IRC log for #asterisk on 20100112

00:02.02*** join/#asterisk etfonhomey (n=etfonhom@173-8-16-185-WashingtonDC.hfc.comcastbusiness.net)
00:04.44CRCinAU_urgh
00:04.45CRCinAU_https://issues.asterisk.org/view.php?id=13405
00:05.13CRCinAU_Submitted: 2008-08-30 - been confirmed, still not merged :(
00:10.56*** join/#asterisk cesar_CR (i=cesar@201.201.41.242)
00:15.47*** join/#asterisk corretico (n=laguilar@201.201.46.106)
00:17.59*** join/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net)
00:19.29LemensTShey all. im wanting to show channel info on a php page. I can do exec("/usr/sbin/asterisk -rx 'show channels' |grep 'active channels' | awk '{print $1}'") and retrieve the info, but i was thinking this may take more processing than using ami?
00:19.56LemensTStrying to build the status page like in freepbx
00:20.07*** join/#asterisk coppice (n=chatzill@202.64.176.25)
00:24.29*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
00:25.57dzuphow many simultaneus call can i hav in 6 meg internet conecction?
00:26.19*** join/#asterisk jo8330 (i=d04149c8@gateway/web/freenode/x-nfaxtayhbtvwvezg)
00:26.28jo8330howdy all
00:27.36jo8330is it possible to specify caller id name from making PSTN calls from a PRI?
00:28.36jo8330i was surprised to find i could specify any originating caller id number.. could play some good pranks lol
00:29.07Kobazjo8330: no, name is set at your provider
00:29.54dzupKobaz: if i rent 6megs internet access with this ISP, how many calls can i make at the same time?
00:30.27jo8330i see
00:31.02*** join/#asterisk roe (n=admin@smtp.harmonytownshipschool.org)
00:31.34DarkFibredzup: whats the upstream?
00:32.08dzup2048kbs i believe
00:32.30DarkFibredzup: what codec you going to use?
00:32.52dzupthe default ones
00:32.59roeCan anyone suggest a good system integrator that sells compact server grade systems appropriate for use with asterisk.  I perused the voip-info website but nothing popped out at me, all I could find were 'solution providers'.  I'm interested in just the server, preferably half-depth with 2 or more expansion slots, it seems to be slim pickings
00:33.25dzupDarkFibre: is 2048Mbs  <--note the M
00:33.37Kobazdzup: divide by your codec bitrate
00:34.07dzupKobaz: which is the regular SIP account?
00:34.19dzupdont know that value, thats why am asking heh
00:34.19DarkFibreyes but you will also have to allow for "regular" internet traffic
00:34.45DarkFibreg711 (ulaw) is 880K per channel
00:34.53dzupDarkFibre: can i get along with 6 calls?
00:34.53DarkFibre*80K
00:35.03Kobazwhat?
00:35.09KobazDarkFibre: where did you pull that from
00:35.17Kobazulaw is 64kbit
00:35.28DarkFibreyou are correct
00:35.40DarkFibremy spreadsheet allows for normal internet use :)
00:35.48Kobazheh
00:36.13DarkFibremy apologies I deal with management all day and I found it easier to build it in than try to argue it :)
00:37.04DarkFibre128kbit per call raw dzup
00:37.05dzupthat will be like 26 some
00:37.15DarkFibreyou need 2 channels per call
00:37.48coppiceulaw is 230Mbit
00:38.03coppicewe all use the hourly rate, don't we?
00:38.19dzup2048 / 128 = total lines, correct?
00:38.19dzupis there any network overhead i have to concider?
00:38.29DarkFibreI use per second for call center apps
00:38.32DarkFibreyes
00:39.10DarkFibrealso depends on how much internet usage you have going on (web, email, etc)
00:39.26DarkFibrehow many reps you trying to serve
00:44.37*** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com)
00:45.15Kattymy poor hair
00:45.19Kattymy poor poor hair.
00:45.25Kattyit's so brittle and tangled :<
00:47.07coppicebecome a witch. you'll be right in character
00:47.48dzupsoo out of 2048mbs at 64kbs give me 32 calls at anytime ?
00:48.01dzupis there any consideration on network overhead?
00:52.27ManxPower-workdzup: 2 channels of overhead IIRC
00:52.50ManxPower-work1 channel I never understood (it's an e-1 thing) and the D-channel for PRI messages
00:53.27dzupManxPower-work: yeah and my bots thats line minus 4 lines
00:55.23nnyheh how do you reload apps in 1.6?
00:56.34nnynm
00:56.36nnymodule reload
00:57.32nnyheh so fing close
00:58.12jo8330hmm you can fake caller id
00:58.23jo8330it works... "Name <num>"
01:01.21nnysoo close
01:01.23nnygrrr
01:01.46*** join/#asterisk epoxy (n=epoxy@pdpc/supporter/active/epoxy)
01:01.58*** part/#asterisk epoxy (n=epoxy@pdpc/supporter/active/epoxy)
01:03.33nnyso if I run a feature against "peer" wouldn't CHANNEL be the channel they are on?
01:03.38nnyin this case BRIDGE
01:05.19nnyooh nm
01:11.01*** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id)
01:11.19*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
01:12.48*** join/#asterisk chendy (n=chatzill@113.88.48.69)
01:17.53*** part/#asterisk _zen_ (n=_zen_@cpe-74-66-140-78.nyc.res.rr.com)
01:33.51jo8330i'm still in disbelief how easy it is to spoof caller id
01:34.24jo8330you could do some really malicious things
01:38.06*** join/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
01:39.28nnyKobaz: suckcess
01:39.58nnyKobaz: can dial internal extensions, and each person can transfer to his/her cell as well as dial in or out on the trunks and do the same
01:40.18nnyjo8330: can't spoof on PSTN usually though
01:42.17jo8330i would hope not. it works here for me though
01:42.44nnynice
01:42.49jo8330i want to make some michael jackson calls.. lol
01:42.55nnyyeah our telco forces w/e number you have from them set
01:43.00nnybut VoIP doesn't
01:43.32jo8330i had to look through the source to see how to specify name
01:43.35jo8330i couldn't find documentation on it
01:44.01nnyyou mean Set(CALLERID(name)
01:44.23jo8330i'm using agi originate which takes CallerID as a single string
01:44.41jo8330but i guess that would have worked too
01:44.48ManxPower-workjo8330: The docs are in /path/to/src/doc/
01:45.12ManxPower-workspecifically channelvariables.tex and some AGI documentation
01:45.27nnygod damn that was the hardest thing I have had to grasp in asterisk yet
01:45.44*** part/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
01:47.07*** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id)
01:55.30*** join/#asterisk sgimeno (n=santiago@80.33.64.226)
01:56.06*** join/#asterisk lordmortis (n=lordmort@203-206-67-161.dyn.iinet.net.au)
01:56.24*** part/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net)
02:10.16*** join/#asterisk minotaur01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)
02:10.43*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
02:11.28*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
02:12.25nnyohhh joy
02:12.32nnytime to try and get SLA working on a SPA962
02:13.07*** join/#asterisk minotaur01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net)
02:16.01*** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
02:17.02nnyanyone know of a doc for SLA and * that isn't based on 1.2?
02:28.17*** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147)
02:28.26LemensTSmeow
02:30.10*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
02:34.37*** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru)
02:37.05*** join/#asterisk sjobeck (n=sjobeck@74.85.224.166)
02:37.35*** part/#asterisk sjobeck (n=sjobeck@74.85.224.166)
02:40.00jo8330what does agi_callington: 33 mean?
02:44.11*** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361)
02:53.14*** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net)
02:58.10Kattypeeks in
03:04.00*** join/#asterisk denon (n=denon@synapse.subneural.net)
03:04.00*** mode/#asterisk [+o denon] by ChanServ
03:06.26Kattyhi denon
03:07.01Kattyanyone alive?
03:07.04Kattypokepoke
03:10.14voipmonkKatty - :) its a girl!
03:10.16voipmonk:)
03:10.41p3nguinHappy birthday?
03:10.51voipmonknot yet :)
03:10.54voipmonkjust sonogram
03:10.57p3nguinah
03:11.07ManxPower-workYou can SIPAddHeader.  How do you delete a header?
03:14.55dgilmoreafter some digging and resorting to a very basic iptables config for some reason iax is not getting natted by my router
03:17.23p3nguinport forwarding fail?
03:23.19dgilmorep3nguin: i turned all the port forwarding off
03:27.32*** join/#asterisk _bugz_ (n=bugz@adsl-76-194-247-113.dsl.lsan03.sbcglobal.net)
03:30.11*** join/#asterisk hluesea (n=hulusika@88.247.127.66)
03:32.13p3nguindgilmore: That could be why it doesn't work.
03:32.27p3nguinYou have to forward the IAX2 port to the Asterisk server.
03:33.05*** join/#asterisk sjobeck (n=Adium@74.85.224.166)
03:33.16*** part/#asterisk sjobeck (n=Adium@74.85.224.166)
03:34.30ManxPower-workJust because you are using IAX2 doesn't mean you don't have to forward a port
03:38.03*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
03:38.46*** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
03:40.07devmodHello
03:40.42devmodI copied app_mp4.c into asterisk's app dir, recompiled nad installed. When I tried loading the module I get some undef symbols... http://pastebin.com/m6711d9c4 Any ideas?
03:40.45nnyhey quick Q, I have parkcalls=*5 in faetures.conf how the heck do I enable it
03:41.07dgilmorep3nguin: not at all
03:41.12nnyi tried DYNAMIC_FEATURES=parkcall)
03:41.14nnyno workie
03:41.19dgilmorepethe traffic is being forwarded without being natted
03:41.57dgilmorep3nguin: ive never forwarded the iax port in the past.  i connect to my providers  no one connects to me via iax
03:42.49p3nguinnny: Well, the option should be "parkcall => *5" for that type of parking.
03:43.10nnysorry
03:43.11nnyparkcall => *5
03:43.13nnyyes
03:43.25nnybut if I try to use the feature, asterisk ignores the keystroke
03:43.32p3nguinnny: But you really don't need to do it that way if you are willing to do two-step (semi-attended) transfer to park calls.
03:43.49nnyp3nguin: no I am used to 2 step, I want a single button press for what I am testing right now
03:44.04nnybasically long story short trying BLF + Park Calls instead of SLA for stupid needy clients
03:44.34nnyI started looking into SLA and considered a slow suicide with blunt objects and salt
03:44.38ManxPower-worknny: would you like a cool link?
03:44.38p3nguinnny: Did you make sure you included the k and/or K options in the Dial() command?
03:44.48nnyManxPower-work: I looove cool links
03:44.52nnyp3nguin: DOH
03:44.56ManxPower-workhttp://knowledgebase.polycom.com/knowledgebase/End%20User/Tech%20Alerts/Audio/Using_SPIP_SSIP_Asterisk_TB43565.pdf
03:45.14ManxPower-worknny: you are suffering from fatigue
03:45.21nnyno kidding heh
03:50.15*** join/#asterisk corretico (n=laguilar@201.201.46.106)
03:55.49Kattyvoipmonk: omg
03:55.51Kattyvoipmonk: omgomgomg
03:55.53Kattyvoipmonk: IT"S A GIRL
03:55.58voipmonk:)
03:56.04Kattyvoipmonk: TIME FOR PINK BOMBER HATS
03:56.06Kattyhugs voipmonk
03:56.13Kattycongrats!!!!!!!!!!!! :>
03:56.26Kattypicture picture picture!
03:56.27KattyPICTURE!
03:56.36voipmonkgoing up tomorrow
03:56.37voipmonklol
03:56.41Kattyarg >.<
03:56.42Kattytoo long!
03:56.47voipmonkshe kicks
03:56.49voipmonkand punches
03:56.54Kattywell of course she does.
03:56.55voipmonkalready
03:56.57Kattyshe's a /baby/
03:57.12voipmonkshe hid her face from us for a minute, too
03:57.22Kattydoubt that was intentional.
03:57.25*** join/#asterisk Kumbang (n=kumbang@167.205.24.69)
03:57.30Kattydid you get a video clip?
03:57.37*** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net)
03:58.02voipmonkbought the cd
03:58.04voipmonkyes i did
03:58.10Kattyooooh and are you posting that too? :>
04:04.53*** join/#asterisk mrbnet_ (n=ryanbant@c-75-73-142-28.hsd1.mn.comcast.net)
04:05.17mrbnet_do I need to install dahdi if I am only running sip?
04:05.41Kattymeet,e
04:05.43Kattymeetme
04:07.59ChannelZmrbnet: you need dahdi's dahdi_dummy if you want to use conferencing or something else I'm forgetting
04:08.21eppigybehind the boat house
04:08.25eppigyill show you my dark secret
04:09.10mrbnet_ChannelZ: thank you
04:09.52*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
04:17.43*** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-rttwvqqdryjblqov)
04:22.17*** join/#asterisk soman (n=somnath@118.102.130.6)
04:22.59*** part/#asterisk nny (n=scott@64.203.239.83)
04:25.01*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
04:33.55*** join/#asterisk nny (n=scott@64.203.239.83)
04:34.32nnyquick q, if I am using *5 + Dial Kk to park a call, and I pick it back up, it ignores the next feature code. Is there a way to fix this?
04:35.13p3nguinHow are you picking up the parked call?
04:35.54nnyexten => 1,1,ParkedCall(1)
04:38.38p3nguinExplicitly in the dialplan, or implicitly like ParkedCall() is normally used?
04:38.48nnyExplicitly in the dialplan
04:41.30p3nguinThis is an excellent question.  This might be something a developer would know how to accomplish.
04:41.43p3nguinI can't imagine no one else ever ran into this before.
04:43.59*** join/#asterisk kc8pxy (n=gecko@75-145-57-201-utah.hfc.comcastbusiness.net)
04:45.13nnyi'll check tomorrow, thanks p3nguin
04:47.06[TK]D-Fendernny: Dial a non-bridging Local channel to do the pickup with the appropraite options
04:47.52nny[TK]D-Fender: hmm you mean other than  1,1,ParkedCall(1) i assume
04:47.59nnylike Dial (Local/1) ?
04:48.10[TK]D-Fendernny: Dial the Local channel with options targeting a pickup exten
04:48.10p3nguinI wondered if something like that would work, but I couldn't see how it could.
04:49.47*** join/#asterisk pawz (n=pawz@ppp118-208-191-139.lns20.bne4.internode.on.net)
04:50.08p3nguinI've always parked calls using the regular "Trnsfer button, 700, Trnsfer button" method, so I've never noticed that problem.
04:50.23nnyjust noticed features.c:3289 park_call_exec: Invalid timeout 'stdexten' provided too
04:50.27nnywonder how it
04:50.31nnyit's getting that
04:51.03nnyonly two areas in the dialplan that have stdexten, it's a macro for dialing certain extensions, once it is defined, and once again it is called against the extension that uses it
04:52.14*** join/#asterisk sjobeck (n=Adium@valdisere.sjobeck.com)
04:52.31*** part/#asterisk sjobeck (n=Adium@valdisere.sjobeck.com)
04:52.33kc8pxyi'm making a new asterisk box, and i have a few choices on my setup. my question is, does a hardened kernel play badly with asterisk??  it's been too long.
04:55.22nny[TK]D-Fender: you mean like 1,Dial(Local/1,,Kk)
04:55.23nny?
04:56.20nnyhttp://pastebin.com/m1b250d6a is what I have right now
04:56.46nnyer
04:57.09*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
04:57.14[TK]D-Fendernny: nny Don't use the parked calls context at all
04:57.44[TK]D-Fendernny: and your "1" is RECURSIVE
04:57.49nny[TK]D-Fender: actually the parked calls context is parkedfeat
04:57.52[TK]D-Fendernny: Shoo-on-sight programming error
04:58.03[TK]D-Fendert
04:58.15[TK]D-Fendernny: #9
04:58.33[TK]D-Fendernny: [parkedcalls]  do not create it, do not use it, do not include it
04:58.42nny[TK]D-Fender: sorry that was a hack suggested on voip info ti get BLF working properly
04:59.08nnyhttp://www.voip-info.org/wiki/view/Asterisk+cmd+ParkAndAnnounce
04:59.09[TK]D-Fendernny: nny and THAT context name didn't match up anyway
05:00.01nnyi'll review
05:00.52nnyThe hint should point to the extension AND context defined in "features.conf" NOT to a REAL extension as you would expect!  is what I was trying to emulate
05:02.41nnyhmm loooks like parkhints=yes is implemented now
05:03.27nnywell I dunno reading this v
05:03.29nnyer https://issues.asterisk.org/view.php?id=5779
05:05.12nnydammit I dunno that's from the patch I think
05:05.27nnyyou'd think BLF monitor of parked calls would be easier to do :\
05:07.53nnyshit i'm lost now
05:08.27nnyso I am suppose to remove the entire parkedcalls extension and just have the hints?
05:09.59[TK]D-Fendernny: Do not USE that context.  You do not need it.
05:10.21nnyok I do need the hints, obviously
05:10.29nnyso I removed the include and the context below
05:10.34nnyand just have [hints] now
05:12.00nnyand somehow after I park the call it jumps right to the next line in the context that was used to initally dial.. but not always
05:12.45nnyi'll just mess with it tomorrow, I am tired and not thinking straight anymore
05:12.52nnythanks all, bbl
05:12.58*** part/#asterisk nny (n=scott@64.203.239.83)
05:13.40*** join/#asterisk dkirker-openmobl (n=dkirker@adsl-99-36-41-192.dsl.snlo01.sbcglobal.net)
05:19.40*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
05:23.36ManxPower-work[TK]D-Fender: http://knowledgebase.polycom.com/knowledgebase/End%20User/Tech%20Alerts/Audio/Using_SPIP_SSIP_Asterisk_TB43565.pdf
05:25.02[TK]D-FenderManxPower-work: OMG... thats actually... USEFUL
05:25.17[TK]D-FenderManxPower-work: And condensed..... who did we have to bribe? :p
05:27.40[TK]D-FenderManxPower-work: seriously thorough
05:37.15*** part/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com)
05:50.02*** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil)
05:55.38*** join/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
06:05.23*** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110)
06:10.34*** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com)
06:12.48*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
06:19.47*** join/#asterisk ibolmo (n=ibolmo@cpe-67-11-89-134.rgv.res.rr.com)
06:20.21ibolmohey guys, just starting out. i'm wondering if i should have my own server, or use a vps like slicehost (if at all possible) to host asterisk
06:21.07ZX81if you're just starting out, I would have thought you'd be better off with a server if it's not too much of a mission to get hold of
06:21.16ZX81you can pretty much just use any old computer
06:21.40ZX81throw debian or your favorite linux on it
06:21.45ZX81and then asterisk
06:21.51ibolmowell i mention slicehost, because i'm a little confused on other people's setup
06:22.09ibolmoi haven't asked, but i don't think the physical copper lines are going into the slicehost boxes right?
06:22.18ZX81no idea
06:22.22ibolmoit'd be local to voip then to slicehost and then to our networks
06:22.58ZX81looks like it might work
06:23.04ZX81only thing is
06:23.19ZX81if you come across any problems, there might not be that many people doing the same thing
06:23.33ZX81so if the problem was solely related to the vps you might have troubles
06:23.47ZX81if you can get a short term account, go for it :)
06:23.51ZX81try it out
06:23.55ZX81see if it works :)
06:24.10ZX81it's definitely looking like the way for some products for the future
06:24.37ZX81it will also depend on the quality of their connections to the places you might have voip providers/devices etc
06:25.02ZX81and how often, what happens if they go down
06:25.11[TK]D-Fenderibolmo: Who says you need physical copper?
06:25.21[TK]D-Fenderibolmo: That environment has nothing to do with learning *
06:25.27[TK]D-Fenderibolmo: And is a backwards waste
06:25.32ZX81if you've got a physical box with normal phone lines you can fail over to them if you lose net
06:25.39[TK]D-Fenderibolmo: Go get your own box, grab the book and get to work
06:25.40ibolmoi meant PSTN
06:25.47ibolmo(still learning some of the terms)
06:25.55[TK]D-Fenderibolmo: You also don't ened PSTN access to learn *
06:26.01[TK]D-Fenderneed*
06:26.03ZX81totally
06:26.09ibolmosure [TK]D-Fender i understand
06:26.12ibolmothinking ahead
06:26.15ibolmoand curiosity
06:26.21ibolmoof why someone would want a hosted solution
06:26.40ibolmoe.g. http://lylix.net/
06:27.08[TK]D-Fenderibolmo: Because of fear of having to power a server in case of failure, in order to centralize multiple locations where you can't afford the bandwidth in one place you control, etc
06:27.28[TK]D-Fenderibolmo: Because other hosts come preconfigured with GUI's so admins can be morons
06:27.35[TK]D-Fenderibolmo: Any of a ton of other reasons
06:27.56[TK]D-Fenderibolmo: In the end they sacrifice control
06:28.06[TK]D-Fenderibolmo: And pray it was all worth it
06:28.33*** part/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
06:32.04[TK]D-Fendercheckout time, later all
06:35.12*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
06:51.58*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
06:52.58*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
06:54.35*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)
06:57.31*** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com)
06:58.13*** join/#asterisk cuco (n=Diego@local.xorcom.com)
06:58.53*** join/#asterisk soman (n=somnath@118.102.130.6)
07:02.39sun28moin
07:07.40ibolmoi guess some of the features in switchvox are closed source?
07:08.36*** join/#asterisk benngard (n=benngard@213.88.138.230)
07:09.02sun28ibolmo: right
07:10.01ibolmobut i take it that there's some OS alternatives?
07:17.00*** join/#asterisk infobot (i=ibot@rikers.org)
07:17.00*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.0 (2009/12/18), Asterisk 1.6.1.12 (2009/12/18), 1.6.0.20 (2009/12/18), 1.4.28 (2009/12/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #aster
07:17.12kc8pxyibolmo:  please excuse my ignorance. what switchboard features?
07:17.31ibolmohttp://www.digium.com/en/products/switchvox/smb.php?tab=features
07:17.36sun28voice recognition ?
07:17.57ibolmoe.g. Record others' calls
07:18.34ibolmoCRM Panel
07:18.51ibolmothose two are probably the first i'll want to find
07:22.54*** join/#asterisk ayrjola (n=ayrjola@gw.ccxtech.fi)
07:34.19*** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za)
07:34.44*** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185)
07:37.45kc8pxyi am trying to connect my asterisk server to my vonage account. all the info i've been finding, is like 2006. is ther recent howto that explains this? and can i hook it up for any number of vonage boxen?
07:45.52*** join/#asterisk tengulre (n=tengulre@125.71.208.16)
07:49.10jkroonhi guys, what exactly does prematuremedia do?
07:49.35jkrooni'm trying to find info on google but google isn't being helpful to me this morning.
08:00.49*** join/#asterisk Faustov (i=user@gentoo/user/faustov)
08:01.07*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
08:01.50*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
08:02.34*** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net)
08:04.49*** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com)
08:05.30*** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie)
08:05.54*** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
08:06.21*** join/#asterisk fofware (n=chatzill@190.7.25.160)
08:20.48*** join/#asterisk jtodd (i=n6qlzbwd@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) [NETSPLIT VICTIM]
08:20.49*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) [NETSPLIT VICTIM]
08:20.49*** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) [NETSPLIT VICTIM]
08:20.49*** join/#asterisk corretico (n=laguilar@201.201.46.106)
08:20.49*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) [NETSPLIT VICTIM]
08:20.49*** join/#asterisk mocker (n=mocker@206.55.118.85) [NETSPLIT VICTIM]
08:20.49*** join/#asterisk mrbnet (n=mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) [NETSPLIT VICTIM]
08:20.49*** join/#asterisk Rawplayer (n=kevin@wickedleaks.nl) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk |AnToS| (n=31749@93-44-17-172.ip95.fastwebnet.it) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk bpgoldsb (n=bpgoldsb@ip24-250-198-162.ga.at.cox.net) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk bzing2 (n=dr105@dhcp-194-66-208-235.canterbury.ac.uk) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk festr_ (n=festr@77.93.216.244) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk plundra (i=404@article.se) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk mahlon (i=mahlon@martini.nu) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk De_Mon (i=de_mon@fl-76-3-31-48.dhcp.embarqhsd.net) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk dc3 (n=dc3@tinderbox.complexminds.com) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright)
08:20.50*** join/#asterisk clintc (n=clintc@n128-227-2-41.xlate.ufl.edu) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk RypPn (n=TuMbL@rosscom.co.uk) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk zaoul (n=zaoul@231.109.33.65.cfl.res.rr.com) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk p3nguin (i=gpz5GvdF@24-171-73-146.dhcp.mtvr.il.charter.com) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk ChannelZ (i=channelz@burner.com) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk toppe (n=root@cs181001225.pp.htv.fi) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk hardwire (n=spencers@39.183.dowl.anc.borealisbroadband.net) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk dustybin (i=dustybin@thinkdebian.org) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) [NETSPLIT VICTIM]
08:20.50*** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk sbrath (n=sbrath@unaffiliated/sbrath) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk porter (n=terdon@unaffiliated/porter/x-000001) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk steve745 (n=W33DS@pool-96-243-156-36.tampfl.fios.verizon.net) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk Corydon76-dig (i=indigo@c-69-137-80-31.hsd1.tn.comcast.net) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk justdave (n=dave@unaffiliated/justdave) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk digitalirony (n=digitali@my.ass.looks.just.like.your-face.info) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) [NETSPLIT VICTIM]
08:20.51*** join/#asterisk x86 (n=x86@p3m/member/x86) [NETSPLIT VICTIM]
08:20.51*** mode/#asterisk [+ooo jtodd angler Corydon76-dig] by irc.freenode.net
08:22.12*** join/#asterisk af_ (n=getsmart@88-149-241-228.dynamic.ngi.it)
08:23.21*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
08:24.30ChannelZstupid network
08:28.53jkroonlol @ ChannelZ
08:32.14*** join/#asterisk corruptor (n=corrupto@85.94.32.98)
08:32.31TommyBottenHow can I see who is a watcher for a specific hint?
08:33.15*** join/#asterisk oej (n=olle@ns.webway.se)
08:33.19*** join/#asterisk eddyo (n=edd@91.72.212.165)
08:33.41eddyohi all, anybody online now for a fresh setup of an asterisk ?
08:33.50TommyBottenhm?
08:35.07eddyoi want to install freepbx, im asking if somneone is here
08:36.20TommyBottenWell, there are people here. But FreePBX specifics are on #freepbx
08:37.50*** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
08:39.05*** join/#asterisk CRCinAU_ (n=CRCinAU@irc.crc.id.au)
08:39.06jkroonTommyBotten, sip show subscriptions iirc.
08:39.20CRCinAU_can feel another anti-digium rant coming on :
08:39.21jkrooneddyo, #freepbx :)
08:39.38CRCinAU_where the smeg can I file a bug report on Digiums commercial Fax app? :|
08:40.05jkroonissues.asterisk.org ?
08:40.14CRCinAU_I go to the support area to lodge a case, and it says sorry, you need to log in. So I log in, go to the support area, and it tells me I have to register my device.
08:40.40CRCinAU_I can't tell it I have a device because nothing matches the fax application in the area to register
08:41.15CRCinAU_jkroon: I've been given rabid lectures in the past that issues.asterisk.org is only for the free open source stuff and not the commercial Digium offerings
08:41.28jkroonCRCinAU_, ah ok :)
08:41.51CRCinAU_yet I have no fucking idea where the hell I'm supposed to lodge a bug report.
08:42.06CRCinAU_as there's no place in the digium web site where I can file it
08:42.13TommyBottenjkroon: Thanks a bunch :)
08:42.17CRCinAU_well, not without registering a device? :|
08:42.41jkroonrofl.  yea well, honestly, I'm not a fan of digium at this point, but they do head the asterisk development, so whether one likes them or not really is irrelevant.
08:42.44jkroonTommyBotten, pleasure.
08:42.48CRCinAU_I just get directed to this: https://www.digium.com/en/users/support_choose_product.php?tab=log
08:42.53CRCinAU_which doesn't let me do shit.
08:43.09CRCinAU_as a company, it's like they have their head up their ass.
08:43.20CRCinAU_and I don't mean a small amount :\
08:43.48CRCinAU_it's like they sooooo want to do the right thing, but only see it from an admin viewpoint - not as the view of someone who has to deal with them.
08:44.45CRCinAU_I even found a link that said "Create a new support case" which then told me "In order to obtain support, your elegibe products and/or subscriptions need to be regsitered"
08:44.57CRCinAU_then they have 3 options - and the commercial fax offering is none of them.
08:45.07*** join/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
08:45.16TommyBottenThey have a chat feature... Have tried that?
08:45.40mchouI think they do that to discourage reports from users i=of free software
08:46.17mchoui.e. not bona fide digium customers
08:46.30CRCinAU_TommyBotten: but why offer shit for sale, when they can't offer support for it?
08:46.33jkroonmchou, that makes sense.  but honestly, i've logged so many report queries surrounding G.729 licenses and haven't received a single reply, ever.
08:46.46CRCinAU_TommyBotten: where is this chat thing/
08:46.47CRCinAU_?
08:46.50jkroonthink I've lost probably in excess of 100 channel licenses to date.
08:47.03CRCinAU_jkroon: hahahah yeah - I had the same issue too
08:47.05jkroonthat starts adding up.
08:47.14CRCinAU_I think I lost ~20 or so
08:47.25CRCinAU_so I said fuck it and just used the free ones.
08:47.25jkroonand it's BS that you can just "take a NIC and move it from machine to machine"
08:47.32*** part/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
08:48.08jkroonCRCinAU_, i use this as part of a commercial offering, I really need (want) to keep it above board.  but i have to admit digium is slowly driving me up the creek.
08:48.27CRCinAU_jkroon: I had issues where a new kernel would shuffle my nics around - each time it did a shuffle, the licenses would be invalid.
08:48.50TommyBottenCRCinAU_: I can't find it. Might have been removed.. It's been a while since I surfed their pages.
08:48.57CRCinAU_so the licenses could really be invalid after a single reboot.
08:49.24CRCinAU_jkroon: I really want to submit a bug report on something that they sell - but fucked if I can find where.
08:49.37CRCinAU_and sadly, I can't justify spending 2-3 hours trying to do it.
08:49.54CRCinAU_I have everything in a text docuemnt - just looking for somewhere to submit it.
08:50.23mchouCRCinAU_: how about just via email?
08:50.45CRCinAU_this is a simple one where if a fax fails, then the # of sessions isn't decreased after the fail.
08:50.45ChannelZThey have a phone number.
08:50.58*** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es)
08:50.59CRCinAU_so 1 failed fax would give you 1 used session, another failed gives you 2 used sessions, etc etc
08:51.00jblackhttp://www.digium.com/en/company/careers/  -- Looks like they're trying to fix that
08:51.00casixhello
08:51.06CRCinAU_until you restart asterisk :\
08:51.34CRCinAU_ChannelZ: phone number is only helpful if you live in the US.
08:51.47jkroonCRCinAU_, ditto, had that one too.  but even force-shuffling the order back using udev didn't help.
08:51.57ChannelZWhat, your phones don't call other countries?
08:52.03casixwhen I park a call with the cmd Park, the parked listen the number of the room where is parked. How can I aviod this? I don't want he listen the number
08:52.14jkrooncore show application Park()
08:52.22CRCinAU_ChannelZ: noooo, I don't want to have to be up at 3am in the morning for their opening hours.
08:52.22jkroonwithout the brackets.
08:52.29casixthx
08:53.27casixjkroon: there is no option for that... this means is not posible to avoid this?
08:53.59casixi'm using asterisk 1.4.23.2
08:54.15jkrooncasix, I don't recall having that problem when I tried it two years back, just re-read how that thing works and make sure you're using it right.  I can't remember what the exact semantics was.
08:54.23ChannelZThere's no 's' option listed in 1.4?
08:54.31jkroonyour problem seems globally valid.
08:54.50jkroonso i can't imagine it not having a solution.
08:55.24casixok, I will search
08:55.26casixmore
08:56.33*** join/#asterisk ayrjola (n=ayrjola@gw.ccxtech.fi)
08:56.50*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
08:57.37*** join/#asterisk [netman] (n=netman@193.153.155.169)
08:58.15CRCinAU_fuck it.
08:58.44CRCinAU_emailed the bug to customerservice@digium.com - as they are the ones that sent me the license details for the commercial fax app.
08:59.10CRCinAU_stupid that you have to look that far into things just to submit a support case :\
09:01.26CRCinAU_I'd love it if a startup would fork asterisk an then just offer real support :)
09:01.44CRCinAU_oh - and have a decent web design that worked with people, not against them.
09:02.15TommyBottenThere exists a fork of Asterisk
09:02.35CRCinAU_oh?
09:07.36*** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net)
09:08.08TommyBottenCallweaver
09:08.13TommyBottencallweaver.org
09:08.38TommyBottenIt has more or less forced asterisk to adopt some features, but it didn't really take off.
09:08.39*** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99)
09:08.43TommyBottenBut it's a good project none the less.
09:12.01*** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net)
09:16.58*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
09:17.22*** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr)
09:18.07*** join/#asterisk mpe (n=mpe@217.74.220.92)
09:19.21*** join/#asterisk mpe (n=mpe@217.74.220.92)
09:21.11*** join/#asterisk mpe_ (n=mpe@217.74.220.92)
09:21.25eddyohi there
09:21.26eddyohttp://pastebin.ca/1748043
09:21.30eddyoany suggestion
09:21.57*** join/#asterisk mpe (n=mpe@217.74.220.92)
09:22.09ChannelZyou don't have gcc installed
09:22.25TommyBottenYou're missing a compiler
09:22.28TommyBottenah
09:22.33eddyohow to install the complier?
09:22.39TommyBottenwhich distribution are you running?
09:22.54ChannelZand why are you asking questions about LAME in an Asterisk channel?
09:26.30*** join/#asterisk mpe_ (n=mpe@217.74.220.92)
09:27.53*** join/#asterisk mpe (n=mpe@217.74.220.92)
09:28.23jkroonok guys, given an A number and a B number (and channels as per cdr-csv), how can I extract the SIP conversations (one on each leg of the call) for that particular call?
09:29.40*** join/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
09:29.49*** part/#asterisk ZX81 (n=Matt_Rid@121-74-232-236.telstraclear.net)
09:32.59*** join/#asterisk mpe (n=mpe@217.74.220.92)
09:36.58*** join/#asterisk disposable (i=disposab@blackhole.sk)
09:38.07disposableis there anything like denyhosts (which is for ssh) but for sip registrations that come to my asterisk boxes? i'm getting millions of scripted registrations from various addresses and it's too laborious to keep adding them to hosts.deny manually
09:39.08jkroonperhaps it's time to write an asterisk script for fail2ban.
09:39.36*** join/#asterisk Jenna (n=JJ@unaffiliated/jenna)
09:41.41jkroonand yes, there is a deny= and allow= mechanism in asterisk.
09:41.56jkroonmy personal preference is to just add an iptables rule to drop packets coming from those subnet.
09:46.22JennaHiyall
09:46.37Jennaanyone encountered this error before:  "find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available"
09:47.14disposablejkroon: thank you
09:47.20sun28check res_mysql.so
09:47.34*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
09:49.25Jennasun28, hmm. weird. nowhere to be found
09:49.41Jennasun28, you know what package this belongs to ?
09:50.02Jennadebian 5 i.e
09:50.04TommyBottenJenna: It is a module loaded by asterisk.
09:50.24TommyBottenJenna: try "module show like mysql"
09:51.31Jennanope. not there
09:51.59jkroonok guys, just FYI, if anybody is reporting call drops, ignored registrations etc, get them to check that SIP connection tracking in the kernel is switched OFF.
09:52.00Jennahmm. although asterisk connects fine thru ODBC with the my db.
09:52.30JennaTommyBotten, my confs btw if ur interested http://pastebin.ca/1748056
09:54.28TommyBottenThus, module show like odbc works fine?
09:55.49eddyohttp://pastebin.ca/1748066
09:56.10TommyBottenYou can try "module load res_mysql"
09:56.33TommyBotteneddyo: Which distro are you running?
09:56.36eddyoTommyBotten can u tell me what could be the reason?
09:56.48TommyBotteneddyo: The reason is because you are missing a compiler
09:57.08eddyoi already installed gcc
09:57.12eddyoand its working fine
09:57.24JennaTommyBotten, yup I have res_odbc.so,  func_odbc.so & cdr_odbc.so
09:57.42TommyBotteneddyo: Which distro are you using? It might be that you need more than just 'gcc'
09:57.56TommyBottenJenna: ok. try to load the mysql module
09:58.29*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
09:58.29ChannelZeddyo: g++
09:58.31JennaTommyBotten,  there is no res_mysql,  but I do have res_config_mysql
09:59.48*** join/#asterisk mpe (n=mpe@217.74.220.92)
09:59.51TommyBottenJenna: Sounds right. Try that one
10:00.26eddyoCentOS release 5.4 (Final)
10:00.27eddyoKernel \r on an \m
10:00.46ChannelZinstall g++
10:01.35TommyBottenYou'll probably need to do "yum groupinstall 'Development Tools'"
10:01.51TommyBottengcc or g++ often leaves out tools like bison,m4 etc.
10:03.15eddyoChannelZ (K) ur the man
10:03.20eddyothx TommyBotten
10:03.23eddyoit worked
10:03.25*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
10:03.59ChannelZbed time
10:04.54TommyBottentake care
10:05.15eddyobye buddy
10:11.53*** join/#asterisk sjobeck (n=Adium@valdisere.sjobeck.com)
10:12.06*** part/#asterisk sjobeck (n=Adium@valdisere.sjobeck.com)
10:14.36*** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de)
10:16.32*** join/#asterisk mpe_ (n=mpe@217.74.220.92)
10:16.59*** part/#asterisk mpe_ (n=mpe@217.74.220.92)
10:17.39*** join/#asterisk mpe (n=mpe@217.74.220.92)
10:19.26*** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk)
10:19.29*** part/#asterisk mpe (n=mpe@217.74.220.92)
10:19.35shamelessn00bHi guys
10:19.58shamelessn00bwhenever I try to play this audio file in raw format it plays incredibly fast
10:20.12shamelessn00band if I convert it to 8k 8 bit wav
10:20.13*** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com)
10:20.17shamelessn00bit doesnt play at all
10:20.39shamelessn00band when I convert it to 8k 16 bit wav asterisk transcodes it to slin and plays it
10:21.15shamelessn00b8k 8 bit raw plays incredibly loud and fast
10:23.11*** join/#asterisk mpe (n=mpe@217.74.220.92)
10:23.57*** join/#asterisk mpe_ (n=mpe@217.74.220.92)
10:26.34*** join/#asterisk GNU\colossus (i=anonymou@truschnigg.info)
10:26.38GNU\colossushi all
10:29.58shamelessn00bhi
10:30.06*** join/#asterisk waa (n=waa@balrog.credipar.com.br)
10:36.57*** join/#asterisk Amorsen (n=Amorsen@94.127.50.7)
10:37.18AmorsenIs there a way to see the current setting of maxdatagram for t.38
10:38.53TommyBottenAmorsen: I remember reading about it in the 'administrators guide to fax' guide that digium released.
10:38.54*** join/#asterisk romain (n=romain@peerfuse.org)
10:39.36AmorsenThanks TommyBotten
10:41.32TommyBottenMy pleasure, Amorsen
10:45.50GNU\colossushow can I determine which codecs an IAX2 peer supports?
10:49.17AmorsenAh, a sip show peer foo shows it as   T.38 MaxDtgrm: 400
10:51.05*** join/#asterisk donatas (n=donatas@office.kis.lt)
10:51.58*** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net)
10:52.19donatasHi guys. Could anyone give me any information (url,pdf) about how to start with voip + analogical phone line
10:53.49shamelessn00bif I encode a file in u-law format
10:53.56shamelessn00basterisk doesnt play it
10:54.07shamelessn00b(using playback(path to file))
10:54.54shamelessn00bwhereas if I change the format to 8k 16bit it transcodes the file to slin and plays it on the channel
10:55.06AmorsenWhat do you call the file?
10:55.24Amorsenfoo.ulaw is the expected name I believe
11:00.01AmorsenI really dislike Asterisks extension sorting...
11:00.13Amorsen_4588334[0-5]XX comes before _4588334[2-5]X9
11:02.52*** join/#asterisk oej (n=olle@ns.webway.se)
11:07.55jkroondoes anybody know about any tool that'll either parse asterisk "sip debug" output and reconstruct the conversations, or some other sniffer-based tool that'll do so?
11:09.05jkroonbasically I'm looking for something that'll scan the SIP (and IAX/2) traffic for anomalies and report on them, and possible just log each individual conversation to a text file of sorts.
11:09.19*** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903)
11:12.04shamelessn00bAmorsen: the extension is test.wav
11:12.13shamelessn00bfilename rather
11:12.24shamelessn00bencoding is ulaw
11:13.19shamelessn00balso, if I set the bit rate of a wav file to 8 (without tinkering with the encoding) it plays very fast
11:13.37shamelessn00bbits per sample
11:13.38shamelessn00b*
11:13.48shamelessn00bif I change the bits per sample to 9
11:13.53shamelessn00bit plays normally
11:14.01shamelessn00bwhy is that
11:15.22TommyBottendonatas: You should read 'the book'
11:15.24TommyBotten~book
11:15.25infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
11:17.41*** join/#asterisk RicWright (n=Ricwrigh@firewall.a-n-t.net)
11:18.15*** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it)
11:20.14shamelessn00bTommyBotten: is there a way to add effects in realtime to SIP/DAHDI calls like changing voice etc
11:20.17shamelessn00bhttp://lobstertech.com/2005/oct/31/asterisk_voice_changer/
11:20.22shamelessn00blike this application does
11:21.03TommyBottenshamelessn00b: I've seen a few of those applications around, but never used them myself.
11:21.33TommyBottenshamelessn00b: Whats wrong with the one you pasted?
11:21.52shamelessn00bnothing really
11:22.06shamelessn00bI'd like to see if there are others available
11:22.29shamelessn00bwith more effects
11:22.30shamelessn00betc
11:23.21TommyBottenAh... I know there are a few... but I can only suggest voip-info.org or google
11:23.34donatasTommyBotten: what book ? :)
11:23.41TommyBottenWhat kind of 'effects' are you referring to?
11:23.48TommyBotten~book
11:23.49infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
11:23.56TommyBottendonatas: As seen above
11:25.24shamelessn00bTommyBotten: effects that you can apply to normal audio files
11:25.30shamelessn00blike sox has many effects
11:25.36shamelessn00bphaser
11:25.37shamelessn00breverb
11:25.40shamelessn00bstuff like that
11:25.44GNU\colossushow can I determine which codecs an IAX2 peer supports?
11:26.19TommyBottenGNU\colossus: Make a call, watch the SIP invite header and response
11:26.57TommyBottenshamelessn00b: Hmm... That sounds like something that would add a bit to latency
11:27.05GNU\colossusTommyBotten: how can I get at the header? can * do display it for me, ot will I have to sniff UDP traffic?
11:27.11GNU\colossuss/ot/or
11:27.18GNU\colossus(sorry, I'm a total asterisk newbie :))
11:27.26shamelessn00bTommyBotten: yeah but its worth trying
11:27.43shamelessn00bwould be cool
11:28.00TommyBottenGNU\colossus: Asterisk can display it for you. "sip set debug" and then add either peer or the IP
11:28.26GNU\colossusthanks, I'll try that right now :)
11:28.47TommyBottenYou're welcome
11:28.57GNU\colossusTommyBotten: ah, hrm, does that work for IAX2, too?
11:29.32GNU\colossusidk if iax2 also transmits a SIP header, but I'd guess it doesn't? :)
11:29.33TommyBottenGNU\colossus: Ah... debug does, but I'm not sure how the header/response looks in IAX2
11:29.46GNU\colossusI see. thanks very much nevertheless!
11:30.10TommyBottenGNU\colossus: Sure... but be sure to try 'iax2 set debug' ... whatever.
11:30.13TommyBottenIt might help you out
11:30.23TommyBottenand if it does, please tell :)
11:30.51GNU\colossuswill debug messages go to the asterisk console, or to syslog(), or somewhere else entirely?
11:31.16TommyBottento the asterisk console
11:31.34GNU\colossusyeah, I'm already viewing it :)
11:32.11RicWrightHi All, sorry for the noob question, ive just installed the Asterisknow program, ive managed to web browse to the ip of the machine the software is running on, however its not taking my password when i click "freepbx admin" is this the same password as the root? as the setup only asked for one password
11:32.53TommyBottenRicWright: Probably not. But you should try #freepbx ,as it is focused on the GUI part. This channel is mainly asterisk core :)
11:33.04RicWrightah sorry thanks :)
11:33.10TommyBottenno worries :)
11:34.00GNU\colossusTommyBotten: well, at least there's no human-readable codec (list) obvious to me
11:35.07TommyBottenGNU\colossus: I'm not surprised :/
11:36.28GNU\colossusbtw, my console sometimes displays warnings like these, even when no activity is going on: "frame.c:1289 ast_parse_allow_disallow: Cannot allow unknown format 'libc'" - I guess that's a typo ("ilbc" -> "libc"), but I can't find the string "libc" anywhere in our configs.
11:36.35GNU\colossusany ideas what could be wrong there?
11:37.06*** join/#asterisk Jenna (n=JJ@unaffiliated/jenna)
11:37.51TommyBottenThat must be a typo, yes
11:38.01TommyBottenAre you using ilbc?
11:38.22GNU\colossusnot that I'd know.
11:38.39GNU\colossusI never managed * before, and took over just yesterday though ;)
11:38.47TommyBottenAha :)
11:39.26GNU\colossuswe're using alaw exclusively to communicate with out remote IAX" peer, from what I've seen so far
11:39.35TommyBottenWell, it seems like the typo is in the config files
11:39.56TommyBottencan you try a 'grep -ri libc /etc/asterisk/*' ?
11:40.08GNU\colossusthe config files don't contain that string though.
11:40.12TommyBottenhmm.. ok
11:40.20TommyBottenStrange
11:40.48TommyBottenBecause it seems like it should be in sip.conf on an 'allow' line
11:41.22GNU\colossusis there any other, non-file possibility for a config statement of that kind to appear?
11:41.31GNU\colossus(in a RDBMS or some such)
11:41.57TommyBottenSure there is. Check for res_mysql or res_config_mysql
11:42.05TommyBottenAns see wether it has been enabled and is in use
11:42.19TommyBottenyou can also do 'show module like mysql' from the command line
11:42.50GNU\colossusres_mysql.conf contains what seems like a database configuration for one of our mysql hosts.
11:43.03GNU\colossushow would I check to see if it's enabled in *?
11:43.19TommyBotten'module show like mysql'
11:44.15TommyBottenIf it is loaded, then check extconfig.conf to see what asterisk loads first. file, database, odbc etc.
11:44.20GNU\colossus3 modules loaded
11:44.25GNU\colossususe count is 0 for all 3 of them
11:45.02GNU\colossussipusers => mysql,asterisk,sipuser
11:45.06GNU\colossusis a line in extconfig
11:45.11GNU\colossusand there are 3 more like that
11:45.17GNU\colossusso I guess it's enabled ;)
11:45.50TommyBottenYep :)
11:46.15TommyBottenIf you see res_config_mysql.conf ...or something similar, you can see which database and table(s) it uses
11:46.19GNU\colossusI guess I can work this out on my own from here. thank you very much for your kind and very competent help! :)
11:46.54TommyBottenYou're welcome.. And good luck :)
11:48.14donatasWhat do I need if I want to connect my analogic phone line to my asterisk?
11:48.19*** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
11:49.40TommyBottendonatas: You'll need a hardware extension card. If you need it connected to the analog line you need a 'FXO'-card. If you need a analog device such as a phone connected to your asterisk, you need a 'FXS'-card.
11:50.08TommyBottendonatas: Digium and Sangoma are both cards that I have tried and found working pretty good.
11:50.50JennaTommyBotten, thanx reloading res_config_mysql did the trick
11:51.14TommyBottenJenna: Glad to hear it
11:52.08Jennabut one problem. asterisk -c mode keeps on scrolling forever ... despite the verbosity turned off. In the end I have to kill it (the process)
11:52.41GNU\colossusTommyBotten: I found it ;)
11:53.58TommyBottenJenna: Why don't you just run it in the background? ... and attach whenever you'd like?
11:54.25donatasTommyBotten: And how about voice gateway ? What function do they do ?
11:54.26Jennabut Im afraid it would fill up my asterisk.log in no time
11:54.50TommyBottenJenna: What kind of information are you seeing?
11:55.21*** join/#asterisk TimeRider (n=steve@78.32.26.1)
11:55.22TommyBottenJenna: There are verbose messages and debug information
11:55.26JennaTommyBotten, [Jan  8 09:41:03] NOTICE[4288]: chan_sip.c:16257 sip_poke_noanswer: Peer '7777' is now UNREACHABLE!  Last qualify: 3
11:55.48TommyBottenJenna: That should be avoided by 'core set verbose 0'
11:56.03JennaI have run asterisk -c  (note there is not vvv switch
11:56.18TommyBottenJenna: And you can always use logrotation ... or if you really really want it, set /dev/null as your logfile :D
11:56.38TommyBottenJenna: The default verbosity level is 3
11:57.20JennaTommyBotten, :) that did cross my mind (log rotate) but the behavior seemed erractic.
11:57.46Jennabut now that u've mentioned that the default verbosity level is 3 then I guess the behaviour is predictable
11:58.05TommyBottenJenna: Absolutely. Logrotation and /dev/null are cures for symptomes ... not the real problem
11:58.12TommyBottenYes
11:59.28tzafrirJenna, why do you run asterisk -c ?
12:00.18Jennatzafrir, just to check up the things.
12:01.01JennaTommyBotten, nope. even if I /etc/init.d/asterisk start it its screems out the message. leaving me with no option but to kill it.
12:02.01TommyBottenJenna: Did you set the verbosity level to zero?
12:02.27Jennayup I did. but no luck
12:02.33TommyBottenHmm... strange
12:02.55TommyBottenah... there we are
12:03.01TommyBottencheck out your 'logger.conf'
12:03.11TommyBottenit says something like 'console => notice,warning,error
12:03.19Jennanow when I run  asterisk in the background (init.d/asterisk start..)  & when I tailf /var/log/asterisk/messages its the same dreaded message filling up the log/screen
12:03.20TommyBottenJust remove notice from it
12:03.40TommyBottenAnd in that case, remove it from the line which says 'messages' as well
12:06.22*** join/#asterisk coppice (n=chatzill@202.64.176.25)
12:07.39JennaTommyBotten, yeah that did the trick
12:08.13Jennabut now "sip show peers" does not display the registred 7777 user
12:08.35TommyBottensip show peer 7777
12:08.37TommyBottenWhat does it say?
12:08.42*** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk)
12:09.06JennaPeer 7777 not found.
12:09.28TommyBotten.... really?
12:09.42TommyBottenHumm
12:09.51Jennayup. not user
12:09.56TommyBottenThat should not affect it.
12:10.09Jennano sip user i.e.
12:10.11TommyBottenIs it defined in mysql?
12:10.59Jennayup the end client ekiga does say the it has got registred
12:11.16TommyBottenhumm
12:11.18TommyBottenthat is strange
12:12.00*** join/#asterisk sergee (n=serg@voip1.west-call.com)
12:12.35Jennayup weird
12:13.24*** join/#asterisk smooth_penguin (n=smoove@59.95.46.245)
12:13.31TommyBottenTry reloading res_mysql
12:22.09somanHi,  I am using asterisk 1.6 with TE121 card installed and connected to the PRI cable.  When i have started asterisk, i am getting the following error  " chan_dahdi.c:11826 pri_dchannel: PRI Error on span 0: We think we're the network, but they think they're the network, too."
12:22.52*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
12:22.59somanand when i changed the signalling to "pri_cpe" , then it says "we think we are the cpe, but they think they are the cpe too"
12:24.15TommyBottensoman: And the card is not looped? :p
12:25.43somanTommyBotten: How to check that, i am sorry not aware of it
12:26.17somanTommyBotten: http://pastebin.com/m7f746b89    here is my chan_dahdi.conf and /etc/dahdi/system.conf
12:26.52TommyBottensoman: If it is a dual port card, and you plug it back onto itself, you could get that behaviour
12:27.05TommyBottenI was mostly meant as a (bad) joke
12:27.21somanCan you tell me what could be the problem
12:27.30tzafrirsoman, what do you connect to?
12:27.35tzafrirPSTN? some device?
12:27.51soman<tzafrir>: PSTN
12:28.24tzafrirso you should have:  signalling=pri_cpe
12:28.49tzafrirand then run  'dahdi restart'
12:29.02somanok, then i am getting "we think we are cpe, but they think they are cpe too"
12:29.08somanok, let me check
12:29.12tzafrirso you do have a loop
12:30.09JennaTommyBotten, hmm weird  WARNING[4625]: cdr_addon_mysql.c:316 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf
12:30.31soman<tzafrir>: ok let me restart dahdi
12:30.44*** join/#asterisk v00tace (n=v00tace@p50989ed2.dip0.t-ipconnect.de)
12:32.49soman<tzafrir>: I stopped asterisk, restarted dahdi,
12:33.10soman<tzafrir>:  and now i am getting the error "chan_dahdi.c:11826 pri_dchannel: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too."
12:33.54tzafrirsoman, so you have a loop somewhere along the way to your provider
12:34.08TommyBottenJenna: It's "just" CDRs for mysql. Are you using it?
12:34.30soman<tzafrir>:  so how to solve that issue, what should i do now?
12:34.31Jennayeah
12:34.51JennaTommyBotten, but it was fine couple of hours ago
12:36.18soman<tzafrir>:  You mean that it is the problem with the PRI connection that i have?
12:36.18TommyBottenJenna: Is cdr_mysql.conf available?
12:36.20*** join/#asterisk khronos (i=tom@sip02.kineticvoip.com)
12:36.25JennaTommyBotten, my configuration resembles this one but only Im on debian (http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation#Setuprealtime)
12:36.45tzafrirsoman, one device along the way is in loop mode?
12:37.02TommyBottenJenna: Just do a 'ls /etc/asterisk/cdr_mysql.conf'
12:38.26Jennabut Im using cdr_odbc thing
12:38.47TommyBottenhmmm
12:38.55TommyBottenin that case, just unload cdr_mysql
12:42.49shamelessn00btzafrir: there??
12:44.41soman<tzafrir>:  how to test whether the device is in loop mode?
12:44.48somanI dont have any ideas
12:45.04tzafrirshamelessn00b, basically, yes
12:45.10shamelessn00b:) hi
12:45.25shamelessn00bneeded to ask a few questions on how asterisk plays files
12:46.13shamelessn00bI was doing some file conversions, need to improve my knowledge on whats the best file format to be played back on asterisk
12:46.24shamelessn00busing sox to convert wav files
12:48.39TommyBottenshamelessn00b: You can use asterisk to convert files aswell
12:48.45*** join/#asterisk orn (n=orn@office.sip.is)
12:48.55TommyBottenshamelessn00b: I've been having a bit of trouble with sox` compatability
12:49.08shamelessn00bthats transcoding, I want to avoid that
12:49.29TommyBottenAs in... you can do it once
12:49.30shamelessn00balso, we record in 44k 16 bit uncompressed wav format
12:49.33TommyBottenand store the file
12:50.07shamelessn00bcan you convert to mono
12:50.24shamelessn00bor apply noise reduction/normalization filters
12:50.32shamelessn00b?
12:51.11*** join/#asterisk Jenna (n=JJ@unaffiliated/jenna)
12:52.20soman<tzafrir>:  I had, "t1e1override=1" option set in /etc/modprobe.d/dahdi file
12:52.50TommyBottenshamelessn00b: mono, yes. But no filters as I know of
12:53.13soman<tzafrir>:  when i commented line, now i am not getting the previous error,   but getting the new error that "Unable to register channel 1-15,17-31"
12:53.43shamelessn00bwell, whenever I convert from sox and I specify bitrate, any value < 9 makes the audio file play to fast in asterisk
12:54.03shamelessn00bmy question is normally people use 8k 16 bit files to be played in asterisk
12:54.16shamelessn00bbut on phones channel capacity is restricted to 64k
12:55.11shamelessn00band 8k 16 bit is 128k
12:55.33shamelessn00bso asterisk is implicitly doing something that degrades the audio quality
12:55.44coppicebut the phone channel carries compressed data
12:56.05shamelessn00bcoppice means??
12:56.21shamelessn00bisnt asterisk playing back in RAW
12:56.23shamelessn00b?
12:56.34shamelessn00bwhich is uncompressed
12:56.43tzafrirsoman, because it has an incorrect signalling
12:56.45coppiceyour 128k signal becomes 64k after alaw or ulaw compression
12:56.54tzafrirThe default signalling is pointless for PRI
12:57.17shamelessn00bwhy cant I give my asterisk a ulaw file and it plays it back as such
12:57.24shamelessn00bwithout an compression
12:57.27tzafrirshamelessn00b, you can
12:57.43tzafrirThough wav -> ulaw conversion is cheap enough
12:57.43shamelessn00bwhenever I try, it gives me error messages
12:57.50coppiceyou can, but it doesn't buy you much
12:57.57soman<tzafrir>:  actually the jumper is off on the device, and more over i have removed the T1E1override option...
12:58.02shamelessn00bnot in terms of processing
12:58.07tzafrirA wav file has a header which makes it much simpler to play by other programs
12:58.10somanCan you tell me what i have to do now
12:58.25shamelessn00bI mean if I give it a converted file myself I would know how exactly its going to sound like
12:58.56tzafrirshamelessn00b, 'core show file formats'
12:59.49shamelessn00bhttp://pastebin.com/m7ae4bd9a
13:00.32shamelessn00bI had this wierd unexplanable observation
13:00.51shamelessn00bI converted a wav file and set bits per sample to 8 and sample rate to 8k
13:01.03shamelessn00basterisk played it too fast
13:01.07soman<tzafrir>:  but if I enable that option, i am getting the previous error again
13:01.12shamelessn00bthen I increased the bits per sample to 9
13:01.17shamelessn00bthe file played ok
13:02.00shamelessn00band if I convert it to u-law instead of the default encoding it doesnt play at all
13:04.11shamelessn00bu-law is 8k 8 bit
13:04.36tzafrirHow do you play it?
13:05.13*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
13:08.05*** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net)
13:08.06SuPrSluG8k 16 bit
13:09.05shamelessn00btzafrir: using the application Playback()
13:09.28*** join/#asterisk catojo (n=catojo@189.24.66.25)
13:09.37tzafrirand you get garbage for audio?
13:10.10shamelessn00bSuPrSluG: im writing sox foo.wav -r 8000 -e u-law -c 1 -t raw bar
13:10.25*** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163)
13:10.36shamelessn00band the output file it gives me is 8k 8 bit
13:11.06shamelessn00btzafrir: if I set the bits per sample to any integer less than 9
13:11.11shamelessn00bit plays very fast
13:11.20shamelessn00band if I convert to u-law
13:11.25shamelessn00bit doesnt play at all
13:11.52tzafrirshamelessn00b, Asterisk identifies the type of the file by its extension.
13:12.25shamelessn00bsox foo.wav -r 8000 -e u-law -c 1 -t raw bar.raw
13:12.29shamelessn00bI tried this aswell
13:13.06tzafrirIf the file is not .au it will not be considered a raw ulaw file
13:13.09shamelessn00bsox foo.wav -r 8000 -e u-law -c 1  bar.wav (and this too)
13:13.28shamelessn00boh
13:13.32shamelessn00blemme try au
13:13.32tzafrir(which is odd, as Australia has A-Law)
13:13.38shamelessn00blol
13:14.15shamelessn00band why doesnt it play files with bit rate of < 9
13:14.25shamelessn00b?
13:15.33coppicewhat do you mean by setting the bit rate?
13:17.02shamelessn00bbits per sample sorry
13:17.30coppicedoesn't sox only let you set the bits per sample in multiples of 8?
13:17.50coppicebits per sample is meaningless for ulaw, anyway
13:18.55shamelessn00bcoppice: you can even use values like 11.5
13:19.17shamelessn00btzafrir: no the extension .au isnt working
13:20.09shamelessn00bI wrote sox -S foo.wav -t raw -e u-law bar.au vol 0.94 rate -s -a 8k remix -
13:20.38tzafrirshamelessn00b, for reference: http://downloads.asterisk.org/pub/telephony/sounds/
13:20.46shamelessn00bthen when I play this file is asterisk
13:21.05tzafrirsome alaw and ulaw there as well
13:21.20shamelessn00bhttp://pastebin.com/m7c4ab6b5
13:21.52shamelessn00bok I'll try playing those files, maybe I'm converting incorrectly
13:27.52soman<tzafrir>:  Do i have to contact the PSTN provider to check if there is any interface loops running on their switch
13:28.16tzafrirsoman, also worth testing
13:28.43tzafrirwonders if there aren't other people more exepirnced with such hardware and lines here
13:29.17shamelessn00bIm using sangoma
13:29.28soman<tzafrir>:  Is there any  other way to check for it
13:29.31shamelessn00band you can check using wanpipemon tool
13:30.18shamelessn00bdunno, Im making a wild guess, try ifconfig and monitor the sent recieved packets
13:30.23shamelessn00bif its looped
13:30.27shamelessn00band you play a file
13:30.44shamelessn00byour recieved packets would increase
13:32.06*** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
13:37.19*** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net)
13:40.04*** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it)
13:42.09*** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk)
13:50.59*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
13:55.53ManxPower-worktzafrir: I'm sure there is. 8-|
13:58.44*** join/#asterisk Akiraa (n=Akiraaaa@79.112.10.125)
13:59.45*** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org)
14:00.23shamelessn00bhi ManxPower-work
14:01.53*** join/#asterisk etfonhomey (n=etfonhom@ip-64-32-192-35.iad.megapath.net)
14:06.02*** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org)
14:12.08*** join/#asterisk hluesea (n=hulusika@88.247.127.66)
14:12.55*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
14:18.10*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
14:20.42*** join/#asterisk Tagor (n=none@s55928c6d.adsl.wanadoo.nl)
14:22.34TagorIs it possible to call from a software phone to an asterisk server?
14:23.23TagorI mean that the software phone doesn't login to the asterisk server
14:23.37tzafrirTagor, yes
14:23.48TagorBut instead of calling a land line number, you enter the ip address/internal number
14:27.21creativxthats what voip was meant for ;)
14:27.28creativxone of the things atleast..
14:29.19*** join/#asterisk ruyo (n=psantos@195.23.253.223)
14:32.29TagorAnd how can I do this?
14:32.59*** join/#asterisk ltd (n=z@pat.transact.net.au)
14:33.50[TK]D-FenderTagor: "allowguest=yes" and "context=whatever" under [general] and put the extens you want anonymous people to be able to call there
14:35.52*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
14:38.35TagorThanks [TK]D-Fender :)
14:39.36[TK]D-FenderTagor: You're welcome
14:44.56*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:51.28*** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk)
15:03.31*** join/#asterisk etnos (n=etnos@adsl-2-204-26.mia.bellsouth.net)
15:03.51*** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek)
15:03.59*** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek)
15:06.14*** join/#asterisk viperdude (n=jon@195.74.96.122)
15:08.34*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
15:09.30benngardcan any1 help me and see any errors here: exten => s,n,ExecIf($[${CALLERID(num)}=739709025]?Set(CALLERID(all)=Joar Benngard<0739709025>))
15:11.07ManxPower-workbenngard: the only thing I can see is maybe you are using a version of Asterisk that works slightly different for ExecIf (I seem to recall some changes to it at some point.  See "core show application execif" to be sure.
15:11.39ManxPower-workOh, and NEVER EVER put the leading 0 or 00 on the Caller*ID number.  It's not part of the callerid
15:12.55[TK]D-FenderManxPower-work: says who?
15:13.49*** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com)
15:14.57*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
15:15.37benngardthe execif works in 95% of all cases but somteimes i got weired error messages in cli
15:15.54benngardand why shouldnt i put the leading zero?
15:16.20[TK]D-Fenderbenngard: You're perfectly fine to do so.
15:16.30[TK]D-Fenderbenngard: and the command can error out if the CID is BLANK
15:16.31benngardthx
15:16.33ManxPower-work[TK]D-Fender: Me!
15:16.48benngardhidden number?
15:17.05ManxPower-workbenngard: I suspect sometimes the callerid is empty so you would be comparing nothing to the number.  Try:
15:17.24ManxPower-workexten => s,n,ExecIf($["${CALLERID(num)}"="739709025"]?Set(CALLERID(all)=Joar Benngard<739709025>))
15:17.31*** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org)
15:17.41ManxPower-workif you wrap one side of the = in quotes you MUST wrap the other side in quotes.
15:18.10benngardthx, didnt know that
15:18.23benngardis still learning
15:18.54ManxPower-workbenngard: In most place in Asterisk quotes are literal.
15:19.40ManxPower-workSo if you had $[${FRED} = "BOB"] it would not work because FRED might equal BOB, but BOB != "BOB"
15:20.25*** join/#asterisk Heretic (n=fallen@dsl-246-89-133.telkomadsl.co.za)
15:24.38angryuser_when i have 70 persons on hold in queue the music on hold os ok, when i got more 100, it is crappy, i have asterisk 1.4.27 and no timer installed, do i need a timer for that ?
15:24.40somanHi, Can anybody help me how to get rid of the error " chan_dahdi.c:11826 pri_dchannel: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too."
15:25.05somanI am using asterisk 1.6 with TE121 single span card installed
15:25.07*** join/#asterisk ibolmo (n=ibolmo@cpe-67-11-89-134.rgv.res.rr.com)
15:25.43mockerwoo, bandwidth.com numbers are provisioned.
15:26.20jayteesoman, where is the span getting it's timing from? the telco? pastebin your /etc/dahdi/system.conf file and your /etc/asterisk/chan_dahdi.conf file.
15:26.21tamielsoman: "signalling" in chan_dahdi.conf for this span ?
15:26.31jaytee~pb
15:26.32infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
15:26.38[TK]D-Fendersoman: Same answer we told you yesterday... stop being CPE, or tell them to
15:26.50[TK]D-Fendersoman: And you never answered what the hell youplugged your card into
15:27.10angryuser_do i need a timer source to have a good music on hold quality ?
15:27.13jayteebows out after realizing this is an old issue
15:28.19beekmorning jaytee, [TK]D-Fender
15:28.20somanJaytee: tamiel: http://pastebin.com/m19b131a3  here is my confi
15:28.40*** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel)
15:28.41somanjaytee: yes, span is getting the timing from telco
15:29.14*** join/#asterisk Zarnivoop (n=stmo@custchecker.rixport80.se)
15:29.24somantamiel: signalling is pri_cpe
15:30.18Zarnivoopis it possible to make the applicationmap in features.conf work for the callee? I only get it to work for the caller.
15:30.38soman[TK]D-Fender>:  even i tried with with signalling=pri_net,  but then i am getting the error chan_dahdi.c:11826 pri_dchannel: PRI Error on span 0: We think we're the network, but they think they're the network too."
15:30.53[TK]D-FenderZarnivoop: either they don't dial it fast enough, or DTMF mode isn't right on their side
15:31.08[TK]D-Fendersoman: And you still aren't answering my question.
15:31.38Zarnivoop[TK]D-Fender: I have tried all possible combinations, so I do not think that is the issue. unless you have tried it successfully?
15:31.46soman[TK]D-Fender>:  I have a PRI connection from telco,
15:31.48Kattyhi
15:31.50*** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net)
15:32.05somanPlugged the PRI cable in to the card
15:32.23ManxPower-worksoman: if you tried both pri_cpe ad pri_net then your telco has a loopback on your PRI.  This is a classic symptom
15:32.33*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
15:32.41ManxPower-workContact your telco tell them to take the loop off
15:32.51[TK]D-FenderOften a simple button to press on the SmartJack to disable
15:33.15soman<ManxPower-work>:  Thanks for the input.. i will try that..
15:33.54soman[TK]D-Fender>: that sounds good but how to do it, I dont find any button
15:34.21[TK]D-Fendersoman: I didn't say always, find a clue, then find your telco's # and call them
15:34.22ManxPower-worksoman: it would depend on the model of the smartjack.  I've never seen one with customer accessable buttons, but I know they exist.
15:34.53[TK]D-Fendersoman: Also make sure you're using a straight-through cable
15:35.22soman[TK]D-Fender>: ok,
15:35.31soman[TK]D-Fender>: Thanks for the inputs.
15:36.07soman[TK]D-Fender>:  <ManxPower-work>: thanks for support.. will stay in touch
15:40.03Kattyhi?
15:40.15[TK]D-FenderKatty: Mew.
15:40.18Katty:>
15:41.05beekHellooooo, Katty!
15:41.16Kattyhi beek
15:43.17Zarnivoop[TK]D-Fender: found the features.conf issue; the DYNAMIC_FEATURES variable needs to be set on callee channel (by means of inheriting it by underscore'ing the variable).
15:43.55Zarnivoopsupposedly fixed by automatic inheritance in later asterisk versions, also
15:43.59[TK]D-FenderZarnivoop: Makes no sense.....
15:44.13[TK]D-FenderZarnivoop: pastebin your features.conf  and dialplan for this
15:44.38Zarnivoop[TK]D-Fender: I'll just paste you the issue url: https://issues.asterisk.org/view.php?id=14657
15:45.51[TK]D-FenderZarnivoop: OK, if you were on an older version of * before the fix then fine...
15:46.04Zarnivoop[TK]D-Fender: yup, I am
15:47.16[TK]D-FenderZarnivoop: Glad you ID'd it then
15:49.13*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:50.37kc8pxyi'm trying to hook my asterisk server into my vonage. the newest howto i can find is 2006. is that really the newest? if not, where do i find current info on it?
15:51.30*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:51.53[TK]D-Fenderkc8pxy: Does it not work?
15:53.28ManxPower-workkc8pxy: It seems that most Asterisk peope decide it is better to switch away from Vonage in order to save money.
15:53.30ManxPower-workpeople
15:54.59*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
15:54.59*** mode/#asterisk [+o putnopvut] by ChanServ
15:55.01kc8pxy[TK]D-Fender: I'm trying to formulate,  then test a config for a friend. they currently have 10 vonage lines and use the heck out of the unlimited did/dod.
15:55.09kc8pxyminutes that is
15:56.07[TK]D-Fenderkc8pxy: Maybe you should try the instructions and see for yourself
15:56.43ManxPower-workkc8pxy: how many mins/month?
15:56.43kc8pxyManxPower-work: understood. know a good provider that could beat vonage's price for unlimited DID/DOD minutes? the manager is a friend, but she's all about the bottom line.
15:56.48*** join/#asterisk davidcsi (n=davidcsi@89.141.97.190.dyn.user.ono.com)
15:57.07*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
15:57.59ManxPower-workkc8pxy: you will never know the answer to that until you tell me how many mins/month you use.
15:58.23kc8pxyManxPower-work: calculating.
15:58.28davidcsihello, I want to have multiple moh while ringing b-number. I'm using "m" on the dial command, but if i have many users and each want a different moh, can this be done? how would i do this in streaming? is it possible to request a dynamic file in streaming?
16:00.28ManxPower-workdavidcsi: that is a STANDARD feature of Asterisk.  re-read the docs.
16:00.31ManxPower-work~answers
16:00.32infobothmm... answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
16:01.36*** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org)
16:03.00kc8pxyManxPower-work: i'm trying to estimate a LITTLE high for now, because i know they would hate to run out of minutes. but it's not terribly high..  12k/line/mo
16:03.12davidcsiManxPower-work: I'm already using "m" option on the dial command, and i know streaming can be done, but i've seen nowhere hos to actually request streaming dynamically, i.e. how to pass the argument of the filename to request via streaming, only static streaming
16:03.15*** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk)
16:05.00kc8pxygar..   time-critical errand i need to run.
16:05.05ManxPower-workdavidcsi: You would not use streaming, you would use different MoH classes
16:05.09kc8pxybbl
16:05.28davidcsiManxPower, so it can not be done?
16:05.33*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
16:05.48davidcsivia streaming, i mean
16:06.13ManxPower-workdavidcsi: in theory, but it would be unreliable, hard to set up and I'm not going to help you with streaming
16:06.42ManxPower-workRegardless of if you do streaming, you'll have to set up different MoH classes
16:06.50davidcsiso how many classes is it possible to have? unlimited?
16:07.06ManxPower-workdavidcsi: I don't know.  I assume at least 64.
16:07.16kc8pxydavidcsi: memory is the limit??
16:08.12ManxPower-workkc8pxy: 12,000 mins * 1.2/cents/min = $144/month.  Do you pay less than that for your "unlimited" service?
16:08.56kc8pxyManxPower-work: 25$/mo
16:09.21Kobazjust make sure they don't cut you off for excessive usage
16:09.31ManxPower-workKobaz: I'm sure they will eventually
16:09.48ManxPower-workThere is no such thing as real "unlimited" non-local phone service.  All carriers have a cap.
16:12.18kc8pxyManxPower-work:  that estimate is rather high,  but it would not be good to estimate less than half that, and that's still much higher.
16:12.35benngardnice problem i ran into, when for example 0317998976 calls my extension 0317998975 i want both my sip-phone to ring and my mobile, np but when i send then call to my mobile, i need to strip the first zero from a-number...
16:12.41viperdudeManxPower-work: well on-net services can be without charge and unlimited and non-local so the lines are a bit blurred there days
16:13.01kc8pxyManxPower-work: since the carrier is vonage,  perhaps they are using the law of large numbers to balance out the money.
16:13.14ManxPower-workVonage must have a very high cap (Teliax has softcap of 1,500 mins/month residential, 2500 mins/month business, for example) on their unlimited account.  Either that or they will disconnect you.
16:14.03bmoracaviperdude, i've never seen a company charge nothing for on-net...most of the time, they'll charge whatever the minimum rate is for any of their providers
16:14.23bmoracaManxPower-work, vonage has a 3000 minute soft-cap...it's in their TOS
16:14.58[TK]D-FenderMust be pretty doughy by noe....
16:15.02[TK]D-Fendernow*
16:15.07viperdudebmoraca: on-net is usually free and unlimited on VoIP.. if I saw a ITSP who had fair usage on it they would not get my business
16:15.25viperdudevonage has soft cap for on-net?
16:15.41bmoracathey have a softcap, period.
16:15.50viperdudelol that is pants
16:16.06viperdudethat just in US or globally?
16:17.00bmoracai don't know...but looking again, i see they raised it to 5000 at some point
16:17.03bmoracaused to be 3000
16:17.22bmoracai don't use vonage
16:17.59*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
16:19.36viperdudei wouldn't touch vonage with a 10ft barge pole, even if I wasn't CTO of a ITSP.
16:22.24*** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
16:23.45*** join/#asterisk QubeZ (n=qube@64.128.254.34)
16:23.48QubeZhello all
16:24.06QubeZwhat is the diff between active call and active channel? i'd assume it should be one in the same but my graphs are differing.
16:24.33ManxPower-workQubeZ: An active channel can be anything including a SIP registration
16:24.47QubeZahh ok
16:29.01*** join/#asterisk CunningPike (n=CunningP@204.239.8.157)
16:30.31leifmadsena call is a bridge between two channels
16:31.02leifmadsenphone(1) ---channel---> asterisk ---channel---> phone(2)
16:33.26RicWrightso a call forward in effect?
16:34.17Kobazwhat?
16:34.21*** join/#asterisk moy (n=moy@189.162.161.141)
16:34.33NuggetShaniqua don't live here no more.
16:34.33RicWrightsorry wrong windows :(
16:35.31*** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com)
16:37.02*** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr)
16:38.48*** join/#asterisk vally (n=kvirc@217.243.245.34)
16:41.58*** join/#asterisk rocksfrow (n=kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net)
16:45.11*** join/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com)
16:47.10titterCan someone help me with app_fax ... I get the call to go through, and it seems like it is trying to transmit the file, but I get this error
16:47.12titterhttp://pastebin.com/m71ea95ca
16:48.32*** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de)
16:49.43*** join/#asterisk HenrikJott (n=info@d83-183-134-141.cust.tele2.se)
16:50.35*** part/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com)
16:51.10HenrikJotthi everyone! I have a problem. When one of the agents in my queue hangs up a call, every agent waiting in the queue hears a beep sound except the agent who should get it. Does anyone know anything about this?
16:51.53*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
16:52.39*** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br)
16:53.49*** join/#asterisk hohum (i=dcorbe@apollo.corbe.net)
16:54.10HenrikJottnobody knows about beep sound playing on hangup?
16:55.28*** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
16:55.43*** part/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
16:56.41*** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
16:56.58*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-pndpwkoeqfztxgdv)
16:56.58NaikrovekHenrikJott: just be patient
16:57.15Corydon76-digfish-bulb: !!!
16:57.17Naikrovekit is disturbingly slow in here right now; when people start paying attention again i'm sure you'll get an answer
16:58.37HenrikJottNaikrovek: sorry =) i guess i was a bit unpatient there! it´s just that i´m going nuts with this problem, hehe
16:58.45Corydon76-digOr nobody has heard of that happening
16:58.49Naikrovekyeah
16:58.57Naikrovekoften times people don't answer because they don't know how to help
16:59.02Naikrovekbut stick around
16:59.09HenrikJottNaikrovek: I will!
16:59.15titterjust find a big hammer
16:59.20kc8pxyleifmadsen: i have a weird idea for that model. i want my asterisk box to grab my vonage account, and then feed it back out to the vonage ATA. (i'm trying use some of asterisk's features, w/o having to buy a new ATA/FXO.)
17:05.07leifmadsenkc8pxy: I'm not sure what you mean by that, but have fun :)
17:05.23*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
17:05.23*** mode/#asterisk [+o Deeewayne] by ChanServ
17:05.29keith4does anyone happen to have a ruler and an AEX800 laying around?
17:05.31leifmadsenthat "model" is just a standard incoming channel / outgoing channel that is bridged together to form a call
17:05.53Kobazkc8pxy: buy the vonage business account, and you get sip access, without needing an ata
17:08.28tzafrirhas a ruler
17:10.20Naikroveki'm such a newb.  call parking is nice and i just discovered it
17:10.33p3nguinoh
17:10.37p3nguinAll this time...
17:11.15Naikrovekwell we've gotten to a point now where i can't trust that person X will not be at a phone that isn't theirs
17:11.25Naikrovekphrased weirdly...
17:11.43Naikrovekwe've gotten to a point where i can't trust that a person will be at their own extension
17:12.05p3nguinyeah
17:12.09Naikrovekin fact multiple people work away from their extensions now, in a control room with a dozen desks.  anyone could be at any desk
17:12.14Naikrovekso i park and page the room
17:12.16p3nguinPark the caller, then go searching for person X.
17:12.28p3nguin"Your caller is on 708"
17:12.31Naikrovekyup
17:12.45Naikrovekasterisk is nice
17:13.31titterasterisk is amzaing.
17:13.48Naikrovekif it were written in java i could really contribute
17:13.54Naikrovekas i don't know C
17:14.12Naikrovekwhich would require that java realtime be better than it was last time i checked up on it
17:14.15beekNaikrovek: not much difference between the two... you should have no trouble learning it.
17:14.15p3nguinIf it were written in Java, I wouldn't expect it to work right.
17:14.18beekducks
17:14.23titterlol
17:14.26Naikrovekwhy duck
17:14.32Naikrovekit woudln't take me long to learn it
17:15.12Naikrovekjava has come a loooooong way
17:15.12Kattyi do believe i have located a squirrel nest in an old, mostly dead tree, near the north side of the house
17:15.27Naikrovekwhen i started learning java i was sure it was useless
17:15.28Naikrovekthen i learned it
17:15.29titterdinner?
17:15.41Naikrovekit is not useless, and it runs very nicely indeed
17:15.43*** join/#asterisk thehar (i=thehar@thehar.xmission.com)
17:15.53titterjust slow
17:15.57Naikroveknot slow at all
17:16.00Naikroveknot anymore
17:16.02titterlol
17:16.03beektitter: Not at all slow.
17:16.13Kattythe appearance of baby squirrels in feb/march is promising :>
17:16.15Naikrovekjava 1.0, 1.1 slow.  java 5 and 6 - faster than C in some regards
17:16.29beekI happen to like Java very much.
17:16.32titterI am just used to PIX devices
17:16.44Naikrovekdon't judge java by the ASDM
17:16.49Naikrovekor the SDM
17:17.02Naikrovekgui apps and server apps have very little in common performance wise
17:17.45*** join/#asterisk jfreema (n=root@hpbx-1.peace-communications.net)
17:18.21titterOh I know, just the tools I use commonly are written in Java, and the GUI is terribly slow ... epscially the Oracle SQL Dev tool
17:18.52Naikroveki wrote some middleware between MS Ops Manager and HP Service Desk in Java and C# and it was a server application.  that thing performed unbelievably well
17:18.53jfreemagreetings
17:19.02titterI prefer C# myself
17:19.14Naikrovekfar and away the bottleneck was the C# api for MOM
17:19.19beekpinches nose shut at smell of MS
17:19.24Naikrovekwhich was plenty fast itself
17:19.28titterlol
17:19.55jfreemaHoping for some help... my exhaustion addled brain isn't having much luch with a couple of problems with my 1.6.2 system on Cent 5.4... 2 problems
17:19.58titterI never liked ASP.net until it was my job ... then I learned it, and realized it was terribly simple
17:20.09Naikrovekjfreema: just ask the question
17:20.13Naikroveks
17:20.21beek~ask
17:20.22infobotQuestions 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.
17:20.26jfreema#1. SIP registration seems buggy... tcpdump shows sip registration messages reaching the box, but asterisk appears to ignore them
17:20.29titterI wrote a GUI for provisioning my Polycom phones with Asterisk with C# and ASPnet 3.5
17:20.47Naikrovektitter: care to share?  my polycom provisioning script is in Perl
17:20.54jfreema#2. User's saved vm greetings aren't being played
17:21.07Naikrovekit works nicely but i'm always on the lookout for better stuff
17:21.51beekjfreema: #2 is because you didn't use the 'u' or 'b' option on the Voicemail command
17:22.05HenrikJottI have a problem whith beep sounds to agents in queue. All agents avaliable (waiting, not paused) in the queue hear a beep sound when one agent hangs up a call. Does anybody know anything about this?
17:22.13titterI will release it soon, still working on it ... I used it to provision the 20 phones in my office here
17:22.21titterI will send you a pm
17:22.24kc8pxyhas not used asp since a hobbist project building a logged chatroom using metarefreshes, and access as a backend.
17:22.34kc8pxy:D
17:23.03*** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com)
17:23.50beekjfreema: assuming that the Asterisk user has the proper permissions to the /var/lib/asterisk directory
17:25.19beekjfreema: sip registrations work fine.  You should add a few details, such as which * version, and pastebin your configs.  Does the CLI show anything when you attempt the registration?
17:25.21[TK]D-Fenderjfreema: And #1.... show us * SIP debug from CLI in a pastebin
17:25.23[TK]D-Fender~pb
17:25.24infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
17:25.26[TK]D-Fender^^^^^^^^
17:25.39*** join/#asterisk jkroon (n=jkroon@dsl-244-25-144.telkomadsl.co.za)
17:26.12beekjfreema: [TK]D-Fender is a fifth-level black belt grandmaster of Asterisk-fu.    Take advantage and ask your questions.
17:26.24*** part/#asterisk RicWright (n=Ricwrigh@firewall.a-n-t.net)
17:26.27jfreemadebug doesn't show anything for registrations
17:26.41jameswf~[TK]D-Fender
17:26.42infobot[TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy.
17:26.46jayteetitter, are you going to release the tool under GPL?
17:26.53jfreemaI have some users registered and some not, that were yesterday, with no changes on the endpoint
17:26.53[TK]D-Fenderjfreema: Show me your tcpdump that does show them, and show me you actually activating SIP DEBUG @ * CLI
17:27.54*** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net)
17:28.21titterjaytee: probably not ... it's nothing terribly amazing
17:28.41jfreemawill be amom ent on the tcpdump, but here's the cut from the cli
17:28.44jfreemahpbx*CLI> sip set debug ip xxx.xxx.xxx.xxx
17:28.44jfreemaSIP Debugging Enabled for IP: xxx.xxx.xxx.xxx
17:29.09titterjaytee: ill post the code somewhere, and hopefully someone will take it further. i wrote it to suit my needs.
17:30.21[TK]D-Fenderjfreema: GLOBAL, not IP based
17:30.31*** part/#asterisk davidcsi (n=davidcsi@89.141.97.190.dyn.user.ono.com)
17:30.43[TK]D-Fenderjfreema: and then show mat that backup + tcpdump that shows otherwise
17:31.46HenrikJottI have a problem whith beep sounds playing to agents in queue. All agents avaliable (waiting, not paused) in the queue hear a beep sound when one agent hangs up a call. Does anybody know anything about this?
17:32.38p3nguinhenrikjott: Agents are not using the phones when the phone emit a beep?
17:33.17jfreemahpbx*CLI> sip set debug on
17:33.19jfreemaSIP Debugging re-enabled
17:33.46p3nguinjfreema: Is there any reason why you are pasting that in the channel?
17:33.57*** join/#asterisk emaia (n=emaia@adonis.iportalmais.pt)
17:34.13[TK]D-Fenderjfreema: And where is the contradictory tcpdump that shows traffic ocming in?
17:34.28jfreemagetting it now
17:34.33*** part/#asterisk emaia (n=emaia@adonis.iportalmais.pt)
17:34.35*** join/#asterisk emaia (n=emaia@adonis.iportalmais.pt)
17:36.04HenrikJottp3nguin: no, they don´t have a call connected. They are waiting for a call but they have a connection to the * server. Our server dials a lot of numbers and when one gets answered we connect it to a waiting agent (and * plays a beep to the agent the call gets connected to). But when that agent hangs up the call (or the other party hangs up the call) all agents waiting in the queue hear a beep, but not the agent that actually makes the han
17:36.55*** part/#asterisk emaia (n=emaia@adonis.iportalmais.pt)
17:36.59bpgoldsbIs it possible to wildcard Sip hints? i.e. if I want 'hint(SIP/200) 200 => Dial(SIP/200);', but for extensions 201-250 also, all in one line.
17:37.15*** join/#asterisk ozziffied (n=ozziffie@70.159.19.194)
17:37.55bpgoldsbMaybe something like hint(SIP/${EXTEN}) XXX => Dial(SIP/${EXTEN);
17:38.21HenrikJottp3nguin: I´m also using MixMonitor to record the call, could it be mixmonitor that makes the beep?
17:38.23ManxPower-workbpgoldsb: MAYBE in 1.6.x
17:38.31bpgoldsbManxPower-work: I'm using 1.6.2 :)
17:38.57ManxPower-workbpgoldsb: Generally it's a bad idea to wildcard extensions or hints for a phone in a production PBX.
17:39.07bpgoldsbWhy is that?
17:39.34ManxPower-work1) because it's a bad idea to use the extension as the sip userid
17:39.38ManxPower-work2) see 1)
17:40.11*** join/#asterisk jdoe (i=jdoe@falseprophet.ca)
17:40.11bpgoldsbI keep all my phone extensions in the 300-399 range, so it seems like using 1 line instead of 100 would be.. beneficial
17:40.29ManxPower-workbpgoldsb: What happens when 25 of those people want something your default does not support.
17:40.41ManxPower-workbut it's a moot point.
17:40.46bpgoldsbManxPower-work: I work for a company where management tells them 'too bad' :P
17:41.02ManxPower-workbpgoldsb: I guess it sucks to be you.
17:41.10Kobazbpgoldsb: i was with a company like that
17:41.13Kobazbpgoldsb: you know what happened?
17:41.14bpgoldsbManxPower-work: Hah.  Did.  I leave next friday :)
17:41.20Kobazbpgoldsb: the company failed miserably
17:41.25ManxPower-workKobaz: I bet users got what they wanted.
17:41.55bpgoldsbEither way, without jumping around with could/should/shouldn't, is there a way to do it?
17:41.58p3nguinhenrikjott: So you have agents on hold in a queue rather than letting agents wait for their phones to ring?
17:42.04ManxPower-workbpgoldsb: it is also hard to do something like exten => _XXXX,1,Dial(SIP/${EXTEN}) when your sip userids are like 0004257623-a like a larger scale PBX might be setup.
17:42.24ManxPower-workbpgoldsb: If there is it would be documented in UPGRADE*.txt in the Asterisk source directory.
17:42.43bpgoldsbI understand there are reasons why it might be a bad idea, but then again, I know my system and company well enough to feel it's safe :)
17:42.52ManxPower-workSetting the SIP userID to be the same as the extension is a classic n00b mistake.
17:43.18bpgoldsbReally?  It's been working absolutely fine for us for 3 years
17:43.18ManxPower-workbpgoldsb: go read UPGRADE*.txt again.
17:43.33ManxPower-workbpgoldsb: What do you do when you want one extension to dial multiple devices?
17:43.39HenrikJottp3nguin: yes! they are all logged in and have a call going on with *. they never actually hang up their phones, we disconnect the call from asterisk and make them available again. So, yes, they are on hold =)
17:44.09p3nguinI can't imagine trying to conduct calls in that manner.
17:44.19KobazManxPower-work: well, it can be accomplished either way
17:44.27HenrikJottp3nguin: and we do not need any sound at all on hangup, so if there is a way to turn that off altogether that would work as well
17:44.32p3nguinSounds like you have a MeetMe for every agent rather than normal queues.
17:44.40*** join/#asterisk pirulo (n=andres@65.102.99.5)
17:44.53KobazManxPower-work: if you have two sip devices with userids 301, 302, and then you have an asterisk extensions with 303, that rings 301+302
17:44.55bpgoldsbManxPower-work I don't see how thats an issue
17:45.16bpgoldsbYa, 212 => { Dial(SIP/300,SIP/301); }
17:48.17jfreemathe interesting thing on my registration problem is that if I change the sip.conf such that host=xxx.xxx.xxx.xxx, where xxx.xxx.xxx.xxx is the endpoint ip, the peer comes up, then I can change it back to dynamic with no problems
17:48.23ManxPower-workyes, but would it make more sense to REALLY abscract the userids?
17:48.35*** join/#asterisk joako (n=ston3d@opensuse/member/joak0)
17:48.48ManxPower-workwhy do you have extra { } and ;?
17:49.01[TK]D-Fenderjfreema: If your peer is trying to register to * it HAS to be "dynamic"
17:49.04*** join/#asterisk Ad-Hoc (n=nimbus@62.1.130.79.dsl.dyn.forthnet.gr)
17:49.06KobazManxPower-work: depends
17:49.07[TK]D-FenderManxPower-work: AEL <-
17:49.26bpgoldsbAEL + Habit
17:49.35KobazManxPower-work: in 90% of the systems i set up, i never see the need to abstract userids
17:49.36[TK]D-Fenderbpgoldsb: and s/,/&/
17:50.10bpgoldsbManxPower-work: I would argue if there isn't a need to abstract them, then why do it?  It just confuses other people
17:50.11HenrikJottp3nguin: okay, but this is a telemarketing system and agents need to connect instantly with the caller so we can´t wait for the agent to pick up..
17:50.15*** join/#asterisk neutr0n (n=neutr0n@24.248.196.205)
17:50.16jfreemaFender- I know.. which is what's strange.. it won't register at all unless I set host to the current ip of the endpoint, do a sip reload, then set it back to dynamic and do another sip reload
17:50.22ManxPower-workbpgoldsb: What do you do when someone calls with a problem with extension 300?  Is it extension 300, extension 212, or device 300 that has the issue?
17:50.43ManxPower-workjpeeler: YOU ARE WRONG!
17:50.57p3nguinhenrikjott: In that situation, I would have Asterisk call the agent and then call out to a person.
17:51.02KobazManxPower-work: depends on how extension 300 is reached
17:51.05ManxPower-workjpeeler: host=ip.ad.dr.ess PREVENTS the device from registering
17:51.14KobazManxPower-work: what do you do when someone calls with a problem with sip userid 32948sd34aaab
17:51.20rocksfrow[TK]D-Fender, i'm actually saving ~$200 / mo switching my service from analog to PRI :-)
17:51.29neutr0nis there a way i can get the md raid setup screen to come up with a boot parameter?
17:51.29ManxPower-workKobaz: exactly.  I just have the user tell me the MAC on the bottom of the phone and I never have to wonder which deviceI should be working with.
17:51.40neutr0ndmraid isnt working for me i am getting ata timeout errors
17:51.45[TK]D-Fenderjfreema: No.  If a device is to register to * you shouuld NOT be putting in an IP period.  this is not legal
17:51.48ManxPower-workneutr0n: You might want to ask on a channel where people ask such questions.
17:51.58bpgoldsbManxPower-work: I imagine you work in a much bigger environment where you have MUCH different issues than me.
17:52.08KobazManxPower-work: but you still have the same problem
17:52.10[TK]D-Fenderrocksfrow: Congratulations
17:52.10bpgoldsbOur Asterisk setup has a peek of 30 users
17:52.17*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
17:52.21bpgoldsbAll in one small building
17:52.21ManxPower-workbpgoldsb: Maybe so, but I suspect my users are just dumb as rocks.
17:52.26KobazManxPower-work: you now need to see, how device 32948sd34aaab is reached... in order to fix a callflow problem
17:52.32HenrikJottp3nguin: but this way is basically the same, the only difference is the agent already have a connection. And everything works fine except for the beep! =)
17:52.34jfreemaI understand this.... that's why I'm looking for help with it
17:52.51[TK]D-Fenderjfreema: Whats to help?  Leave as dynamic.  the end
17:52.54ManxPower-workjfreema: You could start by not saying things that cannot be true.
17:52.56bpgoldsbWell in 3+ years of debugging it, I've never had an issue with using a exten -> sip mapping
17:52.57neutr0nManxPower-work: which channel would that be?
17:53.07ManxPower-workneutr0n: not #asterisk.
17:53.08[TK]D-Fenderneutr0n: ##linux
17:53.09KobazManxPower-work: 6 of one, half dozen of the other... it's all the same... it's just names/labels
17:53.18jfreemasimply putting the endpoint config in sip.conf with host-dynamic should work after a sip reload, since the endpoint is registering with *
17:53.19bpgoldsbneutr0n: Are you using Ubuntu or something else?
17:53.20neutr0nits an asterisk install
17:53.27neutr0nthere is no mdraid setup menu
17:53.31[TK]D-Fenderneutr0n: No, its an OS install
17:53.31ManxPower-workneutr0n: Asterisk does not have MD or RAID.
17:53.36neutr0nas with the default centos install
17:53.51*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
17:53.52ManxPower-workPerhaps you are thinking of AsteriskNOW or some other GUIfied/ISO distro of Asterisk that is not supported here.
17:53.52[TK]D-Fenderneutr0n: then go ask in #centos
17:53.53neutr0nbut yet it uses dmraid using md
17:53.57bpgoldsbneutr0n: add bootdegraded=true to the kernel parameters at boot.
17:53.59neutr0nthe install is modified
17:54.09Kobazneutr0n: join #linux
17:54.09[TK]D-Fenderneutr0n: Asterisk has nothing to do with disk interfaces.
17:54.36jfreemabut it doesn't.... I have to put it in to forc asterisk to attempt to register with the endpoint, (host=ip), then change it to dynamic... if I go straight to dynamic, the endpoint never registers with Asterisk, and Asterisk acts like it doesn't see the endpoint attempting it at all
17:54.40neutr0nlol no its asterisk latest version
17:54.44jfreemahence my confusion, lol
17:54.53ManxPower-workjfreema: you are still saying things that cannot be true.
17:54.56Kobazneutr0n: it's like asking for help with your car engine, in a stereo store
17:55.01jfreemaI know
17:55.04[TK]D-Fenderjfreema: woah.. to have * register to an outside poitn?  Your SIP peer has NOTHING to do with registration
17:55.18bpgoldsbneutr0n: Asterisk is an application inside the Linux OS.  It's not a version of Linux.
17:55.24VxJasonxVAnyone have any good suggestions for a SIP/IAX2 iPhone App?
17:55.24ManxPower-workKobaz: maybe he's just a jerk?
17:55.27neutr0nbut it has to do with the modified asterisk installation - i am just trying to get the mdraid setup screen back
17:55.27KobazManxPower-work: heh
17:55.40ManxPower-workSee.  Asshole won't leave.
17:55.41jfreemait's really acting like a statefull firewall config issue, but there's no stateful firewall... endpoints are on the same segment
17:55.46[TK]D-Fenderneutr0n: You have OS problems.  this is not an Asterisk problem.  Move on.
17:55.47VxJasonxVpreferably one that has no ability to "buy credits" from a provider? something good and vendor neutral?
17:56.01Kobazneutr0n: you wont get any help here with raid
17:56.04jfreemaip tables is setup to allow all traffic into the box on udp 5060 and the rtp streams
17:56.15bpgoldsbneutr0n: You're probably using something like AsteriskNOW or Trixbox, which are based on Centos.  Ask in #centos.
17:56.19[TK]D-Fenderjfreema: Please clarify what you are doing.  Is AStERISK registering to an outside service, or is an outside device registereing to ASTERISK?
17:56.25Kobaz...back to work
17:57.25*** join/#asterisk roe_ (n=roe___@216-164-215-125.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com)
17:57.46jfreemaAsterisk currently is acting as the call control server in my configuration. Endpoints should be registering with it, except for the sip trunks to other providers, and the iax2 trunk, which are not in question here
17:58.09jfreemawhen I configure an endpoint - Polycom, Snom, Adtran IAD, doesn't matter
17:58.15[TK]D-Fenderjfreema: Your wording is reversing for every wuestion I ask.  Please get your head screwed on straight.
17:59.26jfreemaThe problem I have is that when I configure an endpoint, if I set host=dynamic, the endpoint never registers, and asterisk never even shows the registration attempt. verbosity on the console is set to 7, and I've enabled sip debugging
17:59.46ManxPower-workjfreema: If the endpoint is not registering then the endpoint is not registering.  Fix the endpoint.
17:59.54[TK]D-Fenderjfreema: Then it isn't registering at all
18:00.04jfreemahowever, if I set the endpoint to host=ipaddress, do a sip reload, then set it back to host=dynamic, do another sip reload, it registers just fine
18:00.31[TK]D-Fenderjfreema: No * setting will magically make a registration packet come in <---
18:00.40[TK]D-Fenderjfreema: that is BS
18:00.40jfreemaI understand this
18:00.41ManxPower-workjpeeler: That is a CLASSIC "you don't have a port forward working" problem.
18:00.48[TK]D-Fenderjfreema: PASTEBIN
18:00.54ManxPower-worknow turn of your firewall and try it again
18:00.55jpeelerManxPower-work: you're not talking to me
18:01.04ManxPower-workjpeeler: correct.
18:02.00jfreemaPastebin what? I've got live calls on this system and the sip debug global is very large
18:02.13jfreemawhich is why I was debugging just the endpoint ip
18:02.25[TK]D-Fenderjfreema: You are telling us things that cannot be true on this and providing no backup
18:02.32[TK]D-Fenderjfreema: You are looking patently nuts
18:02.46ecranejfreema: Set up another system for doing tests/debugs and use that.
18:03.01[TK]D-Fenderjfreema: And every time someone things they are being "smart", the reverse is usually true
18:03.11[TK]D-Fenderthinks*
18:04.18ManxPower-workjfreema: If the registration packets are not getting to Asterisk then you have some OTHER issue.  The fact that setting host=ipaddress then setting it back to dynamic indicates you have a firewall or NAT or port forwarding issue.  You need to fix that and no amount of Asterisk debugging will get it fixed.  It is a non-Asterisk problem.  Use wireshirt or TCPdump to see if the packets are arriving on the Asterisk box.
18:04.27jfreemaheh... it works on my dev system.. just not on the production one
18:04.46jfreemaonly diff between the two is 1.6.2 on production vs 1.6.1 on the dev at the moment
18:04.57jfreemaand I know I look nuts
18:05.09ManxPower-workhost=ip.add.ress will initate a packet to the destination, automatically opening up any automatic NAT traslation.  That translation is still open (at least for a little while) when you set host=dynamic.
18:05.23Kattyjfreema: go stand in my front yard.
18:05.27Kattyjfreema: the squirrels love nuts.
18:05.41ManxPower-workThen since there is no traffic on that port the NAT translation will close and you the phone won't be registered anymore.
18:05.47*** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147)
18:05.49jfreemalol... that's why I'm so confused...it SHOULD work, but it's not... I thought there was an off chance selinux may be causing problems, but I'm not seeing anything in /var/log/messages
18:06.44jfreemathere's no NAT between the endpoints and the *, so that can't be an issue. I first thought firewall as well, but the only firewall in the mix is iptables
18:07.04[TK]D-Fenderjfreema: which of course I trust as much as the rest of your scenario....
18:07.59ecranejfreema: roll the dev system back to 1.6.1 and see if the problem shows up there
18:09.09jfreemawell... the endpoints and the asterisk box are on the same L2/L3 segment
18:09.46[TK]D-Fenderjfreema: You aren't getting it.  These stories are worthless.  You've got nothing to show us.  This is a waste of time.
18:09.47jfreema* prod is .16/24, dev is .15/24, endpoint I'm testing with is .13/24
18:10.24jkroonjfreema, do you have SIP connection tracking in your iptables mix?
18:10.57jfreemahttp://pastebin.com/m7d8d8465 is the iptables -L output
18:11.04jkroondo you have the modules as much as loaded?
18:11.12jkroongunzip < /proc/config.gz | grep SIP
18:11.35jkroonitpables -L is pointless, the module breaks it before it even gets to the filter table in some strange cases.
18:12.02bmoracaanyone have any idea what would cause an as5400 to lose CNAM information between ISDN (q931 debug shows it being delivered) and SIP (sip debug shows that the as5400 does not pass the CNAM)?
18:12.21jfreemawhich modules am I looking for?
18:12.59jfreema\/proc/config.gz doesn't exist on my system
18:13.11jkroonany module with the phrase "sip" in it, or if it's compiled into the kernel the CONFIG_NF_CONNTRACK_SIP and CONFIG_NF_NAT_SIP config options.
18:13.20jkroonjfreema, check in /boot for a file called "config" ?
18:14.04jkroonnf_nat_sip and nf_conntrack_sip is the modules.  I've seen them just "randomly" drop SIP packets.
18:14.10jfreemano sip modules in lsmod output.. looking at the rest
18:14.59devmodanyone has an idea on how to create this sort of prompts using open source software? http://www.aupix.com/index.htm?/support/astvideo.htm
18:15.41jfreema[root@hpbx asterisk]# cat /boot/config-2.6.18-164.9.1.el5 | grep SIP
18:15.41jfreemaCONFIG_IP_NF_SIP=m
18:15.42jfreemaCONFIG_IP_NF_NAT_SIP=m
18:15.45jkroonjfreema, it may not even be it at all, i just realized again today though that that module caused me lots of headaches.
18:15.54jkroonthen you're probably ok if the module isn't loaded.
18:16.10Kobazlsmod | grep sip
18:16.45jfreema[root@hpbx asterisk]# /sbin/lsmod | grep SIP
18:16.45jfreema[root@hpbx asterisk]# /sbin/lsmod | grep sip
18:16.46jfreema[root@hpbx asterisk]#
18:16.52jfreemadoesn't appear to be loaded
18:17.15*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:17.16jkroonthe symptom you're describing though is exactly what I saw on one of my systems.  if you run a tcpdump for the "registering" ip, does tcpdump show the reigstration packet?
18:17.25jkroonjfreema, grep sip (all lower case)
18:17.29jfreemayes, it does
18:17.34diatonic_afkdevmod, I would think ffmpeg could encode them to the proper format after you generate the video
18:17.49jkroonok, so it's getting lost at some point between the iface and asterisk.
18:17.51*** join/#asterisk oej (n=olle@ns.webway.se)
18:17.55jfreemaright
18:18.03jkroonit sounds EXACTLY like i was seeing with that mod.
18:18.23jfreemahmmm
18:18.55Kattydude.
18:18.58Kattythere are geese in the back yard.
18:19.01Kobazare you perhaps sending your traffic through a linux box router that is using contrack_sip
18:19.02KattyHONK
18:19.03jfreemaI originally thought it was an selinux problem as I'm running on a fairly generic Cent 5.4 platform with updates
18:19.33jfreemanope on the linux router... the endpoints are on the same segment as the *
18:19.38jfreemaso no router needed
18:19.50*** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147)
18:20.04jkroonjfreema, ok, two "tests":  1.  disable iptables, 2. disable selinux.
18:20.09*** join/#asterisk Alagar (n=Administ@122.164.43.192)
18:20.16jkroonthat way you can at least confirm whether it's one of them or not.
18:20.26jfreemagimme a minute to let calls in the system clear
18:20.31ManxPower-workAwesome!  I have a user just like many users here.  The user problem report:  I think that when you press transfer it starts playing right away, and then when you complete the "park" it starts again. Is there a way to either have park continue the same or have it play only in "park" and not right away when you press "transfer"?
18:21.11jkroonwtf?!?
18:21.19ManxPower-workjkroon: Exactly!
18:21.21Kobazhaha
18:21.24bmoracai don't believe i've experienced that problem...or, at least, none of my customers have complained about it
18:21.47devmoddiatonic_afk, so encode to h263 and h264 but what codec to use for the audio?
18:23.00HenrikJottHow can i playback a sound on a specific channel? what i want to accomplish is to playback a beep sound to one agent, but right now the beep plays to the other party... any thoughts?
18:23.46ManxPower-workbmoraca: What problem.  The problem if the MoH starting and then MoH starting?
18:23.56bmoracayeah
18:24.00KobazHenrikJott: more details
18:24.17KobazDETAILS
18:25.02ManxPower-workHenrikJott: change the option you are using (you don't say which option) to play the beep to the other party.  With a better question, you might get a better answer.
18:25.41jfreemaok.. so turning off itpables reveals a lot
18:25.41*** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
18:26.01jkroonbroken iptables ruleset?
18:26.04HenrikJottKobaz: when a call gets connected to one of the agents in the queue i want to playback a beep sound. Right nog i use the command "Playback beep_connection" which is a custom file but it plays to the incoming call and not to the agent..
18:26.08ManxPower-workjfreema: Aren't you glad we all spent our time to convince you to turn it off?
18:26.24HenrikJottManxPower-work: What do you mean by option?
18:26.31*** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com)
18:26.40jkroonHenrikJott, core show application Queue
18:26.41jfreemaI don't have a problem with turning it off.. it's the leaving it off that gives me nightmares
18:26.57KattyQwell: are you gonna play STO?
18:26.57ManxPower-workHenrikJott: pastebin an example of the incorrect stuff in the CLI, pastbin your dialplan too
18:26.59jkroonalso look at the sample queue.conf file for "announcement" options to be played to the agent.
18:27.07QwellSTO?
18:27.14KattyStar Trek Online MMO
18:27.23Qwellno O.o
18:27.25Kattyk
18:27.30ManxPower-workI thought it was the Steller Transexual Organization
18:27.47KattySO WHAT IF IT IS
18:27.51KattyDO YOU HAVE A PROBLEM WITH THAT
18:28.12KobazHenrikJott: queue supports that in the config file
18:28.15ManxPower-workKatty: Hmm?  I used to have a very cool trannie roommate, no I don't have a problem with that.
18:28.20KattyManxPower-work: k.
18:28.39ManxPower-workKatty: I just wasn't aware Qwell was a transecual.
18:28.54KattyManxPower-work: he just plays WoW MMO.
18:29.02KattyManxPower-work: figured i'd ask.
18:29.02QwellManxPower-work: There's a lot you don't know about me
18:29.31*** join/#asterisk wendell (n=wendel@189.7.251.29)
18:29.52HenrikJottKobaz: I know, but i had a problem with it playing beep sounds when i shouldn´t so i temporarily replaced the standard beep file with another one with no sound, so now i just need to make asterisk play a beep to a specific channel.
18:30.02jfreematurning off iptables seems to have resolved the issue.. endpoints are now registering ok, and I'm seeing a couple that have digest mismatch issues that are easy to fix, where I wasn't seeing anything before
18:30.12HenrikJottthis is my pastebin, it´s in php but this is what i´m doing: http://www.pastebin.org/74994
18:30.31jkroonjfreema, two choices from here:  1.  fix your ruleset, or 2. leave iptables off.
18:30.42ManxPower-workHenrikJott: I wish you the BEST of luck.
18:30.58KobazHenrikJott: you can use chanspy whisper to play a track to a channel on demand
18:31.02HenrikJottMy dialplan and configuration is in mysql and that makes it hard to pastebin
18:31.22KobazHenrikJott: you should use the built in queue option that plays a track when someone enters a queue
18:31.38Kobazer, connects to an agent from an queue... rather
18:32.02KobazHenrikJott: that paste doesn't help
18:32.27diatonicdevmod, Do you think the audio is muxed in to those video prompts?.. or would it be carried from a separate file?
18:32.35KobazHenrikJott: simplify your config using standard dialplan stuff, and then paste everything
18:32.44jfreemawhat's needed in the ruleset beyond what's already there... http://pastebin.com/m7d8d8465
18:32.54wendellhi hi champs... i have one problem in dahdi... my log the asterisk is Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection.  You have options:
18:33.52Kobaz~details
18:33.53infobotIf you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever.  Give as many details as you can or else no one can give any suggestions.
18:34.44*** join/#asterisk batphone (n=ping@cpe-67-9-163-227.austin.res.rr.com)
18:34.56Kattyringringringringringringring battyphone
18:34.59Kattydo do be do be do
18:35.08batphonewhat is a good sip re-registration time setting to keep firewalls from removing NAT entries for phones?
18:35.11jfreemathanks for the assist folks.. gotta run
18:35.18ManxPower-workjfreema: put a log rule at the end of your other rules, see what is being logged
18:35.29leifmadsenbatphone: depends on the NAT really... I've used anywhere from 60 to 300 seconds
18:35.34Kobazbatphone: i use 60 seconds
18:35.51*** join/#asterisk DarkFibre (n=dmelouk@127.159.119.70.cfl.res.rr.com)
18:35.57batphoneseems like a phone is effectively down until it either re-registers or an outbound call is made.
18:35.57*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
18:35.58benngardis there any dager involved with having: qualify=yes at every extension in sip.conf?
18:36.05benngarddanger even
18:36.18carrarbox could explode
18:36.27Kobazit might generate a black hole
18:36.28carrarsending fragments everywhere
18:36.44Kobazbenngard: depends what you want to do
18:36.54*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
18:37.11benngardjust like to se round trip when i do type sip show peers
18:37.49batphonewhat is qualify? is that icmp or a SIP packet?
18:37.51batphonei forgot
18:38.00Naikroveksip i think
18:38.00batphonei remember it being a SIP packet, but what kind?
18:38.33benngardi dont know, thats why i am asking the gurus ;)
18:39.03[TK]D-FenderSIP
18:39.07HenrikJottKobaz: where do i set which sound file to play when a agent gets a call from a queue?
18:39.13Naikrovekyou want network round trip time, yes?
18:39.28benngardyes
18:39.31Naikrovekif qualify is on it'll show you how long it take for the phone to get around to responding to the SIP packet
18:39.38Naikrovek(it's not network latency, but much larger)
18:39.38ManxPower-workbatphone: that is correct.  Does your batphone have have a Nat Keepalive setting?
18:39.55KobazHenrikJott: i don't remember offhand, look at the queue.conf config file example, it's in there
18:40.11batphoneManxPower-work: my switches have many different models of phones connected to them, not just polycoms or ciscos.
18:40.33benngard0317998972/0317998972      10.242.10.133    D          5061     OK (5 ms) <- cant be "much higher"
18:40.47batphoneManxPower-work: but its an ongoing issue to have phones disappear from the switch unless they registration and timeout settings match up perfectly.
18:40.49ManxPower-workbatphone: then you'll have to use qualify and hope you have zero packet loss
18:41.06batphonefwiw im not using asterisk, i just come in here sometimes for general voip help and people here are always good about it
18:41.21ManxPower-workbatphone: Oh!  You're totally screwed.
18:41.23*** join/#asterisk cusco_ (n=trilili@2001:0:53aa:64c:4e3:665f:a077:acde)
18:41.26Kobazhaha
18:41.27cusco_hi
18:41.32batphoneim going to poke my switch vendor with this and see if there is a way to get a qualify feature built in.
18:41.49ManxPower-workbatphone: qualify is an Asterisk term.
18:42.09leifmadsenNOTIFY is the message
18:42.21ManxPower-workleifmadsen: I thought it was OPTIONS message
18:42.23cusco_I have one asterisk registered with sip on another
18:42.30leifmadsenerrrr... ya, that :)
18:42.41ManxPower-workleifmadsen: drink more coffee
18:42.47cusco_now I can't find out why the "another" won't register on the "one asterisk" back
18:42.48*** part/#asterisk Rawplayer (n=kevin@wickedleaks.nl)
18:42.50leifmadsenManxPower-work: I'm feeling sick today (and yesterday)
18:42.54leifmadsenneed more green tea
18:43.22Kattycelestial seasonings is giving away free samples of green tea if you friend them on facebook and take a quiz.
18:43.29cusco_sip show registry show 0 registrations
18:43.50Kattycan you say: chinese takeout.
18:43.51ManxPower-workcusco_: sip show registry shows what REMOTE servers Asterisk is registered TO
18:43.57cusco_yes
18:43.58ManxPower-workIs that what you were looking for?
18:44.00cusco_yes
18:44.02ChannelZhas toured the celestial seasonings factory.. even though I don't really like tea
18:44.17KattyChannelZ: more of an iced tea person myself.
18:44.19cusco_there is 192.168.2.7 and 192.168.2.5
18:44.33cusco_on .5 sip show registry shows: 192.168.2.7:5060               N      primario           105 Registered           Tue, 12 Jan 2010 18:42:30
18:44.34ChannelZYeah I like iced tea but I can't drink very much of it
18:44.42cusco_now on .7 shows 0 registrations
18:44.55Kattygoes to get lunch
18:45.04cusco_how can I for ce it to try and register?
18:45.07p3nguinI have sent for my lunch to be delivered to me.
18:45.34p3nguinMcDonald's food/tea should keep me kicking for another 5-6 hours.
18:47.04ManxPower-workcusco_: what machine is connected to which machine?
18:47.29*** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca)
18:48.39batphonequalify has always been one of the cool features asterisk has. with that turned on i dont think i ever had a problem with firewalls.
18:49.09batphoneaside from cases when three or more firewalls were in the way, in which case i would just shoot the customer.
18:49.55devmodany ideas on how to create prompts like these using open source tools? http://www.aupix.com/index.htm?/support/astvideo.htm Any specific requirements for asterisk while encoding?
18:50.17batphoneleifmadsen: i believe qualify sends a SIP OPTIONS ping to the phone, not a notification.
18:50.31Naikrovekwhy would someone put voip phones outside a secure network.  "to access email and receive the daily joke email you must be connected to the vpn, but to handle business over the telephone you do not."
18:50.45leifmadsen<ManxPower-work> leifmadsen: I thought it was OPTIONS message
18:50.51leifmadsen<leifmadsen> errrr... ya, that :)
18:50.55leifmadsenbatphone: ^^^
18:52.50batphoneah
18:53.30batphonesquints at these tiny glowing letters
18:56.58*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
18:58.09*** join/#asterisk chazzm (n=chazz@173-24-238-25.client.mchsi.com)
19:00.59*** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca)
19:05.59*** join/#asterisk dennis00 (i=dennis00@unaffiliated/dennis00)
19:06.05cusco_ManxPower-work: both connected to each other, would you like me to paste some config files?
19:06.36cusco_hold on
19:06.43ManxPower-workcusco_: no.
19:06.46dennis00I recieve this error for package zaptel: -bash: ./configure: No such file or directory. What is the reason?
19:07.04cusco_err
19:07.09ManxPower-workdennis00: chances are you don't have the required software installed to build Asterisk.
19:07.18ChannelZthe file 'configure' does not exist perhaps?
19:07.22dennis00I have autoconf.
19:07.34dennis00But I don't exactly know how configure works to be honest, let me check the dependency list.
19:07.59tzafrirdennis00, also note that the latest version of zaptel is dahdi
19:08.26dennis00I see, stupid 2007 tutorial.
19:08.27ManxPower-workdennis00: You are IN the zaptel source directory when you issue that command?
19:08.36dennis00ManxPower-work: Yes.
19:08.51ManxPower-workwhen you do an "ls" does "configure" show up.
19:09.02dennis00Can somebody give me a link to a simple Asterisk configuration? I have bought a phone number + have a server for Asterisk testing.
19:09.15dennis00ManxPower-work: no, it wasn't  in the tar I guess.
19:09.16tzafrirdennis00, if you're just starting now, use asterisk 1.6.2 dahdi , and such
19:09.27ManxPower-workdennis00: then you downloaded a corrupted tar file.
19:09.27dennis001.6.2 = beta?
19:09.37tzafrirnope
19:09.44ManxPower-workdennis00: no.  If it was beta it would say BETA or RC
19:09.55dennis00There is enough documentation for 1.6?
19:09.56ManxPower-workdennis00: did you download it from downloads.digium.com?
19:10.11ManxPower-workdennis00: all documention is in /path/to/asterisk/src/doc/
19:10.20dennis00digium.com yes.
19:10.35dennis00http://www.kempgen.net/asterisk/current/ confused me.
19:10.41[TK]D-Fenderdennis00: Instructions are in the tarball
19:10.45ManxPower-workdennis00: then you seriously screwed something up.
19:11.15cusco_ManxPower-work: http://paste.debian.net/56499/
19:11.34[TK]D-Fenderdennis00: And nowhere in there do i see them suggesting ./configure for zaptel
19:11.38dennis00hmm,, true, it seems the tar.gz file contains the configure file.
19:11.40cusco_I would like both asterisk to be able to dial each other
19:11.53cusco_192.168.2.7 can dial to 192.168.2.5
19:11.53dennis00I wanted to use this http://www.howtoforge.com/asterisk_pbx_linux tutorial.
19:11.58dennis00But it's heavily outdate.
19:12.02dennis00+d*
19:13.12ChannelZ~book
19:13.13infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
19:13.19[TK]D-Fenderdennis00: Ok, well I'd advise the latest 1.6.1 series release + DAHDI
19:13.31[TK]D-Fenderdennis00: And follow the instructions in the tarball
19:13.51tzafrirdennis00, badly old. As are many of the documents on howtoforge
19:13.54Skeeter-is anyone familiar with spectralink wireless phones??
19:14.08*** join/#asterisk nny (n=scott@64.203.239.83)
19:14.10dennis00[TK]D-Fender: both tools and linux? 1.6.2 is recommended, right?
19:14.12*** join/#asterisk prometheanfire (n=mthode@70-88-176-253-Atlanta.hfc.comcastbusiness.net)
19:14.45[TK]D-Fenderdennis00: 1.6.2. branch is non-beta, but still jsut a little too new for me right now.... that will change soon though
19:14.45tzafrir[TK]D-Fender, why?
19:14.52prometheanfireoffhand does anyone know of a tool to measure bandwidth per registration?
19:15.05[TK]D-Fenderdennis00: I'd say 'fair game" if this is a personal server you can afford some hiccups with
19:15.06ManxPower-workcusco_: put your register => WHERE IT BELONGS in [general]
19:15.11nnyhmm let me take another stab at this. I am working on reparking a call using the features.conf starcode. If I try to park it a second time (1.6.2) it doesn't work. [TK]D-Fender suggested I use a local channel to Dial the call with the feature option, can someone elaborate on this?
19:15.12dennis00ls
19:15.15dennis00wrong window.
19:15.16tzafrir[TK]D-Fender, it's not like he's running a full-blown production on it
19:15.32[TK]D-Fendertzafrir: It may be perffectly stable, but just like Windows.. you wanna wait til the first service pack is out ;)
19:15.35tzafrir. ..
19:15.39[TK]D-Fendertzafrir: Which I jsut accounted for :)
19:15.39leifmadsennny: that's an open issue
19:15.49nnyleifmadsen: ahh good to know, any work arounds?
19:15.59ManxPower-workhugs 1.6.1
19:15.59tzafrir[TK]D-Fender, it has been an RC quite some time
19:16.10tzafririt also has quite a few useful fewtures
19:16.11nnyleifmadsen: like can I Dial(Local\something,,kK) ?
19:16.17[TK]D-Fendertzafrir: Yes, and full release for a short time.
19:16.24ManxPower-workAsterisk seems to always have parking problems.  From 0.65 onward
19:16.26nnyhugs 1.2.CVS
19:16.30tzafrirfor instance, a better timing source, if your kernel is recent enough
19:16.37dennis00Should I also install dahdi tools?
19:16.42[TK]D-Fendertzafrir: and i don't disagree.. but I'd like to see it hit 1.6.2.1 at least.... maybe 1 past
19:16.43tzafrirdennis00, yes
19:17.07[TK]D-Fendernny: procrastination : the art of keeping up with yesterday
19:17.09tzafrir[TK]D-Fender, well, there's already 1.6.2.1-rc1 :-)
19:17.13nny[TK]D-Fender: heh
19:17.21[TK]D-Fendertzafrir: Keep on digging!
19:17.26ManxPower-workcusco_: ; Have another asterisk register here register => gateway:xmedic@192.168.x.x
19:17.40[TK]D-Fendernny: And I did gift-wrap that solution for you already...
19:17.47ManxPower-workcusco_: REGISTER REGISTERS TO A REMOTE SERVER.  How many times must we tell you this.
19:18.06nny[TK]D-Fender: yeah you had mentioned the local thing, I just am not clear on how I can set it up yet
19:18.10ibolmoanyone can point me to some articles on mobile solutions.. if i recall i know that asterisk can route the call to my cell if necessary, but aside from being on a laptop and calling by vpn (an assumption) what other solutions are there?
19:18.24nnyso Local/1@parkedcalls?
19:18.24*** join/#asterisk duckz (n=duckz@86.107.84.186)
19:18.35tzafrir[TK]D-Fender, it works well. And it will reduce the pain of migrating from 1.6.1 later on
19:18.55[TK]D-Fendernny: just make your own exten with your own pickup command and FORGET the standard context entirely
19:19.05nny[TK]D-Fender: roger
19:19.08leifmadsennny: no idea -- I'm trying to find the issue that was reported, but it is possible I was remembering wrong and it was a double transfer that didn't work
19:19.10[TK]D-Fendertzafrir: How much pain is it?
19:19.52*** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com)
19:19.53nnyleifmadsen: i did find some bug reports last night, but they were kind of old
19:19.55leifmadsennny: check the open parking issues and see if any of them match what you're experiencing
19:20.05leifmadsennny: there are many non-old parking issues
19:20.25[TK]D-Fenderleifmadsen: Not Dial DTMF features when calling pickup (because it isn't dial).
19:20.38[TK]D-Fenderleifmadsen: to which I told him to call pickup in a nested Local channel
19:20.52leifmadsen[TK]D-Fender: I don't know why you're telling me as it doesn't affect me
19:21.05nnyhmm
19:21.08nnyI had this one too
19:21.09nnyCalls transfered to parking lot immediatly timeout and ring back extension that tried to park call
19:21.15[TK]D-Fenderleifmadsen: Cludgy, but then again you get what you deserve for using shitty phones without real SIP functionality :p
19:21.32leifmadsennny: I count 12 issues updated when searching for 'park' (closed and open) on mantis
19:21.35[TK]D-Fenderleifmadsen: that's HIS issue with parking...
19:21.40leifmadsensince 12-30-2009
19:21.45leifmadsen[TK]D-Fender: ok...
19:21.51Kobazhttp://www.geekologie.com/image.php?path=/2010/01/07/cupcakes-full.jpg
19:21.56[TK]D-Fenderleifmadsen: jsut clarifying for you :)
19:22.02leifmadsen[TK]D-Fender: I didn't need it :)
19:22.16*** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com)
19:22.37nnyugh hehehe
19:22.39prometheanfirewell, it was worth a try
19:22.47nnyi think i have experienced 3 or 4 of these issues
19:22.48leifmadsennny: just wanted to point you at the mantis issue tracker as that is where you should go first when you have an issue -- try and search for open and closed issues that describe an issue you're having as it may have already been fixed beyond the latest release
19:22.49citywokI've been tasked with creating a phone number that acts like a queue, but calls out to cellphones to find somebody, but must make sure that i am not connecting the inbound caller to a cellphone's voicemail
19:23.06citywokis there a better way of doing this then setting a DIal(,15) or something, setting the timeout shorter than the time it takes to go to voicemail?
19:23.12nnyand I suspect [TK]D-Fender's idea of using ParkandAnnounce etc with a custom feature code will bypass most of them
19:23.28leifmadsencitywok: only way to do that is to disable the voicemail on the cell phone, because it is possible the caller could be on the phone, and no-rings would still cause VM to answer
19:23.28*** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl)
19:23.30nnyleifmadsen: yeah thanks I understand, that was a good idea, it clarified 2 or 3 issues I had
19:23.38ManxPower-workcitywok: Yes.  Make the receiver of the call confirm by dialing a digit before connecting the call.  See the various options listed in "core show application dial"
19:23.54leifmadsenManxPower-work: tricky idea :)  I like it!
19:23.55ManxPower-workThe call may still go to voicemai, but the caller won't hear it
19:24.07ManxPower-workleifmadsen: pretty standard when dealing with analog FXO or cell phones
19:24.10citywokManxPower-work: i had considered that, wasn't sure if it would be possible to do before briding the call
19:24.14leifmadsenManxPower-work: which I don't deal with :)
19:24.24ManxPower-workleifmadsen: I don't deal with analog.
19:24.28citywoki do something very similar for automated outbound calls, but not for bridging calls like this
19:24.39leifmadsenI only deal with PRI and SIP now
19:24.40*** part/#asterisk prometheanfire (n=mthode@70-88-176-253-Atlanta.hfc.comcastbusiness.net)
19:24.51ManxPower-workcitywok: time to do serious reading of Dial options.
19:27.16ManxPower-workBut I sometimes do have to send calls to cell phones and "rescue" them back to the corporate voicemail system
19:27.28citywokManxPower-work: alright, i'll have to see how to do this in the docs
19:27.40citywokdo you have any examples of dialplans to do something like this?
19:28.27ManxPower-workcitywok: STUDY.  THE.  DIAL. OPTIONS
19:29.21ManxPower-workrememeber in the macro you use you need to use Read to wait for the digit.  Background and (maybe) waitexten will always make the answer "true" no matter what the user dialed.
19:29.30*** join/#asterisk jayrod422 (n=jayrod42@pool-173-71-161-106.pitbpa.fios.verizon.net)
19:30.22*** join/#asterisk grnmtn (n=grnmtn@65.183.149.153)
19:30.40dennis00I keep on getting: "You do not appear to have the sources for the 2.6.18-128.1.6.el5 kernel installed.", even though I have installed kernel-headers en kernel-devel.
19:30.44jayrod422hey guys i have a digium analog card.. whenever i call certain numbers and hangup i get a few busy signals and no drop in the loop.. is there a way to tell to dahdi that the channel is free when no more audio is present after so many seconds?
19:31.01jayrod422im running dahdi 2.2.1 rc2
19:31.21ManxPower-workjayrod422: no.  The channel will hangup when you hangup your sip phone.
19:31.36grnmtndumb noob, probably asterisk question: I need to extend and analog phone line over IP - is there a cheap and simple way to do that?
19:31.53citywokgrnmtn: a sip bridge
19:31.55*** join/#asterisk DMeloUK (n=Administ@64.129.95.226)
19:32.00[TK]D-Fendergrnmtn: Linksys SPA-3102 <-
19:32.01citywokthere are quite a few analog to sip bridges
19:32.20ManxPower-workgrnmtn: look up "ATA"
19:32.23ManxPower-work~fxofxs
19:32.24infobotfxofxs is probably An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it.
19:32.34*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
19:32.37grnmtncool - thanks
19:32.42ManxPower-workAn ATA will have some combination of FXO ports and FXS ports
19:33.06jayrod422what if its not a sip phone call? like a loop in a dialplan?
19:33.19*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
19:33.21jayrod422and its waiting for a hangup
19:33.23ManxPower-workjayrod422: then make your loop stop at some point
19:33.29jayrod422lol
19:33.40ManxPower-workIt's standard IVR looping.  This issue is solved EVERY DAY.
19:34.01tzafrirjayrod422, depends. does the provider give you any disconnect supervision?
19:34.30[TK]D-Fendertzafrir: Clearly not
19:34.36jayrod422sometimes
19:34.42tzafririf not, your best hope is to detect those busy signals ("busydetect")
19:35.05*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:35.08jayrod422whenever we call att cellphones the hangups are just busys no drop in voltage
19:35.11jayrod422driving me nuts
19:35.22tzafrirjayrod422, if you're in the US, try using KS signalling
19:36.09jayrod422ok
19:36.11jayrod422let me try that
19:36.22ManxPower-workbusydetect=yes should be "randomlydisconnectmycalls=yes"
19:36.29Kobazheh
19:36.42tzafrirdennis00, rpm -q kernel-devel
19:37.00dennis00kernel-devel-2.6.18-164.10.1.el5
19:37.36*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
19:39.08bmoracadennis00, you may have booted under a different kernel version.  additionally, did you update the kernel version recently without rebooting?
19:39.32dennis00I rebooted after installing kernel-devel and kernel-headers.
19:40.07bmoracayum update kernel, imo...make sure everything's at the latest and greatest.
19:41.25nnywhats the proper formatting to use Gosub in features.conf
19:41.35dennis00I did, still the same message.
19:41.45bmoracadennis00, did you reboot after?
19:41.55dennis00bmoraca: Yes I did.
19:42.03nnyGosub,(context,s,1),MOH ?
19:42.24*** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net)
19:42.52clintcwe just installed a new mediant-1000 gateway... everything is good except for one odd problem - sometimes outgoing callers hear the ring back tone and sometimes they don't - where should we start troubleshooting?
19:43.08bmoracadennis00, what's uname -a say?
19:43.37nnyugh that didn't work
19:43.48ManxPower-workclintc: set progressinband=no in /etc/asterisk/chan_dahdi.conf
19:44.07Qwelldennis00: What exact distro are you using?
19:44.21dennis00CentOS 5
19:44.23clintcManxPower-work: thanks, I take a look at the setting
19:44.33nnyhrrrmmm
19:44.44Qwellwell, you're running a rather old kernel
19:44.56dennis00t
19:44.57ManxPower-workclintc: I doubt it will help, since this is an Asterisk channel, and that was an Asterisk answer.
19:44.57dennis002.6.18-128.1.6.el5 #1 SMP Wed Apr 1 09:19:18 EDT 2009 i686 i686 i386 GNU/Linux
19:45.23Qwelldennis00: you need to install the kernel-devel package for the kernel you are running
19:45.30bmoracadennis00, you're booted to a different kernel version than your sources are fo
19:45.37nnyi'm new to using gosub in features.conf, anyone have a brief example of how it should work? I am not clear on how to pass arguments to it
19:45.39dennis00Yes, I see now, thank you.
19:45.39clintcManxPower-work: it might... I forgot to say that it is our asterisk box that is talking to the gateway :)
19:47.46nnyok six attempts all fail. is this documented somewhere or am I suppose to throw characters at it until I get sick of it and say fucki t?
19:48.08Qwellgosub in features.conf?
19:48.14nnyyeah
19:48.23nnyit says ,[app arguments]
19:48.41nnybut Gosub wants ,app,arguments,etc
19:49.04[TK]D-Fendernny: WTF are you trying to do in there?
19:49.10nnyfuck if i know
19:49.17[TK]D-Fendernny: And what is MOH on the end?  And you should NOT have braces tere
19:49.23QwellWhy are you trying to use gosub from features.conf?
19:49.26nnythere aren't any
19:49.28[TK]D-Fendernny: You don't know WHY you're trying to do a gosub?
19:49.33nnyto fix broking parked calling
19:49.35[TK]D-Fendernny: hold still a sec...
19:49.37nnybroken
19:49.43[TK]D-Fenderreaches for his ClueBat (tm)
19:50.01nnyI am trying to create call parking manually
19:50.12[TK]D-Fendernny: F-ing park it like normal :p
19:50.14nnyif I use macro (which is deprecated) I lose the $EXTEN
19:50.23nny[TK]D-Fender: i'd love to if hints worked for it
19:50.35[TK]D-Fendernny: Who said it doesn't?
19:50.35nny[TK]D-Fender: but obviously that's not a normal feature
19:50.50[TK]D-Fendernny: it is, and documented specifically.
19:51.00nny[TK]D-Fender: wait, i also need the * code
19:51.10nnyer specifically to use the * code
19:51.13[TK]D-Fendernny: for what?
19:51.18ManxPower-worknny: In my experience Gosub won't work with features.conf if you want to use options to it.  the parser seems to be confused.
19:51.19nnyand not xfer -> parking extentsion
19:51.27nny[TK]D-Fender: because people are idiots
19:51.36ManxPower-workUse a Macro then in the macro do the gosub.
19:51.36nny[TK]D-Fender: and SLA is broken and shitty
19:51.45*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
19:51.47[TK]D-Fendernny: If you don't explain this better... I'll lump you in with them :p
19:51.59nny[TK]D-Fender: "Call, Line 1"
19:52.00[TK]D-Fendernny: * SLA is like cake...
19:52.09nny[TK]D-Fender: hmm the cake is a lie
19:52.10dennis00Are a lot of people here using Voipbuster/Betamax?
19:52.14[TK]D-Fendernny: EXACTLY
19:52.30[TK]D-Fenderdennis00: betamax.... you should try VHS instead...
19:52.39[TK]D-Fenderpwned
19:52.47dennis00You know the company Betamax right?
19:52.51dennis00I know it's a poor choice of name.
19:52.52nny[TK]D-Fender: i, in a crack induced haze last night figured BLF + Parking Lots > SLA
19:52.57dennis00Since the last Betamax was a failure.
19:52.59nny[TK]D-Fender: or rather, it's a nice bridge
19:53.13[TK]D-Fendernny: I would never do a drug named after a part of my ass...
19:53.16nny[TK]D-Fender: and the phones I use allow customization to park the call via  button
19:53.38nny[TK]D-Fender: guess you don't want this pipe full of hole than
19:53.59[TK]D-Fendernny: go stock up on KY... you're gonna need it if your trying to turn * into a key-system
19:54.06nny[TK]D-Fender: no shit
19:54.12nny[TK]D-Fender: that's why I am using call parking
19:54.22[TK]D-Fendernny: Same shit, different smell
19:54.22nny[TK]D-Fender: 4 buttons on phone = parking lot, named 1,2,3,4
19:54.29nny[TK]D-Fender: park call, button light up
19:54.40[TK]D-Fendernny: tell them to listen to the #$^ing number.
19:54.42nny[TK]D-Fender: monkey tell monkey 2 pick up "1" etc
19:54.56nny[TK]D-Fender: hey man, you're preachin to the choir
19:55.02*** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110)
19:55.08[TK]D-Fendernny: I *am* the chorus
19:55.09nny[TK]D-Fender: however I have had systems turned down
19:55.13nny[TK]D-Fender: lol
19:55.25ManxPower-worknny: In the macro see ${MACRO_EXTEN}
19:55.47nny[TK]D-Fender: so a small feature like parking + hints + a star code is a small and hopefully convenient work around
19:55.50nnyManxPower-work: my hero stand by
19:56.06nnyI though GoSub was better than macros for features.conf, my bad
19:56.42*** join/#asterisk v00tace (n=v00tace@p4FEC1D58.dip0.t-ipconnect.de)
19:57.16[TK]D-Fendernny: I'm just waiting to hear your users try to park onto the same lot as an existing call and have them drop like flies
19:57.19ManxPower-worknny: It SHOULD be, but the bug I think I found makes Gosub useless in features.conf
19:57.48dennis00What GUI is recommended?
19:57.55[TK]D-Fenderdennis00: vi
19:58.03nny[TK]D-Fender: hmm I am not sure how that would happen, we can expand the lot out. This is just a replacement for transfer, it can scale to as many sidecars/BLFS etc as needed
19:58.04ManxPower-workthe applicationmap does not seem to understand Gosub(context,extension,priority(option1,option2)) it seems to get confused by the options and ()
19:58.13nny[TK]D-Fender: it will still say the number too
19:58.16plundraHmpf, setting up ISDN PRI doesn't feel so easy.
19:58.24dennis00[TK]D-Fender: VI is not a GUI.
19:58.35[TK]D-Fenderdennis00: Its got pixels... don't whine to me boy...
19:58.50dennis00no insult intended.
19:58.54JAMMAN2110nano beats vi for linux noobs
19:59.16JAMMAN2110Especially if they're coming from windows
19:59.16*** join/#asterisk vally (i=kvirc@ip-92-50-112-106.unitymediagroup.de)
19:59.16p3nguinThen those nobs need to learn how to learn.
19:59.18JAMMAN2110ctrl + x = close for example
19:59.23dennis00I used pico a lot.
19:59.25JAMMAN2110Yes p3nguin, I agree, but still
19:59.37beek+1 vi
19:59.51p3nguinIf you can't, or aren't willing, to learn to learn, you're in the wrong industry.
20:00.14p3nguins/willing, to/willing to,/
20:03.58nnyManxPower-work: $MACRO_EXTEN no die
20:04.12nnydice
20:04.53nnycan I pass ARG1 as $EXTEN from the features.conf entry?
20:04.59*** join/#asterisk ruben23 (n=AGENT@122.55.48.243)
20:05.46ManxPower-worknny standby.
20:06.35nnyManxPower-work: http://pastebin.com/m164c5bc2
20:06.40nnyis what I have so far
20:06.49ManxPower-work[applicationmap]report-call-problem => **,self/caller,Macro,report-call-problem,default
20:06.53ManxPower-workdrat.
20:08.46*** join/#asterisk Alagar (n=Administ@122.164.33.195)
20:09.08ManxPower-work[applicationmap]
20:09.08ManxPower-workreport-call-problem => **,self/caller,Macro,report-call-problem,default
20:09.08ManxPower-workIn your macro the number passed to Dial would be in ${DIALEDPEERNUMBER}
20:09.38nnywouldn't that depend on the caller/callee scenario though?
20:09.47nnycan NoOp stand by
20:10.10*** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk)
20:10.19ManxPower-worknny: I don't do anything with the callee, it should put them on hold with "default" MoH class.
20:10.33ManxPower-worknny: "DumpChan" is your friend in the macro.
20:10.41ManxPower-work"core show application dumpchan"
20:10.52ManxPower-workIt's like a SuperNoop
20:10.58v00tacehi guys
20:11.03nnyManxPower-work: well shit
20:11.13nnydial out to cell, use that variable and I get the dialed number
20:11.20v00tacei´m reading at the moment the asterisk book from wintermayer
20:11.25nnywhich means I need to make it a bi directional feature
20:11.28nnysonofabitch
20:11.30v00tacewill install it on a vmware environment
20:11.38v00tacewill it work?
20:11.55nnyso is there a version of asterisk where the normal parking from features.conf doesn't act retarded after a park?
20:12.05ManxPower-worknny: 1.6.1?
20:12.13nnyManxPower-work: this is 1.6.2
20:12.34nnyManxPower-work: so is it only 1.6.1? the problem is right now if I park, and then pick it up and try to park again, it doesn't allow itr
20:12.36nnyit*
20:12.56[TK]D-Fenderv00tace: work=yes, well=depends
20:13.01ManxPower-worknny: what version are you using?
20:13.08nnyManxPower-work: 1.6.2
20:13.10ManxPower-worktry 1.6.1
20:13.21nnyManxPower-work: lol ok
20:13.30*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
20:13.31ManxPower-workThe differences should be minimal for your test case.
20:13.33nnyManxPower-work: I only grabbed 1.6.2 for Originate
20:13.40nnyManxPower-work: does it work in 1.6.1?
20:13.55ManxPower-worknny: Asterisk has historically had very bad and broken parking timeout support.
20:14.01ManxPower-worknny: I don't know if it works in 1.6.1
20:14.12nnyManxPower-work: er Originate not parking
20:14.17v00taceallright, will make a test run
20:14.22ManxPower-worknny: like it being documented to timeout the the caller and ACTUALLY it timing out to "s" extension
20:14.30v00tacebtw. srry i´m also new to irc
20:14.36nnyManxPower-work: noticed that
20:14.44v00tacehow can i send a message with the username infront
20:14.48nnyManxPower-work: or the bug where it freaks out and sets the timeout to 0
20:15.00nnyif ony call parking was a normal feature in phone systems
20:15.00ManxPower-worknny: it's been broken off an on from Asterisk 0.65 onward.
20:15.02[TK]D-Fenderv00tace: try using your keyboard :)
20:15.53*** join/#asterisk jefimenko (n=jordan@unaffiliated/jordanl)
20:16.40*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:16.48ManxPower-workv00tace: DO NOT EVER MESSAGE ME
20:16.53ManxPower-workEVER!
20:16.58v00tacesrry man
20:17.07v00tacelike i said, im new to irc
20:17.40sbrathIf I have a Asterisk behind a iptables fw ( on a seperate box ), and all I want is outbound SIP, do I still have to jump thru the SIP/NAT hoops.
20:17.50jefimenkomy SIP invites are going out with the loopback IP (127.0.0.1) in the SDP, the Via header, and as the domain part of the Contact header. are there any settings in asterisk 1.4 that would be doing this?
20:18.17nnyso I guess the question is, doesn parking via features.conf allow you to repark in ANY version of asterisk?
20:18.24nnydoes*
20:18.27ManxPower-worknny: parking IS a standard feature in PBXs.  It's not a standard feature in KEY SYSTEMs
20:18.42ManxPower-worknny: It did when my users were doing it in 1.2.mumble
20:18.51nnyManxPower-work: yeah this is a PBX
20:19.01nnyManxPower-work: i am using parking ONLY to keep the key system monkeys happy
20:19.11ManxPower-workNortels part to....extension 700 by default
20:19.13nnyManxPower-work: and if it's a simple button press, it does the job
20:19.42nnyManxPower-work: I am not clear as to the issue
20:20.06ManxPower-worknny: It may be tough to do, but if you let the Key System monkeys win this battle, you'll lose the war and spend all your time trying to make the Key System monkeys happy.  Asterisk is NOT a Key System and cannot be setup to be a key system.
20:20.21nnyManxPower-work: if I tranfser to parkedcalls, is the kK inherent
20:20.37nnyManxPower-work: but if I use the feature code, it doesn't pass the kK?
20:20.40ManxPower-workkK are Dial opts.  When the Dial exits.....
20:20.54nnyManxPower-work: so how is it that xfer to the xtension and dial it back allows reparking
20:20.58nnyor is that broken too?
20:21.09ManxPower-worknny: I have not parked a call in 4 years.
20:21.32nnyManxPower-work: it doesn't need to be a key system, it only needs to allow the user to grasp the concept of transfer that isn't bxfer or xfer
20:21.34ManxPower-workSo I can only tell you what has happened historically with Parking
20:21.47nnyManxPower-work: picture this, mechanic in shop, wants to grab call in back
20:22.02nnyhow would you handle this without parking?
20:22.11nnynow, I already tell them park, redial
20:22.35nnybut shit, if the phone can have a BLF, than it's easier for them to understand, especially when all older phone systems had something similar
20:22.55nnyI am not lying to them and telling them that's like 1
20:22.57nnyline 1*
20:23.02ManxPower-worknny: you can't do it without parking.
20:23.09nnyManxPower-work: exactly
20:23.28ManxPower-workSo no matter what you do you will fail.
20:23.35nnyI didn't think turning it into a star code would have been so much work when it's suppose to be listed as a feature
20:23.40nnyManxPower-work: how do you figure?
20:23.57ManxPower-worknny: You are trying to provide a feature to the user that is not supported by Asterisk.
20:24.02nnyManxPower-work: ...
20:24.10nnyManxPower-work: Call Parking isn't supported??
20:24.18nnycause that's what i am doing
20:24.23ManxPower-workThey bought a car without power windows and you are hacking up some motors to stick in the door.
20:24.32nnybad analogies
20:24.35ManxPower-worknny: No.  Key System hold/pickup is not support in Asterisk.
20:24.41nnyUGH
20:24.45nnyi am doign call parking
20:24.53nnyif I wanna call it fist fucking the fruit fly, it's still call parking
20:25.10nnyi am not even gonna lie to them
20:25.12ManxPower-worknny: NO, you are using call parking to try to EMULATE/FAKE  hold/pickup.
20:25.17nny??
20:25.23nnyso BLF with call parking isn't call parking?
20:25.30nnydrop the whole key system thing
20:25.52nnyI am not emulating or faking anything, I am just planning how call parking can be used easier, i.e. through a BLF
20:26.19ManxPower-worknny: I wish you the BEST of luck.
20:26.22[TK]D-Fendernny: You could jsut BLF the lots
20:26.37nny[TK]D-Fender: yeah that's what I am trying to do
20:26.51nny[TK]D-Fender: I was also trying to use the * code as listed in features.conf
20:26.59nnywhich is obviously broken
20:27.07ManxPower-workBTW, does anyone know in 1.4 without resorting to mpg123 of a way to not restart the hold music when a call goes off hold and is put back on hole?
20:27.09ManxPower-workhold?
20:27.24[TK]D-Fendernny: Nope
20:27.42nny[TK]D-Fender: so using the * code in features.conf is suppose to allow multiple parks?
20:27.47leifmadsenManxPower-work: in 1.4 I don't think so -- think that was a feature added in later versions
20:27.57nnyat tihs point I may just say hey, it only works once, sorry
20:28.04ManxPower-workleifmadsen: thanks.
20:28.19[TK]D-Fendernny: huh?
20:28.32ManxPower-worknny: or find an Asterisk version without the BUG.
20:29.00nnyYeah I had asked if anyone knew of one earlier, may have to resort to compiling each one and testing
20:29.03nnysounds efficient
20:29.16ManxPower-worknny: That is was I was recommending.
20:29.17leifmadsennny: you could keep backing up revisions until you find where the bug was introduced and then file a bug report so it could get fixed too
20:29.30ManxPower-workleifmadsen: it will just get broken again. 8-|
20:29.38nnyleifmadsen: yeah I jumped from 1.4.2X to 1.6.2 yesterday for Originate
20:29.56ManxPower-worknny: did it work in 1.4?
20:29.56nnyI need to figure out if the two overlap somehow
20:29.56nnyunknown
20:30.03leifmadsenManxPower-work: what a productive comment
20:30.06nnyI never tested double park
20:30.35nnylol double parking: the bane of all existence in this year of our lord, 2010
20:30.54nnybrb smoke, i'll start slipping back some versions after 5 on my box
20:30.55jefimenkomy SIP invites are going out with the loopback IP (127.0.0.1) in the SDP, the Via header, and as the domain part of the Contact header. are there any settings in asterisk 1.4 that would be doing this?
20:31.02nnyand document the differences
20:31.18leifmadsenjefimenko: localnet and externip or externhost?
20:31.22ManxPower-workjefimenko: remove bindaddr=
20:35.05*** join/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com)
20:44.33*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
20:46.18tzafrirManxPower-work, you don't need mpg123 for that. You can use the "custom" method and provide your own stream
20:46.47ManxPower-worktzafrir: Would that have to be a URL stream?
20:47.13tzafrirYou provide a program whose standard output is a PCM stream
20:47.36tzafrirsox can do that, with proper wrapping
20:47.41ManxPower-worklike mpg123. 8-|  IIRC there are others
20:48.06tzafrirothers that won't waste as much CPU, if you use wav files
20:48.34ManxPower-worktzafrir: I think I'll stick to my official statement to the customer "No, that's not supported"
20:49.02ManxPower-workI deleted the part where I told him he's a nutjob for wasting my time.
20:49.05tzafrirAt 1.0 we made a sox-based script our default MoH provider
20:49.11ManxPower-workGood customer service! 8-)
20:52.15nnyleifmadsen: hmm heh. transfer to parking lot in 1.6.2 puts the caller in the lot
20:53.51sbrathany help on SIP from behind a NAT? I can establish the call, but their is no audio.
20:53.58[TK]D-Fender~sipnat
20:53.59infobot[~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
20:54.02[TK]D-Fender^^^^^ read
20:54.37sbrathok, for the record I asked infobot, but I asket "sip nat" and that wasn't the keyword.. :)
20:56.21jayteeI want to downgrade one of my asterisk servers back to 1.4.x with zaptel. I've removed all the dahdi compiled modules and deleted the service but other than deleting all the asterisk 1.6.x modules in /usr/lib/modules/asterisk are there any other files I need to remove before I do a make install for zaptel and asterisk 1.4?
20:57.11[TK]D-Fenderjaynope
20:57.17leifmadsenjaytee: nope, just the /etc/asterisk stuff
20:57.19jayteecool
20:58.53jaytee[TK]D-Fender and leifmadsen, thanks. Other than the chan_dahdi.conf and some minor changes in iax.conf and extensions.conf I know I can reuse the rest of the files in /etc/asterisk. just wasn't sure if there were any other binaries out there I needed to kill.
20:59.10leifmadsennope, installing will copy over that stuff
20:59.38beekjaytee: I'm curious, why the downgrade?
20:59.50eppigyTRABAJO
21:00.02nnyleifmadsen: ok 1.6.1.12 seems to have it too
21:00.16jayteeI'd imaged the server with the full working 1.6.x install on it so I can restore that when I'm ready to actually upgrade the production system. My boss wanted to hold off till late spring as our network engineer screwed up some network changes in mid-december and he doesn't want to risk the embarrassment of any other problems for awhile.
21:00.52leifmadsennny: right, changes get merged between versions -- so at this point, step back releases until you find one that doesn't have the bug, and then you'll have narrowed down the change between two versions. Then you can look at the log files to find likely candidates for a commit that broke it.
21:01.06leifmadsennny: if you can't find where it broke, then it is likely a bug that has existed for a long time and you are the first one to find it
21:01.18jayteebeek, it was just my standby server that I'd upgraded to 1.6 to test. it was never in production but i want it available as a failover for the next few months as that's it's role 6 months out of the year.
21:01.37nnyleifmadsen: just so I understand it s,n,TrySystem(asterisk -rx "channel originate SIP/${CELL2}@vitel-outbound application Bridge ${CHANNEL}") won't work in 1.4.X right?
21:01.40beekAhhh.. you're running 1.4 in production.
21:01.48jayteeyes
21:01.53beekGotcha
21:01.56leifmadsennny: since Bridge() doesn't exist in 1.4, correct
21:02.00nnyleifmadsen:  mainly due to originate being an app only in 1.6?
21:02.11nnyleifmadsen: ahh i thought it was originate that wan't in 1.6
21:02.13nnyer 1.4
21:02.17leifmadsennny: that too
21:03.18jayteebut I want to move to 1.6 to get rid of my sipX VM that acts as a sipproxy for udp to tcp for communicating with Exchange UM. I tested 1.6.1.10 with Exchange and it works fine. I'll test 1.6.12 or whatever 1.6.1.x is current again when I'm ready to move it to production so
21:03.42nnyleifmadsen: yeah core show app on 1.4.27.1 shows neither
21:03.52leifmadsennny: right... like i said... :)
21:03.58nny:)
21:04.04nnywas just checking :D
21:06.00jefimenkoManxPower-work: bindaddr=0.0.0.0
21:06.16*** join/#asterisk hfb (n=hfb@pool-98-112-226-53.lsanca.dsl-w.verizon.net)
21:07.02titterAnyone see anything wrong with this ... http://pastebin.com/d588e8ef5
21:07.45titterI do not get the email dispatched, but if I run the /usr/bin/fax2mail manually, I get the e-mail instantly
21:08.06[TK]D-Fendertitter: Go show us the actual fax attempt and we'll show you why
21:10.14titter[TK]D-Fender: http://pastebin.com/d119243cf
21:10.31titterv is at 3, can go higher if needed
21:10.51titterNo warnings. I was getting a warning when I have single quotes around the command
21:11.04titterI removed them, no warnings, but no e-mails
21:11.48[TK]D-Fendertitter: you'll have to escape the " 's you put in there.
21:12.05*** join/#asterisk mag-samara (n=mezhuev@81.211.87.226)
21:12.13nny[TK]D-Fender: ok got your idea to work at least
21:12.13titter\" then
21:12.44nnyso I have a work around for parked calls, that allows both *5 and xfer to 700 to both re park calls after
21:13.21nnybasically using both the asterisk features.conf star code to park the calls, as well as a unique context for picking them up and parking them via the transfer method
21:16.46[TK]D-Fendercheckout time... later all
21:16.55*** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195.dyn.user.ono.com)
21:17.23Godfather_o/
21:24.44*** join/#asterisk Alagar (n=Administ@122.164.39.153)
21:25.09devmodany ideas on how to create video prompts like these using open source tools? http://www.aupix.com/index.htm?/support/astvideo.htm Any specific requirements for asterisk while encoding?
21:27.03ChannelZhttp://www.passiveaggressivenotes.com/2009/12/30/wifi-for-passive-aggressives/
21:28.09jayteealmost quittin time, be back later
21:29.22rocksfrowdoes anybody have any suggestions on enabling support for some sort of webpage viewable sip client?
21:29.30rocksfrowlike, a java applet for example?
21:30.01rocksfrowi'm curious about setting up a system for customer service where they use primarily the computer, with a headset, vs using an actual hard phone
21:30.31carrarxlite?
21:30.56carraruse a xlite softphone
21:31.07devmodxlite is so crashy tho
21:31.25carrardoesn't crash for anyone I know
21:31.31*** join/#asterisk Erestar (n=erestar@c-98-236-90-228.hsd1.wv.comcast.net)
21:32.08rocksfrowcarrar: you're speaking of this? http://www.counterpath.com/active-x.html
21:32.13devmodtheir latest beta is awful
21:32.34*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
21:32.37carrarhttp://www.sipquest.com/x-lite.html
21:32.44devmodI do use video tho, maybe it behaves better with audio alone
21:33.00carrarkey word there is beta
21:33.00carrarheh
21:33.01cusco_hi
21:33.13nnyleifmadsen: fyi http://pastebin.com/m5c74d38d is my fix for now
21:33.14cusco_is this iax2 dial correct?
21:33.16cusco_<PROTECTED>
21:33.23nnyso far works as expected
21:33.24devmodgmail was on beta for a long time and it was the best email client out there
21:33.26devmodhaha
21:33.46nnyi still wanna file a bug report, not sure how yet
21:33.46carrarsure if you like your emails read
21:33.58cusco_im getting:
21:34.03devmodi dont use email for sensitive matter :)
21:34.09cusco_http://paste.debian.net/56520/
21:34.15cusco_IAX2/primario-iax-779 is circuit-busy
21:36.16devmodwhen using app_konference, will the video switching feature work only when using speex?
21:36.40*** join/#asterisk Mango_ (n=Mango@d154-20-96-193.bchsia.telus.net)
21:36.43*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
21:36.43*** mode/#asterisk [+o malcolmd] by ChanServ
21:42.57p3nguincusco_: Dial(IAX2/1020304@voipms/1202${EXTEN})
21:43.18*** part/#asterisk Mango_ (n=Mango@d154-20-96-193.bchsia.telus.net)
21:43.21*** join/#asterisk Mango_ (n=Mango@d154-20-96-193.bchsia.telus.net)
21:43.54p3nguincusco_: ...where voipms is an iax2 device defined in iax.conf
21:43.59Mango_Security question.
21:44.12Mango_In my sip.conf, I have defaultcontext set to "incoming'.
21:44.31Mango_How many ways could a caller get outside the "incoming" context?
21:44.52cusco_thats exatcly how I have it
21:45.04cusco_hold..
21:45.09p3nguinGoto, Gosub, Dial, include, et cetera
21:45.33cusco_yes
21:45.34cusco_thats it
21:45.41cusco_what is wrong?
21:45.46cusco_p3nguin: have you seen the paste?
21:45.50p3nguinnegative
21:45.58p3nguin56520?
21:46.01cusco_yes
21:46.26Mango_p3nguin: Okay.  Assuming what they are trying to call does not match in the "incoming" context, what can happen?
21:46.34p3nguincusco_: You're doing it wrong.
21:46.42cusco_err
21:46.43cusco_why?
21:47.27p3nguincusco_: What is the iax peer name?  What is the username on that peer?
21:47.43cusco_my softphone (1000) dials to asterisk, and asterisk should dial primario-iax
21:47.59cusco_the asterisk 1000 is dialing is gateway-iax
21:48.04p3nguinmango_: It'll go somewhere else, I guess.  You'll have to provide more information.
21:48.11cusco_so gateway-iax dials primario-iax
21:48.28cusco_iax2 show registry shows both asterisk boxes registered one on the other
21:48.47p3nguincusco_: Let us try again.  What is the peer name which appears in iax.conf?
21:48.58bmoracaMango_, your default context, or the context under [general], is the context where unauthenticated calls are dropped off
21:49.21cusco_I have two asterisk boxes...
21:49.33cusco_the one that has this dial config
21:49.39p3nguincusco_: For now, you are on one.  I want the iax peer name of the OTHER BOX.
21:49.51cusco_ok
21:49.55bmoracacusco_, what are you trying to do with this dial string here?  iax2/primario-iax:pcmedic/90210332319
21:50.16cusco_[primario-iax] //  type=friend // secret=pcmedic // host=dynamic // context=incoming-pri
21:50.16bmoracawhat is the significance of primario-iax and pcmedic?  i assume the last is the number you wish to dial
21:50.24batphonedont you hate it when you accidentally cc someone who you are talking about?
21:50.26Mango_bmoraca: Interesting.  Under [general] I have [globals] and then [devices].  Devices is for SIP phones on the LAN.  That's probably a bad thing I take it? :)
21:50.34batphonei think i do that about once a year
21:50.55p3nguincusco_: And the username for you which the other box recognizes you as?
21:51.04cusco_gateway-iax
21:51.14cusco_same conf
21:51.53p3nguinThen try Dial(IAX2/gateway-iax@primario-iax/numbertodial)
21:51.56bmoracacusco_, iax2/username:password@host/number  it appears you're missing your host.
21:52.07cusco_ok
21:52.30p3nguinAs I understand it, you don't even need the password since you have a peer config for it.
21:52.59p3nguinDial(Tech/username@peername/numbertodial)
21:53.09bmoracathat would be correct...and if your peer is registering to you or you are registering to it, you don't need the host either
21:53.39p3nguinNo, that's not right.  Dial() still has to know where to send the call.
21:54.08p3nguinOh, you're talking about host as opposed to using a peer name.
21:54.32p3nguinI think I'm on the same page, now.
21:54.36bmoracaMango_, in sip.conf, you have [general] and you have your peer contexts...
21:54.40cusco_http://paste.debian.net/56521/
21:55.01cusco_using: Dial(iax2/gateway-iax@primario-iax:pcmedic/90${EXTEN});
21:55.07p3nguinstill wrong
21:55.19p3nguinIf you would follow what I have said, it would be working.
21:55.23bmoracacusco_, Tech/user:pass@host/number
21:55.25p3nguin(1552.58) <p3nguin> Dial(Tech/username@peername/numbertodial)
21:55.51p3nguinSo drop the :pcmedic entirely.
21:56.04p3nguinYou have a peer configured for it with the secret.
21:56.20Mango_bmoraca: Oh.  I think I misunderstood you.  I get it now - thanks :)
21:56.33cusco_right.. taht works p3nguin
21:56.38p3nguinIf your peer configuration failed, then use bmoraca's method of using a full host name instead of the peer name.
21:56.44cusco_I would think the password would not make a difference
21:56.54cusco_thanks p3nguin bmoraca
21:57.07bmoracacusco_, makes a big difference if you put it in the wrong place, like you have been
21:57.09p3nguinWell, @peername:secret is invalid syntax, so it won't work.
21:57.19p3nguincan't work
21:57.29cusco_thanks
21:58.36nnylol wth
21:58.40nny1.6.2 "asterisk -rx "channel originate SIP/8432980648@vitel-outbound application Bridge SIP/vitel-outbound-00000053"") in new stack
21:58.52nny1.6.1.12 apparently channel originate wasn't it
21:58.56nnyseriously?
21:59.09nny1 version difference and it's changed?
21:59.24p3nguinWould that really surprise you?
21:59.27nnyi think the devs have ADD
21:59.30nnyno ha
21:59.41nnyso wonder what the magic is for 1.6.1.12
21:59.59*** join/#asterisk etfonhomey (n=etfonhom@173-8-16-185-WashingtonDC.hfc.comcastbusiness.net)
22:00.01bmoracanny, what's worse is that the "reload" command doesn't exist in 1.6.2
22:00.27nnybmoraca: i think it's module reload now
22:00.40p3nguinhas been since 1.4
22:00.44nnyso anyone know how to correct "asterisk -rx "channel originate SIP/8432980648@vitel-outbound application Bridge SIP/vitel-outbound-00000053"") in new stack
22:00.46bmoracanny, right...to reload a single module.  i'm talking about a global reload of all configurations
22:00.51nnybmoraca: oh yeah
22:01.16nnyor is that 1.6.2 only
22:01.24nny(channel originate)
22:03.02bmoracanny, "core show help channel originate"...that said, i'm not quite sure what your problem is
22:03.15nnybmoraca: no such command in 1.6.1.12
22:03.36bmoracai believe 1.6.1 might just be "originate" and not "channel originate"
22:03.44bmoracatry "core show help originate"
22:03.47nnybmoraca: you win!
22:03.49nnythanks :D
22:04.08nnyso.. what's considered stable 1.6?
22:04.09bmoracaor, rather, "help originate"
22:04.14nnywhich version should I plan to implement?
22:04.46bmoracai'm using 1.6.2 on my highest volume box...and aside from some weird quirks about config syntax and commands moving around and stuff, it works great
22:05.16nnybmoraca: good to know, may freeze at 1.6.1.12 and stay one back for the moment, unless that
22:05.20nnythat's riskier heh
22:05.57nnyi was on 1.4.X but originate and bridge allowed me to create a seamless xfer to cell phones, so would like to keep it in my production boxes if it's sane
22:06.17jdoe1.6.1.6 to 1.6.2 was painless for me, but admittedly my config is pretty straightforward.
22:06.20bmoracaas long as you patch to the latest, you shoudl be fine
22:06.45bmoracalol...congress has a 27% approval rating
22:06.59jdoebmoraca: more amusingly, that's an improvement.
22:07.33nnybmoraca: eh pacthing may not be good either, 1.6.2 it is :D
22:07.54bmoracano reason not to...
22:08.08nnybmoraca: yeah
22:08.19bmoracaof course, i don't use dahdi for anything but dahdi_dummy on this box, so i can't comment about how well it works/stability/etc
22:08.32nnyyeah on my box I am using dummy as well
22:08.37nnyuse wanpipe for everything else
22:08.41nnyI consult sangoma tomorrow
22:08.49nnyI'll
22:08.50dennis00telnet: connect to address 127.0.0.1: Connection refused << Even though Asterisk is running...
22:08.50Nuggettelnet is eeeeeeevil!
22:08.55bmoracawell, wanpipe goes through dadhi to get to asterisk...sooo...
22:09.03nnytru enough
22:09.55bmoracawanrouter does have some good debug tools, though
22:10.20bmoracahowever, the latest digium cards are just as good as sangoma's cards, so i don't bother with the sangomas anymore
22:10.49nnybmoraca: i hear ya, got burned on a T1 with EC so...
22:12.51ErestarAnyone here have experience with generating large amounts of .call files and queuing them up to be moved to the spooler based on available resources?
22:13.15bmoracaErestar, very hard to coordinate.  best bet is to use AMI for origination
22:14.50*** join/#asterisk TheDavidFactor (n=thedavid@nc-71-0-16-133.dhcp.embarqhsd.net)
22:15.10Erestarbmoraca, Thanks. Have any experience with AJAM ?
22:15.47bmoracaErestar, nope.  the AMI protocol is simple enough that i've always just used sockets to accomplish what i needed.
22:15.53*** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be)
22:16.11TheDavidFactorhey guys, anyone on here know much about the new say.conf options?
22:17.32*** join/#asterisk Dksaarth (n=dksaarth@196-210-210-184.dynamic.isadsl.co.za)
22:17.43DksaarthHi all
22:17.49Dksaarth<PROTECTED>
22:18.00Dksaarthbut getting circuit congested messages when trying to use it
22:18.22Dksaarth(i.e. after setting the max number of calls on the first trunk to 0 - it falls over to the second channel which reports congested / busy)
22:18.33Dksaarthlooking at the pri debug
22:18.40Dksaarthi see the setup acknowledgement message
22:18.46Dksaarth<PROTECTED>
22:18.56Dksaarthdoes this mean that there is a problem with the pbx/box that this line is plugged into ?
22:19.04Dksaarth(the cause of the disconnect message is congestion)
22:19.20Erestarbmoraca, Yeah, I guess there's no sense going over http. So the general idea would be to use the AMI to monitor SIP connectivity and then move .call files in when available?
22:20.46titterAnyone see why my bash script on line 13 doesn't exec? If I copy/paste outside of the CLI it works. http://pastebin.com/d166d227f
22:22.50carrarcheck your enviroment variables
22:23.24bpgoldsbI've loaded chan_alsa.so.  When I try to 'Dial(Console/alsa);' I get 'alsa_request: Already have a call on the ALSA channel'.  However, since no one but me is using asterisk, I don't buy that.  alsa.conf has 'context=intercom'.  Anyone have any ideas why I'm getting the error?
22:24.04carraris everything properly escaped?
22:24.29tittercarrar: I believe so
22:26.02*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
22:30.22*** join/#asterisk TheDavidFactor (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net)
22:32.47*** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl)
22:38.54*** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de)
22:43.33*** part/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com)
22:43.47*** join/#asterisk Erestar (n=erestar@c-98-236-90-228.hsd1.wv.comcast.net)
22:45.37*** join/#asterisk gmarsh (n=gmarsh@mobile-166-137-137-214.mycingular.net)
22:45.56*** part/#asterisk gmarsh (n=gmarsh@mobile-166-137-137-214.mycingular.net)
22:46.23ehsjoar:q
22:46.27*** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
22:47.38Dksaarthanybody here know about my pri disconnect message ?
22:49.20*** join/#asterisk ebroad (n=EB@72.11.213.195)
22:49.31ebroadCorydon76-dig, ping
22:50.24*** part/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net)
22:52.58*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au)
22:53.39ManxPower-workDksaarth: What is the value of HANGUPCAUSE after the failed Dial?
22:55.36Dksaarth<PROTECTED>
22:56.16Dksaarthi don't seem to have a HANGUPCAUSE in the log
22:56.23Dksaarthwhat should I add to get it ?
23:00.55ManxPower-workDksaarth: Did you write your dialplan?
23:01.25ManxPower-work(A GUI could have written it, someone else could have written it, etc)
23:01.29DksaarthI didn't, a collegue did -
23:01.36Dksaarththe first card works
23:01.39ManxPower-workThen you may not understand.
23:01.49Dksaarththe second card doesn't, so i thought i didn't have to look at it
23:02.12ManxPower-workLikely your /etc/asterisk/zapata.conf or /etc/asterisk/chan_dahdi.conf does not include the channels of the 2nd card in the dial group.
23:02.32ManxPower-workDksaarth: turn off pri debug and any other debug, run at verbose 3 and pastebin the cli output of a failed call.
23:02.34ManxPower-work~pb
23:02.35infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
23:03.00ErestarCan someone clear something up for me? I'm doing a new asterisk box and I haven't configured one for a while. For a Digium TDM card, I still want to compile the zapata drivers, but I configure them in asterisk with the dahdi files? Is that right?
23:03.22ManxPower-workErestar: no.
23:04.16titterin System can I include a |? For instaance System(echo "Blah" \| mail -s "mer" email@email.com)
23:04.24titterIt works, but I get a warning
23:04.36titter[Jan 12 15:01:07] WARNING[31096]: pbx.c:953 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan?
23:04.36ManxPower-worktitter: ignore the warning
23:05.04ErestarManxPower-work, Mind elaborating?
23:05.16ManxPower-workErestar: use dahdi or use zaptel.
23:05.33ManxPower-workMaybe you are reading the docs for some GUI that might recommend that, but we don't do GUIs here.
23:05.58ManxPower-workIf a GUI recommended that, it would be ONLY because of limitations of that GUI.
23:06.35titterManxPower-work: thanks ... anyway to just get rid of the warning ... it annoys me.
23:06.59ErestarManxPower-work, Thanks :)
23:07.21ErestarThe asterisk book is from 2007 so I'm feeling a little rudderless here
23:08.08ManxPower-worktitter: yes, but I don't remember how.  Maybe it was documented in the UPGRADE*.txt files that came in the Asterisk source.
23:09.38titterManxPower-work: thanks again, I'll check. also got that fax stuff working I was blabering about yesterday ... fax2email on the fly based on incoming DID. <3 *
23:09.38dennis00so, that's what they call a netsplit right?
23:10.26titterlol
23:10.33*** join/#asterisk denon (n=denon@synapse.subneural.net)
23:10.33*** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk Erestar (n=erestar@c-98-236-90-228.hsd1.wv.comcast.net) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk jmacz (n=jmacz@190.144.75.22) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk mag-samara (n=mezhuev@81.211.87.226) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk roe_ (n=roe___@216-164-215-125.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk jdoe (i=jdoe@falseprophet.ca) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk jfreema (n=root@hpbx-1.peace-communications.net) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br)
23:10.33*** join/#asterisk ibolmo (n=ibolmo@cpe-67-11-89-134.rgv.res.rr.com) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net)
23:10.33*** join/#asterisk ayrjola (n=ayrjola@gw.ccxtech.fi)
23:10.33*** join/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk eddyo (n=edd@91.72.212.165) [NETSPLIT VICTIM]
23:10.33*** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) [NETSPLIT VICTIM]
23:10.34*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)
23:10.34*** join/#asterisk kc8pxy (n=gecko@75-145-57-201-utah.hfc.comcastbusiness.net) [NETSPLIT VICTIM]
23:10.34*** join/#asterisk mrbnet_ (n=ryanbant@c-75-73-142-28.hsd1.mn.comcast.net) [NETSPLIT VICTIM]
23:10.34*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
23:10.34*** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-pxmeyohcxqmjzjqv)
23:10.34*** join/#asterisk fenrus_ (i=fenrus@oklart.com) [NETSPLIT VICTIM]
23:10.34*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk |Rain| (i=rain@ev.il.net) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk mchou (n=quassel@unaffiliated/mchou) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk Raden (n=Raden@71.89.121.119) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk dzup (i=dzup@support.team.at.shellium.org) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk mr_ian (n=mr_ian@S0106001b63f49383.du.shawcable.net) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk oldhack (n=jfincher@cpe-24-27-56-221.austin.res.rr.com) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk MmixX (i=mixed@61.14.191.133) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk dahl (i=dahl@4.84-234-222.customer.lyse.net) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk bmoraca (n=bmoraca@66-242-174-254.ceres.bvn.net) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk memph (n=memph@sd1438.sivit.org) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk rbd_ (n=rbd@rrcs-98-101-33-14.midsouth.biz.rr.com)
23:10.35*** join/#asterisk zamba (i=marius@flage.org) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) [NETSPLIT VICTIM]
23:10.35*** join/#asterisk keith4 (n=keith@unaffiliated/keith4) [NETSPLIT VICTIM]
23:10.35*** mode/#asterisk [+oo denon leifmadsen] by irc.freenode.net
23:10.36Godfather_:-d
23:11.07*** join/#asterisk Mango_ (n=Mango@d154-20-96-193.bchsia.telus.net) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net)
23:11.07*** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk Amorsen (n=Amorsen@94.127.50.7)
23:11.07*** join/#asterisk jasonwert (n=jasonwer@97-83-97-13.dhcp.trcy.mi.charter.com) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk)
23:11.07*** join/#asterisk t (i=tom@freenode/staff/tomaw)
23:11.07*** join/#asterisk xuser (n=xuser@unaffiliated/xuser) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1242515082.dsl.bell.ca) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk steve (i=steve@bouncer.stephen.marsh.name) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk MuffinMan (n=muffinma@asterisk/issue-tracker-bot/muffinman)
23:11.07*** join/#asterisk slima (i=slima@unaffiliated/slima) [NETSPLIT VICTIM]
23:11.07*** join/#asterisk Gido-E (n=gido@lounge.datux.nl) [NETSPLIT VICTIM]
23:11.08*** join/#asterisk miloux (n=KVIrc@milu.rit.se) [NETSPLIT VICTIM]
23:11.08*** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) [NETSPLIT VICTIM]
23:11.08*** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) [NETSPLIT VICTIM]
23:11.08*** join/#asterisk Obeliks (i=obeliks@gentoo/contributor/Obeliks) [NETSPLIT VICTIM]
23:11.09*** join/#asterisk Dksaarth (n=dksaarth@196.210.210.184)
23:11.17DksaarthManxPower-work: http://pastebin.com/d3e23bc3e :)
23:11.22outtolunche probably could have just put dontwarn = yes in [options] section of asterisk.conf
23:12.48ManxPower-workDksaarth: I cannot help you with a HUI
23:12.51ManxPower-workor a GUI
23:12.54ManxPower-work~freepbx
23:12.55infobot[~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, or fpbx
23:13.09titterManxPower-work: Thanks again, I'll check. also got that fax stuff working I was blabering about yesterday ... fax2email on the fly based on incoming DID. <3 *
23:14.25[TK]D-FenderDksaarth: You got an ISDN 34 which either means the target phone # is "busy", or your carrier is congested and can't process your call
23:15.15Dksaarth[TK]D-Fender thanks - i think this is a problem with the other side i am plugged into
23:15.21Dksaarthi will try follow it up with them
23:15.47nnyis there a variable for CHANNEL that would be the OTHER channel in a two way conversation?
23:15.54titterouttolunc: dontwarn=yes does not work
23:15.57nnyor some clever way to grab the information?
23:17.11ManxPower-worknny: still in your IN CALL feature?
23:17.21nnylike say I am 190/00000001 and I am talking to 191/00000000222 can I pull the variable for 191/000000022
23:17.26nnyManxPower-work: how'd ya guess
23:17.37ManxPower-workDumpchan will gump to the display all variables available at that point in the dialplan....
23:17.39nnyManxPower-work: side note, can't use peer/self in features.conf
23:18.07nnyManxPower-work: thanks will try it
23:19.51outtolunctitter: what version of asterisk?
23:20.14nnyDumpChan is sexy, didn't get a chance to try it the first time, thanks
23:20.20titter1.6.1.1
23:20.55ManxPower-worknny: it is the single most useful app for debugging weird stuff like executing diaplan during an active dial
23:21.00nnyindeed
23:21.33nnylooks like BRIDGEPEER is my ticket
23:21.59nnyhot damn
23:22.22*** join/#asterisk Mark22 (n=mark@unaffiliated/mark21)
23:23.17Mark22Hello, is it possible with Dial() to do something like this: call at trunk1 if that fails try trunk2 (fails = doesn't answer, so no one does pick up the phone)
23:24.52*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
23:25.07Mark22I now have: http://yourpaste.net/4491/ (note: yy are 2 numbers and xx are 2 numbers)
23:25.23Mark22portech01-trunk01 is trunk1 in this example
23:26.12Mark22now after the call using the portech01-trunk01 is stopped (after being answerd) it also calls the same number using trunk-05371127yy
23:27.35ManxPower-work~HANGUPCAUSE
23:27.36infoboti heard hangupcause is Q.931 <-> SIP Codes can be found at http://www.faqs.org/rfcs/rfc3398.html
23:27.51ManxPower-workthat is not what it's supposed to say
23:28.15ManxPower-workMark22: See: HANGUPCAUSE dialplan variable info (possibly outdated) at http://www.voip-info.org/wiki/index.php?page_id=569
23:31.17nnywhats the simplest way to play something to only the person who originated a feature code (custom) in the macro itself?
23:31.24*** join/#asterisk vitaminx (n=vitaminx@89.130.31.1)
23:31.56nnyhttp://pastebin.com/m1fed759e is the fineshed cell switch macro btw
23:32.02nnyfinished
23:32.12*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:32.20nnyjust going to have it play a beep or something so the person activating it knows it's in progress
23:33.03nnyeh minus the top NoOp
23:34.17*** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net)
23:34.18Mark22ManxPower-work: thank you, I'll look at that to find the best solution for my problem
23:34.27bpgoldsbIf I want to have an extension that does Answer; Record;, then on hangup, plays back the recording, whats my best option?
23:34.42ManxPower-workbpgoldsb: hangup after you play the recording
23:35.16bpgoldsbFile won't record because the channel was hungup during recording.
23:35.25bpgoldsbSo Record() fails
23:35.37bpgoldsbWe were previously doing it with DeadAGI
23:36.27*** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com)
23:36.35Mark22bpgoldsb: did you try something like: http://yourpaste.net/4492/
23:37.21bpgoldsbMark22: That would work, but not in the case I specified.  The problem is they want the user to be able to hang up instead of pressing '#' to finish the recording.
23:38.33bpgoldsbhttp://yourpaste.net/4493/ is what I was going to use, but since the record doesn't save the file on hangup, it doesn't work
23:41.51Mark22sorry, I don't know a solution for that at this moment (but maybe it is possible with hangupcause)
23:43.10*** join/#asterisk Iamnacho (n=Iamnacho@ip174-70-146-115.ks.ks.cox.net)
23:43.13ManxPower-workbpgoldsb: the playback will never execute because the Dialplan will not continue until the Dial to the console hangs up.  You have a basic and significant misunderstanding of how the dialplan works.
23:44.47bpgoldsbManxPower-work: You're correct on the first part.  And looking at it now I see why it wouldn't.  but since the first part isn't working, I haven't even touched the 'h' extensions code to make sure it works
23:46.49outtolunctitter: 1.6.1.12 pbx.c line 955  ..  && !ast_opt_dont_warn) {  so it should work
23:47.36*** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com)
23:47.37titterouttolunc: dontwarn=yes, then did a service asterisk restart
23:48.05*** join/#asterisk ariel_ (n=ariel_@c-71-196-99-26.hsd1.fl.comcast.net)
23:48.50*** join/#asterisk lanning (n=lanning@208.87.235.224)
23:53.38ariel_Evening folks
23:54.39ariel_I have a small issue with a polycom IP331 phones, They are working with asterisk just fine, but I need to find the setting in sip.cfg to turn on the little phone status icon, it's missing
23:54.59ManxPower-workariel_: the one next to the line appearance?
23:55.04ariel_yes
23:55.16*** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com)
23:55.32ariel_seems it does not like the standard build file I use for all the other IP450, 501's and 650's I have
23:55.45ManxPower-workariel_: The only time I've experienced that issue is when I did not update sip.cfg and phone1.cfg from the versions that came with whatever firmware first supported that model.
23:56.31ManxPower-worki.e. my sip.cfg and phone1.cfg were from before the 331 (I think ours was a 430, actually) was released so didn't have any options in it for that model.
23:57.30ariel_humm ok let me see if I can download the new firmware. I don't like updating unless I have too, things work best if left along when everything else is working
23:59.23ManxPower-workariel_: I understand.  I spent a day fighting it.
23:59.24ariel_getting both the split one and combo see which one will work best in the lab.

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.