box
02:57.23eppigyi just did it
02:57.28eppigyuntil i got it write
02:57.30jayteebout a man named Jed
02:57.36eppigyand my name was jed
02:57.45Jabess[Jan 1 22:58:50] ERROR[1096]: res_phoneprov.c:780 set_config: Unable to load config phoneprov.conf
02:57.45drmessanocranks up the DMC style beats
02:57.58drmessanoThen dont use phoneprov, bro
02:58.04drmessanoPIECE OUT
02:58.06eppigydoes tghe jackson hieghts
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02:58.44Jabessthat is a module?
02:58.54JabessHow Can I unload that?
02:58.56eppigyTRABAJO
02:59.19eppigyDONDE ESTA
03:00.09jaytee~book
03:00.10jbotfrom memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
03:00.23PanGoatnice
03:00.39jayteeI heard that's a really good read for anyone wanting to learn Asterisk that obviously doesn't have the first clue about it.
03:01.16eppigyi have heard opposing views
03:01.26eppigybut i mean
03:01.30eppigylets not mix words
03:02.11jayteeyou have a better book to suggest for noobs, "dave"?
03:02.47PanGoatIt was super useful for me (a * noob). The paper version is particularly handy so you can flip around while working on your stuff.
03:02.48eppigyi am merely making humor
03:02.59PanGoatalso, that video demonstration
03:02.59eppigyit is the same book i read
03:03.16jayteemy paper version is signed by one of the authors
03:03.32PanGoatthe video interview/demonstration was the first thing I was exposed to and really nice to see/hear people workign through * setup.
03:03.58drmessanoI heard that voip-wiki was real good
03:04.11drmessanoLots of good info for Asterisk 1.0 and 1.2 users
03:04.17jayteevoip-wiki is an awesome resource for anyone running Asterisk 1.2
03:04.38drmessanoand asteriskwin32
03:04.48jayteeasteriskwin32 rocks!!!!
03:05.10jayteespecially on Windows ME, runs way better than on Windows 98
03:06.53PanGoatlol
03:07.29PanGoatalthough, I do confess to heresy in that I run one instance of * on Mac OS X Server
03:07.34JabessI still have problem with the noise.. grrr
03:08.30jayteewell, that's not really heresy since it's just an Apple ripoff of the BSD kernel
03:08.44beek"Thank you for calling the XYZ company.  Please wait while I reboot your call..."
03:08.50eppigynothing will beat * running on a ubuntu hyper-v VM
03:08.53eppigyEVER
03:09.17jayteeexcept * running on RHEL 5.2 64 bit XEN
03:09.25eppigyBOOYA
03:10.03beekvirtualized the XEN hypervisor so now I need no hardware to run my systems.
03:10.04eppigythat is one of my goals this year
03:10.07eppigyto get my rhce
03:10.13eppigybeek: nice
03:12.32drmessanobeek: Reminds me of the guy I told to run asterisk on BIOS
03:12.55drmessanoNo OS overhead
03:13.01jayteethat was just a few days back :-)
03:13.02eppigytrue
03:13.42jayteeya just have to remember to put kernel=no in your asterisk.conf :-)
03:13.51drmessanoGoogle apps kinda rocks
03:14.10jayteeomg! asterisk as a google app :-)
03:14.57drmessanoI was thinking about adding Google apps to one of my domains just for the hosted XMPP
03:18.35eppigytrue
03:20.18beekAnyone here ever use the Queue system for non-local agents?   What I'm trying to do is bridge calls to our answering service.  They can only take 5 calls at once while I can receive 23.  While its trivial to do the Dial() to bridge the calls, I was looking for some way to keep the people "in line"
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03:25.13[TK]D-Fenderbeek: will the remote side answer and queue them themselves?
03:25.46beek[TK]D-Fender: Good evening.   Yes... they can queue up to five calls.  The sixth gets a busy.
03:25.59[TK]D-Fenderbeek: then this won't really work...
03:26.31jayteebeek, who hangs up the call?
03:26.46beekjaytee: could be either the caller or the callee.
03:26.55jayteedamn
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03:26.57beekRight now we use forwarding to bypass our PBX.
03:27.19beekI want to get in the middle of the call to offer the caller the option of putting themselves on our outage list.
03:28.11beekMy less elegant method just means that there's really no order but I can get by with it.
03:30.14jayteeif Asterisk offered a way to continue in the current context, extension if the caller hung up I could think of a way to "throttle" the number of calls but the Dial app only offers the ability to continue if the called party hangs up.
03:30.44drmessanoYou need a Q-Box.. 2 RJ-45s, a power cord, and a whole lotta luck
03:31.34beekjaytee: My current plan is simply use group_count to ensure that there are no more than five calls.  If there are more then the caller will hear that "all agents are busy... but if you just want to report your outage press 1" and I can take if from there.
03:32.08jayteebeek, that might work
03:32.11Kobazis there a feature that's like *0 (disconnect), but won't actually completely disconnect the call
03:32.18Kobazand just bail from the Dial() and continue on the dialplan
03:32.47jayteenot that I know of but it would be a nice feature
03:33.24jayteeprobably be better to use an AGI script
03:33.42Kobazcan i use the O option for dial? i wonder
03:33.47beekjaytee: the only possibility is that I'll have two attempts to a single available call.  If I check the DIALSTATUS variable I can see if it returns "BUSY" and act accordingly.
03:34.09Kobazjaytee: agi or no agi wouldn't make a difference, *0 will still hang up the channel completely
03:34.42jayteewell, then it must be all russellb's fault then
03:34.55eppigyTRABAJO
03:53.29drmessanoSABADO DOMINGO
03:57.37eppigytrue
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04:08.58KnightfalAnyone know how to use flite in the dialplan to read from a file?
04:09.55Knightfalthis is what Im using now to read data from a mysql var exten => 468,14,Flite(${name} this is a test.)
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04:18.48[TK]D-FenderKnightfal: Do its instructions say you can point it to a file?
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04:24.41KnightfalIt does not say that you can.
04:25.04KnightfalHowever, I thought there may be something else we could try.
04:25.06Knightfal:)
04:25.39[TK]D-FenderKnightfal: Use something else to read the file and feed it in chunks
04:26.06[TK]D-FenderKnightfal: Ain't Raw-Cat Science
04:27.17KnightfalIll probably just use mysql
04:27.26KnightfalAs always thx.
04:27.35KnightfalHappy New Year
04:41.07drmessanoSorry Ms Jackson
04:42.45drmessanoI am fo real
04:43.12drmessanoNever meant to make your PBX crash, may I offer you some cash
04:46.00drmessanoBombs Over Dialplan is all I got
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06:12.13drmessanooh yeah
06:12.14drmessanowell
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06:21.45[TK]D-Fender<PROTECTED>
06:21.57SkramXi agree
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06:47.44[TK]D-Fendercheckout time.  Later all
06:49.09JAMMAN2110Finally done: http://blog.jamesforman.co.nz/2009/01/controlling-applications-with-asterisk/
06:50.57PanGoatJAMMAN2110: Oh, this is cool man. waaay cool.
06:51.25JAMMAN2110Srsly?
06:52.17JAMMAN2110:)
06:52.44PanGoatI think so. That's actually why I got into * -- to expore the interactive possibiities
06:52.51PanGoatexplore*
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06:53.31JAMMAN2110Im a big fan of home automation and such
06:53.46JAMMAN2110It just seemed like a logical thing to do
06:54.00JAMMAN2110But I couldnt see any EASIER and faster way to do it
06:54.26PanGoatSure, I'm working on one to navigate QuickTime Virtual Reality movies
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06:54.45PanGoata phone has more buttons than an Apple Remote or mouse
06:55.42JAMMAN2110Cool
06:55.55JAMMAN2110My next plan is to have the volume drop to a certain level when someone calls in
06:56.01PanGoatnice
06:56.08JAMMAN2110No idea how I'll do that though
06:56.21PanGoatOh, with *, I'm sure there is a way
06:56.47JabessI need if some user call to any extension does not exist, I want that asterisk reproduce a MP3 file saying: The extension does not exist.! ??
06:56.48PanGoattime to crash, but yeah, nice writeup. keep posting stuff.
06:56.54Jabesswich conf file can be possible?
06:57.03PanGoatnight
06:57.30JAMMAN2110Jabess - Whats wrong with "Your call can not be completed as dialed, please check the number and try again" ?
06:57.42JAMMAN2110Night PanGoat
06:58.43Jabess[JAMMAN2110] well, I have using a softphone, but, that no say me anything if the extension does not exist.... I think that sound is the same softphone..
07:15.09Hanif08team...  I rx "[Jan  2 03:03:47] WARNING[25279]: app_voicemail.c:2861 leave_voicemail: No entry in voicemail config file for '1000'
07:15.16Hanif08someone can shed a light to me..
07:16.11drmessanoSounds like theres no entry in the voicemail config file for 1000
07:16.22JAMMAN2110I dare say... what drmessano beat me to saying
07:16.57Hanif08actually i try to follow exactly as aterisk tutorial..
07:17.16Hanif08just change the extension numbers onlly
07:17.24Hanif08but no luck..
07:17.26drmessanoYou left off the part about having an entry in the voicemail config file for 1000
07:21.12Hanif08[default]
07:21.13Hanif081000 => 1000,1000,hanif_umar@hotmail.com
07:23.20drmessanoand where are you sending the voicemail to in the dialplan?
07:23.44Hanif08yep..
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07:25.15drmessanoyep?
07:25.15JAMMAN2110Got to love getting your questions answered
07:25.23drmessanoWTF dude
07:25.33drmessanoAre you sending the call to 1000@default?
07:25.38drmessanoYou get one answer here
07:26.15Hanif08yes..
07:26.49Hanif08with this one  config at exten.conf
07:26.52Hanif08[tutorial]
07:26.53Hanif08exten => 1000,1,Dial(SIP/1000)
07:26.53drmessanoYou're leaving off something here
07:26.54Hanif08exten => 1000,2,VoiceMail(1000@default)
07:26.55Hanif08exten => 1000,3,Playback(vm-goodbye)
07:26.56Hanif08exten => 1000,4,Wait(2)
07:26.57Hanif08exten => 1000,5,HangUp()
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07:28.50Hanif08drmessano> i can't sending the call to 1000@default ?
07:29.33carrarthats a awesome dialplan
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07:31.39carrarlack of a context in voicemail.conf *IS* @default
07:31.48carrarwell
07:32.12carrar[default] that is
07:32.22carrar*freenode-connect* file serving is offtopic on freenode---please see http://irc.netsplit.de/networks/ for a list of
07:32.22carrar<PROTECTED>
07:32.24carrarwhats that
07:32.26carrarcrap
07:32.38carrarfreenode is broke
07:34.01carrarHanif08, try exten => 1000,2,VoiceMail(1000,u)
07:34.53carraryou would use '@default' in sip.conf
07:36.41Hanif08carrar> ok let my try..
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08:25.21WobbleHi, in the cli is it possible to cut someone off or to transfer them elsewhere? If so, is there documentation for this anywhere?
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08:28.53jqlgoogle asterisk ami redirect
08:29.03Wobblethanks jql :)
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09:23.48tessierHello all!
09:24.46tessierAges ago I used to have my asterisk console set up so that it would show the extensions on the console as calls traversed through them. I thought that happened at some specific debug level. But now I have core debug set to 30 and I don't get that behavior anymore. Anyone know how to enable that? Asterisk 1.4.22
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10:21.40zeeeshanybody there
10:22.19zeeeshtrying to compile asterisk-1.4.22, getting error , will anybody guide "http://paste.lisp.org/display/72900"?
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10:34.23DrukenLPYmorning everyone....
10:34.38DrukenLPYdoes anyone know if i can use agi to access mssql ?
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11:42.39zeeeshwhile compling getting this error "/usr/bin/ld: cannot find -lcap"?
11:44.21Madkisscan I make asterisk accept more than one call at the same time for one extension?
11:44.48DrukenLPYcourse
11:44.53Madkisshow?
11:45.00Nuggetthat's how asterisk works.
11:45.08Madkissew, wait. i have a brain error.
11:45.08Nuggetyou would have had to do something weird for it to work otherwise
11:45.31DrukenLPYltns Nugget
11:45.55Nuggetmoos
11:46.07DrukenLPY:)
12:07.48seanbrightzeeesh: debian?
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12:27.16phpboyDoes anybody know where I can get a template of the little number paper on a snom extension?
12:43.35zeeesh<seanbright>: centos 5.2
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12:56.20Dovidanyone know where i can get a list and or comaprison of 1.4 to 1.6. wut new features are 1.6 etc. ?
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13:01.10zeeesh<Dovid>: http://downloads.digium.com/pub/
13:01.22seanbrightzeeesh: yum install libcap?
13:03.28seanbrightwanders off again
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13:08.15zeeesh<seanbright>: already installed libcap-1.10-26 and libcap-devel-1.10-26
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13:47.11twistedwhat am I doing up
13:50.42coppiceits probably the viagra
13:52.50jjshoezeeesh try running ldconfig
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13:56.10DrukenLPYtwisted: could be worse, i've been up since 4:00.... couldn't sleep
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13:58.28eppigyzeeesh: what distro?
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14:05.31skirmishaanyone here alive
14:05.50eppigyi am dead inside
14:06.10skirmishai will paste from ast-dev channel
14:06.21skirmisha<skirmisha> i have little problem with asterisk 1.6
14:06.22skirmisha<skirmisha> it does not accept # sign in regster option in sip config
14:06.22skirmisha<skirmisha> in 1.4 it is working fine
14:06.22skirmisha<skirmisha> what could be the problem?
14:06.22zeeesh<eppigy>: centos 5.2
14:06.28skirmisha<skirmisha> like user:pass#@server.com
14:06.29skirmisha<skirmisha> sip show registry does not show anything
14:06.29skirmisha<skirmisha> if i remove # at the end it works
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14:07.21eppigyzeeesh: try editing /etc/ld.so.conf
14:07.38skirmishaany ideas
14:10.17zeeesh<eppigy>: while compling getting this error, will u pls take a look "http://paste.lisp.org/display/72900"?
14:12.21jayteezeeesh, did you try what I suggested yesterday when you asked and pasted the same problem?
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14:14.43skirmishai found something strange
14:14.51skirmishaif i add that # at the end
14:14.52jayteedrums fingers on desk.....waiting
14:15.00skirmishathen i see that user in sip show peer
14:15.14skirmishameans remote peer is registered in local server
14:15.17eppigyTRABAJO
14:15.18skirmishawith that user
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14:41.03Zeeeks,1,NoOp(Hello asterisk world)
14:41.47jjshoes,n,HangUp()
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14:53.59Zeeekt,1,Playback(the-number-you-have-reached)
14:55.38Zeeek[TK]D-Fender Happy 2009
14:58.22[TK]D-FenderZeeek: not yet it isn't..... Stupid bug has had the better of me for almost a week now.
14:58.40delcoyotebasic question: is cable-modem net supported with voip , asterisk?
14:58.59[TK]D-Fenderdelcoyote: huh?
15:00.04delcoyoteI mean does asterisk run under ptb boxes telephone line, broadband? but is cable broadband supported?
15:00.13delcoyotebut is it*
15:00.38dlewisdelcoyote: I use Optimum Voice with asterisk with an ATA...
15:00.50dlewis(which is cable broadband "voip")
15:01.09dlewisthe modem converts the signal to POTS
15:01.19dlewisso, you have an RJ-11 coming out of the cable modem
15:01.27dlewisyou have to get an ata to convert it
15:01.39[TK]D-Fenderdelcoyote: Cable bradband?
15:01.45dlewisa linksys SPA, a grandstream ht or a sangoma analog card will work.
15:01.49[TK]D-Fenderdelcoyote: as in INTERNET?
15:02.02delcoyoteyes [TK]D-Fender
15:02.23[TK]D-Fenderdelcoyote: Yes * speaks several VoIP protocols and YES over the internet hence its name
15:02.54[TK]D-Fenderdelcoyote: Internet is internet.  Cable DSL, T1, etc does not matter... packets are packets
15:03.16delcoyoteand some ISP have voip, andsometimes could happen that is not available?
15:03.41dlewisdelcoyote: do you have phone service from the cable company?
15:03.43[TK]D-Fenderdelcoyote: Your ISP could be assholes and block other providers, yes
15:03.45Zeeek[TK]D-Fender Oddly enough, I've been horribly sick for a week myself
15:03.59delcoyotevoip not available n my area
15:04.11dlewisdelcoyote: VOIP is not area specific
15:04.21delcoyotebut im entightled 2 lines
15:04.22dlewisdelcoyote: if you have internet, you can use mostly any VOIP provider.
15:04.54[TK]D-Fenderdelcoyote: what tells you hat it will not work?
15:05.28delcoyotenothing [TK]D-Fender
15:05.41[TK]D-Fenderdelcoyote: Then you are worrying about nothing <-
15:05.50[TK]D-Fenderdelcoyote: Go and TRY
15:06.01delcoyotemodem has 2 sockets for phone
15:06.03delcoyoteI wil
15:06.07[TK]D-Fenderdelcoyote: FORGET THE MODEM
15:06.12delcoyoterouter?
15:06.28delcoyotedd-wrt is better to install it ?
15:06.43[TK]D-Fenderdelcoyote: Just install a soft-phone on your PC and try to connect to Ekiga.net
15:08.39delcoyoteok thanks for the info
15:11.06gambler1hi, is it possible to turnoff crc4 in dahdi?
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15:22.12ZeeekOur #voip-users-conference is warming up the IRC channel at the moment
15:23.04ZeeekLater on I will post a sure-fire working SIP URI to listen live. Should be pretty cool today
15:23.41coppicewell, it is winter :-\
15:23.58[TK]D-Fenderdelcoyote: in some places...
15:24.07[TK]D-Fenderdarn leftover autocompletes
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15:26.22path_is there a way to redirect automatically one extension to another (e.g. if the first destination is busy)
15:27.15path_mi doubt is: that is accomplish through the asterisk or through the softphone
15:27.20Zeeekpath_ it's good to use a macro for that
15:27.48Zeeekthe asterisk dialplan knows when an extension is busy, so you can do it there
15:28.10path_great, I'll look it up Zeeek :)
15:28.23Zeeekyou knew that was coming :)
15:28.44ZeeekLook for a macro called "one-line" or one line phone or such, there a many versions out there
15:28.58Zeeekthat way, you use the same code for each extension
15:29.48path_thanks! I appreciate your help Zeeek :-)
15:30.12ZeeekI hope you find what you seek
15:30.22Zeeeks/seek/seeek/
15:30.58path_me too, tough I'm reading some books so it should be covered
15:31.29path_~books
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15:34.36[TK]D-Fenderpath_: Dial them back to back.  And don't call "devices" as "extensions"
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15:35.58Zeeeksez: reading books is goood
15:36.15path_indeed
15:36.58Zeeeksez: Reading Penthouse is better than reading books
15:37.29Zeeek"Obliqueness is the curse of the reading class", Stephen King
15:37.52coppiceisn't penthouse a book? :-\
15:38.24[TK]D-Fendercoppice: semantically a "magazine"
15:39.28path_hahah
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15:40.22PanGoat*someone* reads those articles in the back
15:41.27ZeeekIn spite of all the jokes about that, Playboy had some of the best articles any magazine has ever published. The great SciFi authors and futurists, for example. Of course I have no direct experience with this... I just heard
15:41.44ZeeekOMNI kind of came out of that world
15:42.17ZeeekRoald Dahl, those were great stories. Many dramatized on Alfred Hitchcock Presents
15:42.28Zeeekis showing his advance age
15:42.50Zeeekoops, wrong channel for philosophy
15:43.15Zeeekor culture
15:43.57Zeeekwith regard to asterisk, we do have a conference today, beginning in a little more than one hour
15:43.57coppiceon Roald Dahl's Tales of the Unexpected too :-)
15:44.42Zeeekcoppice: amazing, brilliant stuff. J-He also wrote "My Uncle Oswald" which I think were stories that originated in Playboy
15:45.02path_Zeeek, conference through IRC ? nice
15:45.51Zeeekno live SIP conference with IRC "back channel" for catty remarks and trying to break John Todd up while he's talking
15:46.24ZeeekIRC #voip-users-conference
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15:47.10ZeeekSIP URI 7463#22622#PIN@proxy.ideaSIP.com
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15:47.55path_thats handy :D
15:48.53ZeeekYou could also just call PSTN: (724) 444-7444 enter DTMF 22622# and then 1#
15:52.18PanGoatZeeek: Agreed, I was actually referencing the Lenny Bruce bit about that - which, in a way, shows my age :)
15:52.40dlewisZeeek: anything visual for this conference?
15:53.49Zeeekdlewis: I was considering a bathrobe video cam but decided against it
15:55.57ZeeekLenny Bruce was a giant, there aren't that many good social critics these days
15:56.15Zeeekoops wrong channel again
16:02.25ZeeekIf you are on Twitter by the way, enter your Twitter name here : http://tr.im/voipform  to appear in a asterisk/voip/telephony directory
16:02.43ZeeekYou can look at the directory here: http://tr.im/voipview
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16:19.09KattyZeeek: !
16:19.11KattyZEEEEEEK!
16:19.13Kattyhugs Zeeek
16:19.23KattyZeeek: where /have/ you been?!
16:19.38ZeeekOMG it's..... {{{Kaaaaty}}}
16:20.10*** join/#asterisk bahjons (n=robert@140.99.23.26)
16:20.48ZeeekKatty: I have been everyone, in all the usual places, here, there, Twitter, the conferences
16:21.08Kattyguess you're just not talkin when i'm around then )=
16:21.18Kattysnob.
16:21.36ZeeekWhispering in your ear, you just don't have the right CODEC
16:22.14Zeeekseeing anthm reminds me I *must* update the AA50 firmware soon, real soon
16:22.19coppiceCELT is the right codec :-)
16:24.01*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
16:28.07bahjonsHi All, I'm new around here. Been a Asterisk 'user' for awhile. But never been actively involved in the community.  I've got a quick question, just need a simple answer.
16:28.07bahjonsQ: Can you set the periodic-announce via the dialplan?  eg: exten => s,n,Set(PERIODIC_ANNOUNCE=/home/Sounds/queue-announce)
16:28.08ZeeekKatty Opening voice in about 10 minutes if you wanna say "hi"
16:29.07Linuturkwoot, I just registered my first softphone to asterisk!
16:30.47ZeeekChampagne for all
16:31.17*** join/#asterisk thotypous (n=asdf@unaffiliated/thotypous)
16:31.43[TK]D-Fenderbahjons: No
16:32.08[TK]D-Fenderbahjons: Are you looking to change the queue overall from that point, or just for that singular caller?
16:32.21bahjonsjust for that caller
16:35.33[TK]D-Fenderbahjons: How many exceptions do you need to handle?
16:38.46*** join/#asterisk sasargen (n=chatzill@174.152.243.255)
16:38.49bahjons[TK]D-Fender: Basically, based on the incoming DID, I want to have a different periodic_announce, periodic_announce_frequency, and servicelevel
16:39.46*** join/#asterisk tobias (n=tobias@cpe-076-182-095-118.nc.res.rr.com)
16:40.00twistedugh ugh ugh
16:40.58[TK]D-Fenderbahjons: Wrong answer, try again
16:41.37path_Zeeek, I've read about macros and queues so far but I'm looking for 'cascading queues'. Is that accomplish through macros too ?
16:41.46[TK]D-Fenderbahjons: And now that you brought those other factors in, the rest can't be done in a single queue
16:42.30[TK]D-Fenderpath_: Macros play no role in this.  People talk like thy are meant for everything.
16:43.00[TK]D-Fendertwisted: Glorious day for you too?
16:43.15path_[TK]D-Fender, so who plays the role?
16:43.31path_apart from Queue() and Agents
16:43.39[TK]D-Fenderpath_: Clarify your intent about "cascading queues"
16:44.46path_[TK]D-Fender, if user1 is busy so the call goes to user2, if user2 is busy goes to user1, user 1 and 2 busy, to user3
16:45.12[TK]D-Fenderpath_: That is just a SINGLE queue.  Not "cascading"
16:45.40[TK]D-Fenderpath_: So 1 queue it is
16:46.01path_uhm now its seems more clear
16:46.36path_thanks [TK]D-Fender
16:46.51*** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-172-rndf-esr-5.dynamic.isadsl.co.za)
16:47.14[TK]D-Fenderpath_: A queue processes people in line based on a strategy.  The one you described is "roundrobin" (goes in circle)
16:47.51path_yes, I was reading about 'strategies'.. roundrobin, ringall, and so on
16:48.33path_so I don't need to add an agent
16:50.06*** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net)
16:50.36*** part/#asterisk pikachu2000 (n=pikachu2@196-209-182-172-rndf-esr-5.dynamic.isadsl.co.za)
16:51.14[TK]D-Fenderpath_: Queues have "members".  'Agents" are a special channel type that involves chan_local.  You can directly add SIP devices, etc as well.  You can also use Dialplan apps to dynamically add or remove members.
16:51.58[TK]D-Fenderpath_: member => SIP/100 <- this device is always a member of the queue. "core show applications like queue"
16:52.53path_I see
16:52.59path_gotta read a lot :)
16:53.09[TK]D-Fenderpath_: that will list the dialplan ways of letting you dynamically add devices as members.
16:54.00jayteewait! macros aren't meant for everything? oh shit!
16:54.19SkramXheh
16:54.41path_[TK]D-Fender, thanks again now I've got a start point
16:54.41bahjonswhat if I just wanted to set the periodic_announce via the dialplan?
16:55.03[TK]D-Fenderbahjons: Answer my previos question.
16:55.19bahjonswhat do you mean by that ?
16:55.40[TK]D-Fender[11:35]<[TK]D-Fender>bahjons: How many exceptions do you need to handle?
16:55.50[TK]D-Fenderbahjons: What is not clear in that question?
16:56.31bahjonsI don't know what you mean by exceptions
16:56.42bahjonsexpections meaning omissions?
16:56.57[TK]D-Fenderbahjons: Queue has 1 announcement.  how many ALTERNATIVES do you require?
16:57.17Zeeekwe're on the air in a minute. Call us: 7463#22622#1@proxy.ideasip.com
16:57.22Zeeekbye for now
16:57.25*** part/#asterisk Zeeek (n=Zeeek@bdx.resmo.net)
16:57.28SkramXlol
16:57.56*** part/#asterisk thotypous (n=asdf@unaffiliated/thotypous)
16:58.15bahjonsI'm dealing with 1K+ extensions/DIDs. Each one will have it's own announcement.
16:58.42bahjonsI want to avoid having 1K+ queues. It's simply not manageable.
16:58.51[TK]D-Fenderbahjons: 1000?  Yup, No chance at all.
16:59.25bahjonswhat's the least I could have.
16:59.28bahjons?
16:59.49[TK]D-Fenderbahjons: if you were in the 5-10 range I'd have had a suggestion, but you are WAY out.
17:00.13SkramXsounds like you should edit the asterisk code. yay for open source ;)
17:00.25[TK]D-Fenderbahjons: I'm sure its wouldn't be too hard to write a patch for this.  Go hire someone for it
17:00.39bahjonsYes, I may need to
17:00.48bahjonsthank you
17:00.58SkramX[TK]D-Fender - what were you going to suggest for 5-10?
17:02.07[TK]D-FenderSkramX: My "cheat" would have been to have you change the language to one you don't have anything in and thus your queue recording could be split amongst those folders and all other souds would fall to the "default" language.  Sure, things would look a little funky, but it'd work.
17:02.22SkramXah... good idea ;)
17:02.22[TK]D-FenderSkramX: Multi-Lingual fun...
17:02.27SkramXright
17:02.52*** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk)
17:03.11*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
17:03.44bahjonsJust need another 100+ languages ;-)
17:04.02[TK]D-Fenderbahjons: No, from your count, 9900
17:04.28SkramXwait
17:05.01SkramXbahjons- you want 1000 custom announcements? how many agents? 1000?
17:05.01bahjons1K+ was our target. Starting out would be alot less (50+/-)
17:05.02SkramXbasically im asking how many queues and how many agents?
17:05.14SkramXand just one queue?
17:06.00bahjons1 queue, 100+ agents
17:06.53SkramXk
17:09.52[TK]D-Fenderbahjons:   Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,gosub][,rule]) <---- HMMMMMMMMMMMMMMM
17:10.41[TK]D-Fender<PROTECTED>
17:10.43[TK]D-Fender<PROTECTED>
17:10.44[TK]D-Fender<PROTECTED>
17:10.46[TK]D-Fender<PROTECTED>
17:10.51[TK]D-FenderCuriose bit of code that is....
17:11.02[TK]D-FenderLooks like its right there...
17:11.19[TK]D-Fenderbahjons: So, new answer : Looks like "yes"
17:12.06[TK]D-Fenderso far..
17:12.22bahjonshmmm, not sure how I could've missed that.
17:12.33bahjonsCan't set the frequency though.
17:12.39bahjonsThis will do for now though.
17:13.13[TK]D-Fenderbahjons: Not sure this wil do it.
17:13.32*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
17:13.38[TK]D-Fenderbahjons: Doesn't appear to be for the periodic... rather the FIRST message they get, not the repeated one...
17:13.40*** join/#asterisk jtodd (n=jtodd@c-76-124-123-117.hsd1.pa.comcast.net)
17:13.48[TK]D-Fenderbahjons: Which means... find a coder :)
17:14.31bahjonsok, will do
17:14.40bahjonsbounty perhaps :-)
17:14.42[TK]D-FenderClosest miss yet!
17:15.05[TK]D-Fenderbahjons: that would do.
17:16.29[TK]D-Fenderbahjons: If you were to put a bounty out I would make it so that ALL queue config values be mapped to Channel vars if not null.
17:16.49[TK]D-Fenderbahjons: ThatbahWhere feasible anyways
17:17.33bahjonsyea, that's what I would do.
17:19.55SkramXOT: haha. My Sprint (EVDO) bill says I used 4,236,423 minutes. I think they mean kilobytes ;)
17:21.54*** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman)
17:23.49verywisemani will use linksys 3102 with * , is it necessary to install zaptel packge ?
17:24.08[TK]D-Fenderverywiseman: no
17:25.26*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
17:25.26*** mode/#asterisk [+o russellb] by ChanServ
17:25.40*** join/#asterisk BCS-Satori (n=somewher@75.148.21.113)
17:25.53BCS-SatoriCan Asterisk act as an H.323 Gatekeeper?
17:26.41jjshoeSkramX on verizon evdo I can't begin to tell you the number of times I've had to call to get my service restored for going over the data limit
17:27.13SkramXi think my theoretical data limit is 5gb. I just used ~4 gigabyte
17:27.17*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
17:27.29SkramXI dont think they guarantee unlimited, do thet?
17:27.31SkramX*they?
17:27.54[TK]D-FenderBCS-Satori: no
17:29.30jjshoewho knows, I don't read the legal mumbo jumbo, I just make noise until they reset the cap :P
17:29.48SkramXhaha
17:30.03SkramXI only use it when I travel/somewhere I don't trust/like the WiFi service
17:33.28jjshoeI used evdo as my primary internet for like 8 months
17:37.21SkramXah, well then
17:38.22*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
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17:59.25*** join/#asterisk Get_The_Fish (n=IceChat7@75.151.94.189)
17:59.34Get_The_Fishhello everyone... I am running into an issue that I cant seem to find an answer on Google for.  When in the comedian mail voice prompt system, pressing 3 for advanced options just gives me to the option to press * for the main menu
17:59.40Get_The_Fishthis happens whether I use *97 or *98.  I see no errors or warnings in the log files, and the asterisk console debugger isnt showing me anything that looks like an issue with asterisk.
18:00.33[TK]D-FenderGet_The_Fish: pastebin the complete failed attempt
18:00.37[TK]D-Fender~pb
18:00.38jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/
18:01.26[TK]D-FenderGet_The_Fish: CLI output, veerbose 10, debug 10.
18:02.40coppicedon't you need to turn up verbose to 11? :-\
18:03.08[TK]D-Fendercoppice: This is... Spinal Tap!
18:03.24Get_The_Fishok
18:07.02Get_The_Fishthis system is in use, so there is a lot of other action in here, kinda hard to find what I am looking for
18:08.47*** join/#asterisk astassistant (n=support@h-67-101-7-84.sttnwaho.dynamic.covad.net)
18:10.59*** join/#asterisk macli (n=macli@nmc.brc.ubc.ca)
18:11.17Get_The_Fishaight, http://pastebin.com/d5e7b4bbd
18:11.27Get_The_FishI think that is everything related only to this call
18:11.34Get_The_Fishit's not very helpful imho
18:12.01Get_The_Fishasterisk 1.4.22 with FreePBX 2.5.2
18:14.16[TK]D-FenderGet_The_Fish: I don't see you listening to your message in there..
18:16.43*** join/#asterisk asteriskmonkey (n=philip@69.77.169.14)
18:17.08Get_The_FishI wasnt...the issue that I am having is that option 3 (advanced options) in comedian mail just takes me back to the main menu... I cannot record greetings, names, etc.  I should tell you that this was working before.
18:17.35asteriskmonkeycheck your directory permissions
18:17.50[TK]D-FenderGet_The_Fish: That isn't where you SET those things
18:18.03[TK]D-FenderGet_The_Fish: You're looking in the completely wrong place.
18:18.16[TK]D-FenderGet_The_Fish: "0" for MAILBOX OPTIONS
18:18.39Get_The_FishTK, oh shit.
18:18.46Get_The_Fishyeah no kidding huh
18:19.19Get_The_Fishwell, since I already look like an idiot, cant do much more damage- what is the advanced options menu for then?
18:19.35[TK]D-FenderGet_The_Fish: Listen to a MESSAGE, press it and SEE
18:20.43Get_The_Fishyeah ok, that makes sense.  USER ERROR!!! hahaha thanks TK for your help and patience.
18:20.58mchouumm, is tollfreegateway having issues?  every toll free number I dial through there comes back with 'Got SIP response 484 "Address Incomplete" back from 204.8.45.222'
18:22.39Kattyzeeek's gone :<
18:22.39[TK]D-Fendermchou: usually means you aren't dialing a complete number per their spec
18:23.07mchou[TK]D-Fender: weird.  used to work before
18:23.37[TK]D-Fendermchou: watch your debug, try different means
18:24.38*** part/#asterisk bahjons (n=robert@140.99.23.26)
18:25.43mchou[TK]D-Fender: To: <sip:18004664411@sip.tollfreegateway.com>;tag=vS25HgXB2pttQ
18:26.34[TK]D-Fendermchou: try without the 1
18:26.37mchou[TK]D-Fender: header says 'Reason: Q.850;cause=28;text="INVALID_NUMBER_FORMAT"'
18:27.59*** join/#asterisk DelphiWorld (n=Miranda@41.221.17.21)
18:28.04DelphiWorldhi my friends
18:28.11DelphiWorldplease any IAX softphone for windows ?
18:28.38[TK]D-Fender~zoiper
18:28.39jbot[~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com
18:30.04DelphiWorldok
18:32.23mchou[TK]D-Fender: same deal even w/o the 1.  dunno what's going on now
18:32.41*** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk)
18:34.40SuPrSluGwhat's a good stress test tool?
18:35.20Juggiea girlfriend
18:42.42SuPrSluGhehe for sip?
18:43.17eppigyhttp://sipp.sourceforge.net/
18:45.52SuPrSluGeppigy,thx
18:47.56eppigynp
18:49.05DrukenLPYJuggie: try a wife... much more stress
18:49.54*** join/#asterisk `Sean (i=Un1x@CPE001dd042bb1e-CM0014045acc3c.cpe.net.cable.rogers.com)
18:53.59*** join/#asterisk thotypous (n=asdf@unaffiliated/thotypous)
18:54.35thotypoushi
18:55.59*** join/#asterisk MrNaz (n=mrnaz@ppp118-208-250-214.lns10.mel6.internode.on.net)
18:57.55*** join/#asterisk MrNaz (n=mrnaz@ppp118-208-250-214.lns10.mel6.internode.on.net)
18:58.12*** join/#asterisk john_fbac (n=FreePBX4@216.186.221.211)
18:58.30DrukenLPYanyone got a simple agi that can do a lookup on mssql ?
18:58.51john_fbacanyone have suggestions for a way to toggle day/night?
18:59.17DrukenLPYin which manner?
18:59.44[TK]D-Fenderjohn_fbac: Wait 12 hours
18:59.57carrarhahah
19:00.05john_fbacnice
19:00.08[TK]D-FenderDrukenLPY: Pick a language, learn how it can interface with mssql.
19:00.44DrukenLPY[TK]D-Fender: i was thinking php... but i dunno.. can i run a linux php and still interface with mssql?
19:01.12[TK]D-FenderDrukenLPY: PHP does whatever it is capable of.
19:01.21[TK]D-FenderDrukenLPY: so WTF is a "linux PHP"?
19:01.55carrarTry a Linux Perl!
19:02.01DrukenLPYhehehe well, kinda hard to run php on winblows and asterisk at the same time...
19:02.19DrukenLPYyou KNEW what i ment...
19:02.28[TK]D-FenderDrukenLPY: No, and even less now.
19:02.39[TK]D-FenderDrukenLPY: So you're running PHP & Asterisk on Windows now?
19:03.23DrukenLPYthe opposite actually... but can the php on my linux system access mssql, i know php on a WINDOWS machine can... but i dunno about linux
19:03.56[TK]D-FenderDrukenLPY: Same codebase.
19:04.07carrarDrukenLPY, PHP has to be compiled to work with whatever db you want to use
19:04.18[TK]D-FenderDrukenLPY: www.php.net
19:04.31carrarso if you have compiled php to work with mssql then it should access it ok
19:05.15carrar--with-mssql
19:05.20[TK]D-Fendercarrar: darn and I was hoping john_fbac would have gotten around to something practical we could have helped him with...
19:05.26carrarhaha
19:06.45[TK]D-Fendercarrar: and no need to show compile options he may already have or be able to load dynamically depending on packaging where applicable
19:06.53carraryeah
19:07.00carrarlots of unknowns there
19:07.08[TK]D-FenderDrukenLPY: So far you only baselessly doubt that you can.  Go read, Go try.
19:08.06carrarphp -m | grep -i mssql
19:08.17carrarif mssql shows up then it's compiled with it
19:08.42*** join/#asterisk voxter (n=voxter@190.241.15.218)
19:09.38DrukenLPYmssql
19:09.41DrukenLPYsweet
19:10.19carrarNow go read: http://us.php.net/mssql
19:10.22DrukenLPYi guess all that is needed is to make the php file for access, and possibly the odbc
19:10.52carrarshouldn't need odbc if you are accessing mssql directly from php
19:11.27carrarsince you have mssql support compiled into php
19:11.38DrukenLPYhmm... good point...
19:11.44carrargo try it
19:11.46DrukenLPYis stupid at times...
19:11.50DrukenLPYlack of sleep
19:25.08*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
19:41.43*** join/#asterisk Vyrus001 (n=Vyrus001@cpe-76-175-69-74.socal.res.rr.com)
19:42.23Vyrus001so does SendDTMF send audiable dtmf tones over a a channel? im trying to use it to do so and i hear nothing?
19:42.47eppigytrippin
19:44.59srdbe neat to be able to integrate asterisk with exchange
19:48.15DrukenLPYasterisk with exchange ?
19:48.27srdmmhmm
19:49.15DrukenLPYuse tts to read your emails?
19:49.25*** join/#asterisk telnettech (n=telnette@d192-24-95-65.col.wideopenwest.com)
19:49.28srdtts?
19:49.40DrukenLPYtext to speach
19:50.21srdguess that wouldn't be a bad solution, but I've never gotte into email being read to me
19:51.30srdmaybe access to voicemail through outlook
19:51.44jjshoeVyrus001 yes.
19:51.45DrukenLPYthats built in damnit
19:51.45srdintegrate the dialplan/directory into exchange somehow
19:52.21jjshoesrd I hit the magic button on my phone and say "import contants" and it imports all my contacts into my phone from outlook.
19:52.29jjshoesorry, "import contacts"
19:52.32srdyou add someone to the AD, it would add their phone/extension too
19:52.37jjshoeI can then say "call srd mobile"
19:52.45telnettechcan someone tell me where i can find specific settings for zaptel and zapata config files for a ISDN30. The pbx is in Aruba
19:53.18srdI was thinking tho of a project I wanna work on (different note)
19:53.34jjshoesrd what project is that?
19:53.40srdintegrating the pstn network into asterisk on a large scale
19:54.10DrukenLPYit's been overdone :)
19:54.14DrukenLPYhehehe
19:54.18srdwell
19:54.29srdI'm talking about on a massive scale
19:54.38srdwhere, instead of offering a voip solution through the internet
19:54.48srdget into the wholesale pstn market
19:54.49DrukenLPYdefine massive
19:55.00srdand pipe the landline into voip
19:55.05jjshoesrd you mean use asterisk simply for routing?
19:55.14jjshoesrd that's been done too.
19:55.17srdtranslation
19:55.33srdyou dial a number on your landline, depending on the route it needs to take
19:55.34DrukenLPYles.net has become a clec using asterisk....
19:55.46srdlike say it's another geographical number
19:56.03srdit will get sent to asterisk then asterisk will connect to that number through sip and then back to pstn
19:56.18srdwhere numbers like 999/112 would be handled by pstn alone
19:56.25jjshoesrd yes, that's called any itsp.
19:56.35srditsp?
19:56.44srdah ok
19:58.01DrukenLPYi'm confused
19:58.01srdtho this wouldn't be going thru the users broadband connection
19:58.01jjshoeit doesn't need to be to be an itsp.
19:58.14srdheh ok
19:58.25jjshoein fact there's even a module in asterisk that let's you use other peoples asterisk boxes to make outbound local calls on the cheap
19:58.29srdbut I'm wondering to myself tho
19:58.31jjshoei don't know what it's called though
19:58.57srdhow to setup a network like that that could handle say a million concurrent calls at any given time
19:58.57jjshoeI know you have to earn minutes yourself, or something silly like that, I can't recall off the top of my head
19:58.59DrukenLPYdundi
19:59.06jjshoeDrukenLPY yeah there you go
19:59.33DrukenLPYnot bad for memory... been years...
19:59.53srdonly hurdle I'd see at the moment would be fax/modem
20:00.06DrukenLPYa million concurrent calls...
20:00.16srdmmhmm
20:00.32DrukenLPYyou the leader of a small country that we don't know about?
20:00.36srdlet's say I had 5 million subscriptions
20:00.49srdno :)
20:01.05srdor rather subscribers not subscriptions
20:01.34srdwould have to take into account how many of those subscribers will be talking on their phone at any given time
20:01.55jjshoesrd it's been done.
20:01.55DrukenLPYsounds to me like your thinking about going a zaptel switching route, with possibly sip between regional servers
20:02.15Linuturkif my card has onboard hardware echo cancel, how do I activate it? turn on echo cancel in /etc/asterisk/zaptata?
20:02.36srdwell I'd get a number of optical carrier lines and down them to E3/E1 to each server
20:02.58srdbuy up wholesale voip from various providers (e.g. Level3)
20:03.21srdand route the calls through the various sip/iax trunks
20:04.33srdsetup one or two redundant data centres to accomodate any outage to ensure 100% uptime
20:04.57DrukenLPYno such redundancy with analog
20:06.48srdwell what I mean is, if your call gets dropped you could always redial the number and continue talking past that point
20:07.17jjshoesrd again, still, look up any major itsp.
20:07.24jjshoe~itsp
20:07.25jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
20:08.15srdok, the names irrelevant tho :)
20:08.20DrukenLPYjjshoe: he's talking about trying to offer analong lines, like the real telephone company does, not over the internet and adaptors
20:08.40srdright so say
20:08.52srdyour power goes out or you have an outage with your broadband
20:09.00srdin the case with a power outage
20:09.06srdas long as you have a corded phone
20:09.10srdyou'd still get a dialtone
20:09.22eppigydo you haver the capital to do this?
20:09.57srdnot yet, will be working on that next
20:09.59DrukenLPYi wish i did
20:10.20DrukenLPYa few billion dollars in the bank would be nice
20:10.38srdhehe
20:10.43DrukenLPYbut, if i had that kinda money, i wouldn't be using asterisk....
20:10.49eppigywell wishing for it is one thing
20:11.25srdI would for 2 reasons
20:11.42srddigium offers licencing for in house development
20:11.50srdand the second being in house development
20:12.09DrukenLPYpoint ?
20:12.11srdto steer away from the GPL
20:13.05DrukenLPYi'd rather go with something that was hot swappable and 1 hour onsite replacement :P
20:13.58srdI don't see why that goal couldn't be attainable thru further development of asterisk and other components
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20:15.51jjshoeDrukenLPY oh, sorry, I didn't see the part where he wanted to OWN the lines
20:15.54jjshoegood luck with that.
20:16.12srdown the lines?
20:17.22srddidn't say anything about owning the copper, was talking about doing the pstn to the customer (last mile) part through the existing telco network
20:17.45jjshoeand how do you think you're going to convince the telco to do that?
20:17.46jjshoelol
20:18.06srdwholesale
20:18.33jjshoeI'll point you to where DrukenLPY says "billion"
20:18.47srdno
20:19.44srdthe line rental through them will be um 25.17 per quarter
20:20.05srdfor one line
20:20.47jjshoewill be um?
20:20.56srd?
20:24.35srdhttp://tinyurl.com/99cosu
20:29.07jjshoeQwell do you know anyone who has a copy of the burning man article do you?
20:29.35srdburning man article?
20:29.51Vyrus001hm... so is there a reason SendDTMF isnt sending tones?
20:30.56*** join/#asterisk UQlev (n=kvirc@91.184.220.73)
20:31.00jayteeVyrus001, how do you know it's not?
20:31.10Vyrus001jaytee: lol well i dont hear anything
20:31.32jayteeVyrus001, well, it works fine here. Go fix it!
20:31.59Vyrus001lol
20:32.30Vyrus001just wondered if there was some reason why i wasnt hearing the tones and it seems like i should be
20:34.20UQlevVyrus001: it depends on your client. Tones may be from server or local
20:36.10jayteeVyrus001, how are you testing? sip to sip? calling in from outside, etc?
20:36.36Vyrus001jaytee: sip to asterisk
20:38.39UQlevVyrus001: what sip-client?
20:38.46Vyrus001sjphone
20:40.02jayteesjphone? is that a softphone?
20:41.02Vyrus001yea
20:42.37jayteeVyrus001, what is set for dtmfmode= in sip.conf and what are the dtmf settings of the softphone? rfc2833?
20:42.57UQlevVyrus001: can you paste [internelal] of extensions.conf?
20:43.49jjshoeQwell hrm found it, http://www.kestrelsp.com/FieldTest/
20:44.48Linuturkso, I've got 3 fxo modules configured, but there aren't any phone lines plugged into them
20:44.53Linuturkshouldn't they be showing a red alarm?
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20:49.46voxterLinuturk: fxo ports dont detect line links
20:50.01Linuturkvoxter: ah, thanks :)
20:50.14Linuturkbtw, there's a nice typo in Asterisk: The future of telephony
20:50.15voxternp
20:50.21Linuturksignaling instead of signalling
20:50.24Linuturkthrew me for a bit
20:50.25Linuturklol
20:50.37Linuturkpage 78 for those interested
20:50.52Linuturk2x on that page :)
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21:06.39UQlevLinuturk: when you start use config patterns there will be more fun :))
21:08.56*** join/#asterisk Maliuta_CA (n=biteme@S010600226b81f07b.fm.shawcable.net)
21:16.05LinuturkUQlev: what do you mean?
21:16.45UQlevLinuturk: I mean typos and mistakes
21:16.59Linuturko
21:17.00Linuturkfun
21:17.38*** join/#asterisk qdk (n=qdk@195.242.194.42)
21:25.03*** join/#asterisk JonOnt (n=nonya@72.34.90.74)
21:26.12JonOntHey guys, any one have any problems with HUDlite, i cant use the chat feature
21:26.38*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
21:27.09jjshoeJonOnt wrong channel, try #trixbox
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21:34.13JonOntthanks jjshoe
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21:47.31jayteeping file
21:48.37`SeanIs there anyone who does paid work??
21:48.45`Seani need someone to configure a 7970 for me :D
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21:56.48Vyrus001jaytee: i dont know what the soft phone settings are but my type is set to rfc2833
21:57.05Vyrus001UQlev: exten => 1,n,Set(first=${RAND(1|100000)})
21:57.05Vyrus001exten => 1,n,Set(second=${RAND(1|100000)})
21:57.05Vyrus001exten => 1,n,SendDTMF(${first}*${second})
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21:59.11jjshoeVyrus001 what do you get if you put in the number 1 ?
21:59.46jjshoeVyrus001 or w1
22:00.17[TK]D-FenderVyrus001: What does that code bit have to do with DTMF & a softphone?
22:00.33jjshoeVyrus001 additionally, what dtmf mode is this?
22:02.33kannanare you here?
22:02.37kannansorry
22:02.45*** part/#asterisk reneger (n=reneger@p3EE2C2AE.dip.t-dialin.net)
22:02.48[TK]D-Fenderkannan: We all are ;)
22:02.49*** join/#asterisk reneger (n=reneger@p3EE2C2AE.dip.t-dialin.net)
22:02.56kannanwrong window
22:03.56*** join/#asterisk Supergrilo (n=fabio@unaffiliated/lovezinho)
22:04.17Supergrilohello! anyone can help >>> http://www.pastebin.ca/1298585
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22:09.29Vyrus001jjshoe: nothing happens when i put w1 or 1 in there
22:10.08Vyrus001dtmf mode is default so im assumeing rfc2833
22:10.12[TK]D-FenderVyrus001: And what is supposed to happen?  Pastebin the complete failed call's output
22:10.32Vyrus001there isnt one, it just dosent read the tones... it thinks its working
22:11.27[TK]D-FenderVyrus001: What doesn't read the tones?
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22:12.13Vyrus001[TK]D-Fender: i hear no tones from the softphone
22:12.23Vyrus001but i also see no console error message
22:12.34[TK]D-FenderVyrus001: Quite possible it masks them because it has no reason to care.
22:12.48[TK]D-FenderVyrus001: pastebin the attempt
22:13.11[TK]D-FenderVyrus001: If you want to know if this are right test DTMF *from* your phone, not *to* it
22:13.54Vyrus001[TK]D-Fender: well i know dialing from the soft phone works, ive tested that, i just cant hear anything from asterisk
22:14.08Vyrus001well dtmf tones that is... i hear my background() clips
22:14.29[TK]D-FenderVyrus001: Dialing from the phone isn't prrof, its AFTER the call reaches *, then test dtmf
22:14.39[TK]D-Fendervrythe initial call to * is not DTMF
22:16.40Vyrus001[TK]D-Fender: well i can call in, then dial a submenu within the dialplan using dtmf
22:16.45Vyrus001would that not cout as working?
22:17.49[TK]D-FenderVyrus001: Yes, it would.  This confirms that your phone ignores recieved DTMF because it doesn't do anything for you to hear it
22:18.16[TK]D-FenderVyrus001: Well... aside from certain other thoughts...
22:18.27[TK]D-FenderVyrus001: pastebin that call... I want a sanity check.
22:18.57Vyrus001[TK]D-Fender: the console does not output anything upon the call being a succsess
22:19.21Vyrus001the dialplan is just set to respond with playbacks and (hopefully) dtmf tones, there is no other soft phone that it calls
22:19.42[TK]D-FenderVyrus001: * CLI will show you what gets exectuted.  Pastebin it.
22:19.55[TK]D-FenderVyrus001: Show me the entire call
22:20.14*** join/#asterisk mangefoo (n=mager@c83-250-205-94.bredband.comhem.se)
22:20.55Vyrus001di4l70n3*CLI>
22:21.03Vyrus001thats ALL my cli displays during and after the call
22:21.07Vyrus001there is no error message
22:21.57[TK]D-FenderVyrus001: "set verbose 10"
22:22.42*** part/#asterisk fiXXXerMet (n=kyle@cmu-24-35-53-185.mivlmd.cablespeed.com)
22:23.23Get_The_Fishcan anyone explain the devstate function in asterisk, or point me to a good resource on it?
22:24.37Vyrus001[TK]D-Fender: [TK]D-Fender http://pastebin.ca/1298592
22:24.42[TK]D-Fender~devstate
22:24.43jbot[~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/
22:24.52russellbjbot: devstate is also http://www.asterisk.org/node/48325
22:24.53jbotrussellb: okay
22:25.43Get_The_Fishsweet russell thank you so much.
22:25.51Get_The_Fishand jbot
22:25.56[TK]D-FenderVyrus001: Enable SIP DEBUG and do it again
22:28.55Vyrus001[TK]D-Fender: http://pastebin.ca/1298596
22:29.31Get_The_Fishrussell, do you have time for a private chat?  I think that I have an application for devstate, and I wanted to run it by someone smarter than me :)
22:29.48[TK]D-FenderVyrus001: Found no matching peer or user for '10.10.1.91:5060'
22:29.49russellbbusy at the moment, but I'm sure plenty of others here would be interested
22:30.13[TK]D-FenderVyrus001: Your softphone has not been set up properly, its call is not being authed
22:30.28[TK]D-FenderGet_The_Fish: shoot
22:30.41Vyrus001[TK]D-Fender: no its not but i hear the other parts of the call so i figure this should be working as well
22:30.56Get_The_Fishno worries.  I have a CRM application that I am trying to integrate with Asterisk, and I was thinking that an option would be to incorporate a SIP stack in the application and have it register with asterisk
22:31.45[TK]D-FenderVyrus001: Yes, it is, and that message tells you to your face.  But that still leaves me thinking it masks received DTMF which shouldn't affent anything
22:31.58[TK]D-FenderGet_The_Fish: So far no DevState in there...
22:32.26Get_The_Fishusing Freepbx's device and user configuration, I can register multiple "devices" with asterisk, associate the "user" with those "devices".  This allows me to send a call to multiple "devices" simultaneously.
22:32.45[TK]D-FenderGet_The_Fish: Unless you're hoping to set flags via presence... but any effort spent doing it that way could be spend doing it another that doesn't require a SIP stack
22:34.18Get_The_Fishwell TK, that is pretty much what I was trying to do... the application, when the user is busy and cannot take a call, would return "busy" to asterisk using SIP and the devstate function...
22:34.46Get_The_FishTK, what other way would you suggest?
22:34.59Kobazmrxvt!
22:35.01Kobazer
22:35.03Kobazwrong channel
22:36.16[TK]D-FenderGet_The_Fish: please try that again and get specific
22:37.53[TK]D-FenderGet_The_Fish: understand that Devstate is ONE-WAY.  From Asterisk.  Thats it.  Its not a way to tell * to do anything.
22:38.59Get_The_Fishok, so, in my application I need to be able to tell asterisk when the user is ready for another call, when the user is busy, and the application needs to be able to get information about the call for a screen pop.  The user is busy or available is what I was looking to use devstate for...
22:39.15Get_The_FishTK, good point...
22:40.12[TK]D-FenderGet_The_Fish: Its all just Linux... only about 15 million ways to make that happen.
22:41.15[TK]D-FenderGet_The_Fish: With FreePBX Your only option might be to set the DND value in AstDB for the device
22:41.27Get_The_Fishlol I hear that.  Unfortunately this is a VB6 application (I know I know, but at the moment I dont have a choice).  There is a M$ SIP library that works with VB6.  I was thinking that this would be the path of least resistance :)
22:42.01Get_The_FishTK, I see where you are going with this... interesting idea
22:42.22[TK]D-FenderGet_The_Fish: Thats about all there is because that sad little gui OWNS YOU.
22:42.37Get_The_Fishlol lol lol I know it does.  Trust me.
22:43.00[TK]D-FenderGet_The_Fish: If you were running your own configs the answer would have been "take your pick, the next 10 carts are on their way"
22:43.32Get_The_Fishfreepbx does a lot for our biz though, so replacing it is a tall order
22:43.42[TK]D-FenderGet_The_Fish: Does a lot?  Like what?
22:44.09*** part/#asterisk Supergrilo (n=fabio@unaffiliated/lovezinho)
22:44.22Get_The_Fisheasy configuration of Asterisk for our biz that doesnt require too much IT interaction.  I can have call center managers create new users, for example.
22:44.50[TK]D-FenderGet_The_Fish: Thats about the end of it.
22:45.17[TK]D-FenderGet_The_Fish: Just don't expect to be able to do anything special once you've sold out to it
22:45.31Get_The_Fishwell, for us, that is a big deal actually.  And I hear you on that TK.
22:46.25Get_The_FishI have an issue with CDR's now that Freepbx's presence isnt exactly helping with, but sometimes you have to choose the lesser of two evils.
22:46.55[TK]D-FenderGet_The_Fish: And other's can only see two possibilities.
22:47.51Get_The_Fishwell, any other alternatives that you can think of?
22:48.00jjshoeGet_The_Fish how can you tell in your crm when an agent is available or not?
22:49.33Get_The_Fishthe CRM has some code for tracking the state of the agent already... basically when an agent leaves an account after entering a call result the agent is listed as available
22:49.46jjshoeGet_The_Fish is this tracked in a database?
22:50.07Get_The_Fishyes
22:50.12[TK]D-FenderGet_The_Fish: Make your own
22:50.17jjshoeGet_The_Fish so why not have asterisk check the database?
22:50.23Get_The_FishMSSQL
22:50.47jjshoeGet_The_Fish so why not have asterisk check the database?
22:51.01Get_The_FishI know that asterisk can talk to MSSQL, however my understanding is that it is problematic...
22:51.16jjshoeGet_The_Fish http://www.voip-info.org/wiki-Asterisk+FastAGI
22:52.08jjshoeGet_The_Fish this is super trivial, and should take you less then half a day.
22:53.30[TK]D-FenderJust have * check the database... hillarious....
22:54.17[TK]D-FenderAnd where super-trivial starts as a "sub 1/2 day" calc...
22:55.03mchou~book
22:55.03jbotfrom memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
22:55.57Get_The_Fishthanks jjshoe.  I will look at this further
22:56.48[TK]D-FenderGet_The_Fish: AGI does not apply.
22:57.19Get_The_Fishhow so, if I may ask?
22:57.49_ShrikEcheck func_odbc
22:58.19[TK]D-FenderGet_The_Fish: because you need * to consider them available or not.  FreePBX controls everything.  About the only flag it check that you can set is that DND flag.
22:59.06[TK]D-Fender_ShrikE: What part of "FreePBX owns his sorry ass" don't people get?  You don't get to go changing the rules about how devices get called like that.
22:59.12jjshoe_ShrikE how does that behave if the db is un-reachable?
23:01.54*** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net)
23:02.18_ShrikEtake it easy
23:02.48_ShrikEjjshoe: I believe func_odbc can have an alternate entry in 1.6
23:05.08[TK]D-Fender_ShrikE: in a setup he controlls, yes a DB lookup is quite doable.  To help survivability it could even cache the last retrieved state to carry through a read-failure
23:05.23jjshoe_ShrikE seems worth looking at, I'm sure the performance is pretty close
23:05.36eppigyholla
23:06.35_ShrikEjjshoe: lmadsen did a nice section on it in the book.
23:10.00jjshoe_ShrikE cool, will look it up
23:11.15jjshoe_ShrikE hrm nice, does have fail over..
23:20.25*** join/#asterisk fiXXXerMet (n=kyle@cmu-24-35-53-185.mivlmd.cablespeed.com)
23:22.10fiXXXerMetI am having some seemingly truly intermittent call audio issues.  I have made at least a dozen inbound and outbound call.  Some calls, we can talk to each other.  Some calls, only one party can hear the other, and some calls, it is the opposite.
23:22.22fiXXXerMetHave been told that this could be a NAT issue, but my ports seem to be setup correctly.
23:22.25fiXXXerMetAny other ideas?
23:27.54[TK]D-Fender~sipnat
23:27.55jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
23:28.00[TK]D-FenderfiXXXerMet: its a lot more than just ports
23:28.01Madkisscurses at hylafax+iaxmodem+foobar.
23:28.20[TK]D-Fendermadsara: uninstall fubar
23:29.12Madkissi have pretty much no idea what's going wrong there. "ANSWER: Ring detected without successful handshake"
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