IRC log for #asterisk on 20100105

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00:13.07rombany new help on http://pastebin.ca/1738079
00:13.09romb?
00:13.50jayteehttp://instantbueller.com/
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00:51.44SingularoAnybody know what to add to trixbox extensions_custom.conf dialplan to call an agi for each incoming/outgoing call?
00:52.23SingularoTried this:
00:52.23Singularo[outbound-allroutes-custom]
00:52.24Singularoexten => _X.,1,DeadAGI,test.php
00:52.24Singularo[from-pstn-custom]
00:52.24Singularoexten => _X.,1,DeadAGI,test.php
00:52.39[TK]D-Fender~trixbox
00:52.40infobotit has been said that trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
00:52.40SingularoBut it stops the call from proceeding.. sort of need to allow it to fallthrough or something?
00:52.55Singularothx
00:53.42SingularoBut its not really a trixbox specific question.. its a dialplan question ;-)
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00:57.54TJNIIAnd the trixbox dialplan is thousands of lines long.
00:59.42SingularoHmmm. ok. lets try a diff way. Is it possible to make a new ext that will simply call an agi, then return to the calling ext?
01:00.00carrarMillions!
01:00.28SingularoExcellent! you give me one example?
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02:01.53manxpowerSingularo: exten => 1234,1,Dial(Local/4567@yourcurrentcontext)
02:02.27manxpowerWhen exten => 4567 ends, control will be transfered back to the calling extension
02:02.29Kattysomeone gave me cooties :<
02:02.35manxpowerYou can also use macros and gosubs w/returns
02:03.57manxpowerSingularo: I don't usually recommend voip-info.org (the site has quite a bit of outdated info), but it can be a good place to find dialplan examples, just remeber it may be for a different version of linux.
02:04.51manxpowerdifferent version of Asterisk
02:05.17TJNIIKatty: I think they have a pill for that now.
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02:08.00TJNIIponders "Or perhaps a topical ointment...."
02:10.18Kattyno, i mean someone got me ILL
02:10.22Kattyfevery. achey.
02:10.29Kattyand all that impending woe of flu/cold
02:10.31TJNIIYea.
02:10.44TJNIIAdvil, Vicks, other OTC remedies.
02:10.51Kattyooh vicks.
02:10.52Kattyafks
02:10.55TJNIIWhat were YOU thiking of?
02:12.17Kattynot being ill?
02:12.24KattyI HAX THE VAPORUB
02:12.32Kattys/HAX/HAZ/
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02:14.01Kattyahhhhhhhhhhh
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02:14.08Kattythe person who invented this stuff deserves an award.
02:15.13notnorthHello. I'm having an issue with VoiceMail() and Dial(). if I Dial() mjultiple channels and no-one answers the subsequent VoiceMail() drops the caller into a state where they can LISTEN to VoiceMails not LEAVE a voice mail. Ideas?
02:15.44Kattyyou should be doing Voicemail(Extension) not VoiceMailMain()
02:15.54notnorththat's what I'm doing
02:16.14Kattythat's what you think you're doing.
02:16.15TJNIIShenannigans.  Pastebin your dialplan.
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02:16.27Kattythat's a lovely word, Shenannigans.
02:16.56KattyLunsford Richardson.
02:17.04Kattya pharmacist from Selma, North Carolina
02:17.18Kattyinvented vick's vaporub in the 1880s.
02:17.21notnorthMind if I paste the 4 lines in here
02:17.25notnorthIt's only 4
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02:18.08TJNIIPlease pastebin the whole thing.
02:18.17notnorthum no
02:18.25notnorthmy configuration is private
02:18.30Kattyit was called Vicks Vaporub Salve in the 1920s.
02:18.31notnorththere are only 4 relevant lines :)
02:19.06TJNIIWell then just pastebin the context that is broken.
02:19.13Kattythat's like taking your car to a mechanic and saying you only need the front half of the car
02:19.30TJNII"I want you to fix it but don't open the hood."
02:19.31notnorthKatty: no, it's not ;)
02:19.34Kattybecause the trunk has PRIVATE DATA
02:20.04notnorthhttp://pastebin.com/m69aa4a8a
02:21.19KattyIT BURNS
02:21.24Kattyjust a little.
02:23.15Singularothx manxpower
02:23.18TJNIIWell, I don't see anything wrong with what you pastebinned, but I didn't think I would.  Your problem is probably somewhere else.
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02:23.52notnorthI understand your point, but I can't paste my entire config :)
02:23.59notnorthany ideas as to where i should look?
02:24.29TJNIIMake sure it is running in the right context, and check your exten lines for typos.
02:24.30Kattyyou could find a conslultant and have them sign a confidentiality agreement
02:26.30TJNII* can jump in the dialplan if you typo a line.  However, if your dialplan is so broken that it can jump to an open voicemailmain you have much bigger problems.  Not to mention that if your system is secure then pastebinning the dialplan is a non-issue.
02:27.12psilikonIs there a way to do something like this 'exten => XXX,n,Dial(SIP/10&SIP/20&SIP/30, 60, m)' but have the phones ring in parallel instead of sequentially?
02:27.39TJNIIMaybe I'm wrong, but I thought that did call them "in parallel"
02:27.43jayteepsilikon, if you do it that way they all ring at the same time
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02:28.06notnorthit does call them in parallel
02:28.10psilikonMaybe it *is* my phones.
02:28.10notnorthat least that is the intent
02:28.40notnorthhm
02:29.55psilikonI see the CLI show each phone dial on a separate line with a slight pause in between. Maybe my pbx is slow.
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02:42.06ruben23hi i have existing asterisk server running, and added a TDM400P device how do i install it..? my lspci is this --->http://pastebin.com/m6d63b1a
02:42.38voipmonkpsilikon: yes you can
02:42.47Kattyhi mister monk
02:43.03voipmonkhello Chef Katty
02:43.22voipmonkruben - u need to configure dahdi
02:43.33Kattysomeone gave me cooties.
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02:43.45Kattyvicks and heat is helping tho :>
02:44.06ruben23voipmonk:i already install the zaptel and asterisk, but now i plugin the card..
02:44.55ruben23voipmonk:on the lspci its not detected as TDM400P, why si ti do i need to install the drivers on it
02:45.59voipmonkwhy si ti do wha?
02:46.45Talkradiomaybe a diff pci slot might help maybe the one you are using is shared
02:46.58Kattyhow's the wifey mister monk
02:47.05voipmonkready to pop
02:47.22Kattyhow long?
02:48.23ruben23Talkradio:it detected but its not identified as TDM400P
02:48.35voipmonkthats fine -
02:48.43ruben23Talkradio:http://pastebin.com/m6d63b1a
02:48.44voipmonkconfigure dahdi
02:48.56voipmonkand load the matching module
02:49.03ruben23voipmonk:im doing zaptel...
02:49.21voipmonkwhat version of asterisk?
02:51.05ruben23voipmonk:--->Asterisk 1.4.21.2
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02:54.58timholumwhat is the best sip trunk provider?
02:57.50Kattyinfobot: itsplist-us?
02:57.51infobothmm... itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
02:57.56Kattyinfobot: itsplist-uk?
02:57.57infobotsomebody said itsplist-uk was UK based ITSps include http://www.voiptalk.org/  http://www.voipon.co.uk/  http://www.gradwell.com/ and a few other tinpot companies you can dig up with google.
02:58.26timholumThanks Katty
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03:12.29teknomegayeah i installed asterisk on the same hardware i just had SipX on
03:12.33teknomegaand its sounds like ass
03:12.43teknomegait has odd artifacts in the voice every now and then
03:12.47teknomegajust random stuff
03:13.17teknomegayet when i install SipX its sounds perfect... i really think its the way asterisk handle media streams to an ITSP
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03:14.41teknomegai shouldn't say it sounds like ass... more it just doesn't sound perfect.. i hear a kerrrrr sound or shhuchhhh... its just weird
03:14.53teknomegalike some sounds get drawn out longer for split seconds sometimes
03:14.56teknomegalike a cell phone
03:18.40TJNIISame codec?
03:20.07jblackI now have the power to ruin your day.
03:20.18jblackYes. You.
03:20.20Kattybut you're not going to
03:20.23Kattybecause that wouldn't be nice.
03:20.36jblackawww, let me do this
03:21.22Kattyhttp://www.thinkgeek.com/homeoffice/gear/c628/
03:21.26jblackstar trek the mmorpg is now available for beta.
03:21.34Kattyi've already preordered
03:21.42Kattyi start playing the 12th
03:21.54jblackAre you serious?
03:22.00Kattyyes/
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03:22.10Kattyryan and his brother have also preordered
03:23.39jblackafter what happened to star wars, I'm surprised you'd risk the money.. and the memories.
03:23.57Kattythe game is more about the people you play with (=
03:24.06Kattyit will be enjoyable, even if the game is crap.
03:24.51jblackhttp://www.youtube.com/watch?v=l_IiP56g6aw&feature=related
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03:26.12rombhello all
03:26.24Kattyhi
03:26.42QbYanyone have any recommendations on a wholesale origination provider.. preferably selling Level 3..
03:27.06rombany people experienced with asterisk + festival ?
03:28.00jblackI hope you enoy going boldly where no man has gone before.
03:28.31Kattyi'm sure i will
03:28.40Kattymight even have to break out the ole communicator badge
03:28.54jblackI have the old walkie talkies.
03:29.04Kattynice
03:29.25TJNIIBAH!  Now I want to watch Star Trek...
03:29.37KattyToS are all on youtube.
03:30.17jblackstar wars wasn't too bad.
03:30.21Kattythinkgeek has a very interesting lasagna pan.
03:31.04jayteeyeah! Tiberius! cuz I wanna stink like Jim!
03:31.57jblackwhat happened to thinkgeek? They turned into a big pile of windows-only stuff.
03:32.18jblackwouldn't a pile of  these be cool? http://www.thinkgeek.com/computing/usb-gadgets/c609/?pfm=Carousel_Lilliput_5
03:32.29jayteeI was watching the TOS episode that had the giant planet eating cannoli in it
03:32.52teknomegaTJNII, yes same codec
03:32.59teknomegaTJNII, only on asterisk
03:33.09Kattythat's cute jblack
03:33.10jblackYeah, that's a good one
03:35.03TJNIIInteresting.
03:35.16jayteethis is still one of my favs http://www.thinkgeek.com/computing/usb-gadgets/9223/
03:37.37Kattyaww
03:37.40Kattythat's adorable
03:37.45Kattydoes it come with a sonic screwdriver?
03:37.52jayteenope
03:37.56Katty:<
03:38.46jayteeI just saw the new Doctor that replaces David Tennant. I'm not impressed
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03:39.21Kattyi really enjoyed David Tennant
03:39.24Kattywho's the new guy?
03:40.42Kattyhrmm.
03:40.44Kattymatt smith.
03:40.48jayteeI forget his name
03:40.59jayteegoofy lookin
03:41.04Kattyhttp://ghostradio.files.wordpress.com/2009/06/matt-smith-dr-who1.jpg
03:46.50filehugs on Katty
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04:02.49timholumI have an extension _8NXXNXXX that should be mached by 84060226 correct?
04:03.40timholumI am getting extension 84060226 rejected becouse extension not found
04:05.37fileN is 2-9, which would cause that pattern to not match that extension
04:05.47fileuse _8NXXXXXX
04:06.14timholumahh :) ok thanks, I knew that one, just did not see it :)
04:09.37timholumMuch better thanks file :)
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08:05.15ChannelZIt's oooohhh so quiet (shhh, shhh)
08:07.49jblackA little too quiet.
08:08.06ChannelZduhn duhn DUHHHHNNNNN
08:11.06TommyBotten:D
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08:16.20milouxif i set a variable inside an AGI script, with $AGI->set_variable will the variable survive and be able to be used in the dial plan after the script is terminated?
08:16.32milouxand the dialplan continues
08:17.56ChannelZyou must be using some PHP toolkit
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08:18.40ChannelZdepending on what this "set_variable" actually does, it probably does, if it's a channel variable
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08:28.57bidossessihi all
08:29.25ChannelZpeople hi
08:29.39bidossessii have a digium wildcard 410 with two x100m modules.will i be able to use asterisk to make calls on the pstn line?
08:30.45ChannelZyes
08:31.00bidossessii'm still confused as to the roles of fxo and fxs so any enlightenment welcome
08:32.00bidossessiChannelZ, was that "yes" addressed to me?
08:32.08ChannelZyes yes
08:33.10bidossessiso once i configure chan_dahdi, i should be able to use asterisk to receive and make calls on the pstn line, right?
08:33.20ChannelZYup
08:34.25bidossessiwell, that's nice to hear. i tried calling the pstn line while running dahdi_monitor but i saw nothing happen so...
08:34.38ChannelZThink of it this way, an FXO port connects to the line on your wall and lets you make/receive calls, as if * were a phone plugged into that line.  An FXS port will allow you to plug a phone into it, as if * was the line in the wall.
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08:37.17bidossessithks for reassuring me :)
08:37.18bidossessinow all i need to do is figure out my configs.
08:37.22phasipHey! I am trying t make asterisk connect trough an outboundproxy, the problem is that i have version 1.4.12, as I understand it, this version is buggy with outboundproxy?
08:39.00ChannelZbio: you configured /etc/dahdi/system.conf and have the right driver running
08:39.01ChannelZ?
08:39.14bidossessican asterisknow be upgraded to 1.6.x?
08:40.06bidossessiChannelZ, i installed asterisknow so i believe everything should be loaded pretty dandi. i ran genconf anyway, but the documentation is a bit obsucre on that part. too puch zaptel and not enough dahdi
08:41.03ChannelZhmm I don't diddle in asterisknow
08:41.49bidossessiChannelZ, oh? what your route then?
08:42.33ChannelZI just run plain asterisk, no gui
08:42.35bidossessii tried my original distro "opensuse", but i had /dev/dahdi issues.
08:43.45bidossessiChannelZ, i don't intend to run a gui either. i've got a dialplan ready fro an old config and i plan to integrate with gosa, so freepbx is useless to me, but i wanted to use a distro that would effortlessly recognize my digium hardware
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08:46.03ChannelZshrugs - sounds like driver issues, I run it on ubuntu swimmingly
08:46.43ChannelZI'm going to install *NOW in VirtualBox to see what it looks like
08:46.56bidossessioh ubuntu... which version of asterisk are you using if i may ask?
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08:47.17ChannelZI've run 1.4 and now 1.6.1
08:47.20bidossessiChannelZ, don't bother, it's just centos init3 style with freepbx included
08:48.41ChannelZso does dahdi_hardware show your card?
08:49.09bidossessiyes it does
08:49.23bidossessihold on, i'll pastebin the message
08:49.44ChannelZwhat message
08:50.13bidossessipci:0000:02:01.0     wctdm24xxp+  d161:8005 Wildcard TDM410P
08:50.18bidossessipastebin overkill
08:51.04bidossessithe pstn line is plugged into port 1 on the card. when i call that number from my cellphone i don't see any activity using dahdi_monitor 1
08:51.44ChannelZso then in /etc/dahdi/system.conf did you set it up for fxs signalling on channel 1-2?
08:52.30bidossessithat would be fxsks=1?
08:52.51ChannelZyah
08:52.56bidossessiyep done
08:53.37ChannelZso then you're using dahdi_monitor 1 and doing what, sending it to a file or..
08:53.45bidossessioh, i'm getting a signal now
08:53.53bidossessii'm using -v
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08:54.15bidossessii just got a signal, so i guess i must have been doing something wrong before
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08:56.22ChannelZgrrph
08:56.26bidossessiwb
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09:11.53creativxw t f
09:12.03ChannelZo m g
09:12.48TommyBottenb b q
09:13.01creativxwhy is the reregistration to my sip provider so slow
09:14.52creativxwith our iax provider it reregs in an instant
09:15.03creativxwhen i do sip reload now the phone# is offline for ~5 mins
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09:19.07ChannelZsip debug I guess and try to figure out whose doing what
09:19.28creativxyeah i guess
09:19.50creativxjust never looked at sip debugging before so not sure what fails where
09:20.41ChannelZwell you can 'show sip registrry' and get a vague idea of what it's doing just by looking at the 'state'
09:20.53ChannelZwhether it's sitting there waiting for a reply, or what
09:20.59creativxafter a sip reload and attempting to call the phone#, i can see the sip provider sending data via sip debug
09:21.14creativxbut im not sure what im looking for.. the call never reaches its context
09:21.42creativxstate says registered so
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09:22.04creativxthis is indeed weird
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09:36.06yangHi ! Is there another way to avoid syntax error if my username consist of / character, like is "provider/yang" ?
09:37.15Chodorenkoyang: /is not valid symbol for name
09:38.17Chodorenkoyang: in dialplan you use "/" as delimeter for different part of command
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09:40.01yangChodorenko: so an username with / will not work at all in the register string... ?
09:40.39creativxok now im puzzled. just doing an extensions reload yields the same problem with the sip provider.. whaaat..
09:41.17creativxcould it be the itsp is using a lot of dynamic IP's, but asterisk thinks its registered to only 1?
09:44.24creativxand it seems no matter what i do with the sip friend / registry line all inbound calls are dumped to the context defined in [global]
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09:48.34Omorikahey guys
09:48.43Omorikai'm having issues with one PRI span
09:48.46Omorikaq921.c:468 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
09:48.52Omorikacan u help?
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10:36.06pawzhi, I have a client who wants to implement a "push to talk" functionality on a website. Can asterisk function like this, as a direct connection without "dialing" etc ?
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10:38.34tzafrirpawz, generally, yes
10:39.07tzafrirpoint the phone to a context where the "default" extension does what you want
10:40.03pawzhmm, ok. i sorta know what that means
10:44.28pawzi understand contexts, and extensions. what do you mean "point the phone" to the context ?
10:44.45tzafrirpawz, that depends on the nature of the phone
10:46.05pawzwell i imagine we're going to have to write some sort of flash-based softphone to enable the functionality from the web. unless there's any other way to initiate a call
10:48.50pawzdo any softphones such as x-lite support PTT ?
10:50.45pawzahh yeah. their site says it does. i have no idea how to use it though
10:54.29m0epawz, I think I read somewhere about flash based softphones being available.. I dont think you guys need to write that.. i could be wrong though
10:54.59pawzyeah i've found a couple now after a quick search. and a nice flex api for creating your own
10:55.19pawzI'm more interested in what i have to do to asterisk to support PTT. I'd like to test it out with x-lite first
10:56.04m0ecool
11:15.05tzafrirpawz, if their phone starts a call with an "empty" number it will land somewhere in your dialplan. Just carry on from the specific point it lands
11:15.38tzafrirNo major issue with Asterisk
11:16.39pawzi think i understand. you just dial a blank number and then deal with it in the default context just like a regular call then ?
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11:31.51bidossessiif dahdi_cfg -vv shows FXS, can it be a mistake? my card has two x100m modules installed, which from my understanding, are fxo modules. how can dahdi_cfg report them as fxs?
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11:40.48tzafrirbidossessi, just create a config for it with 'dahdi_genconf'
11:41.24tzafrirbidossessi, in order to confuse you, FXO modules have FXS signalling, and ivce versa
11:41.56bidossessitzafrir, just great. dahdi_scan reports fxo as it should. :)
11:42.11bidossessinow to figure out why dahdi show channels shows nothing
11:45.50creativxgosh i dislike sip debugging
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11:51.13bidossessiweird. when i cp chan_dahdi.conf.template to chan_dahdi.conf, dahdi fails to start. must i remove the template?
11:51.48xtrac020Hi folks, i've put a X100P SE card into a fresh asterisknow 1.5 system, phones are registered and it looks like no channels are configured?
11:52.11xtrac020Channel map:
11:52.12xtrac020Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
11:52.12xtrac0201 channels to configure.
11:52.43xtrac020and in asterisk I get
11:52.45xtrac020locfpbx*CLI> dahdi show channels
11:52.45xtrac020<PROTECTED>
11:52.45xtrac020locfpbx*CLI>
11:53.14xtrac020and status is as follows
11:53.18xtrac020locfpbx*CLI> dahdi show status
11:53.18xtrac020Description                              Alarms     IRQ        bpviol     CRC4
11:53.18xtrac020Wildcard X100P Board 1                   OK         0          0          0
11:53.18xtrac020locfpbx*CLI>
11:54.04xtrac020I've read the novavox installation guide, but am now kinda lost
11:54.16xtrac020I can do extension to extension calling no problem
12:01.27bidossessiit looks like that's a common problem
12:01.48bidossessii have the same issue right now. where's the installation guide you're following?
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12:08.05xtrac020@bidossessi http://www.novavox.co.uk/docs/install-guides/novavox-x100p-se-installation-guide.pdf
12:08.38bidossessixtrac020, thks
12:08.58bidossessii'll check it out and see how different the setup is from mine
12:11.39xtrac020hmmm, not sure if this is relevant or correct, but in dahdi-channels.conf, there is the line "signalling=fxs_ks"  I would have thought it would be fxo_ks ?
12:11.42bidossessii have a wildcard TDM410P but the symptoms are similar
12:16.47creativxwell wippy ay. insecure was what i needed
12:19.49bidossessixtrac020, fxo_ls maybe?
12:24.49xtrac020FXO ports use FXS signaling. <- This I didn't know, now will look elsewhere
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12:28.11xtrac020my chan_dahdi_additional.conf file is empty, I think this may be a problem ?
12:33.24bidossessiproduced by freepbx?
12:36.54xtrac020yes, but maybe its supposed to be empty, i'm still looking at other stuff
12:37.18xtrac020i think I need to actually configure the channel on the FXO card, but am unsure as to how to do it
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12:42.03tzafrirxtrac020, chan_dahdi_additional is not related to an FXO card
12:42.22tzafrirwhat's the output of lsdahdi ?
12:42.59xtrac020### Span  1: WCFXO/0 "Wildcard X100P Board 1" (MASTER)
12:42.59xtrac020<PROTECTED>
12:43.37tzafrirxtrac020, ok. the channel is configured, but not in use by asterisk
12:44.20tzafrirIf you used dahdi_genconf, look at /etc/dahdi/dahdi-channels.conf
12:44.56xtrac020there is no dahdi-channels.conf in that directory
12:45.03xtrac020touch it ?
12:45.14tzafrirno
12:45.25xtrac020ko
12:45.53tzafrirgrep ^channel /etc/asterisk/chan_dahdi.conf
12:46.47xtrac020that file does not exist either, but there is the .template version of it there
12:49.13bidossessicreating that file kills dahdi when asterisk is restarted. "help dahdi => no such command"
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12:52.30tzafrirxtrac020, what version of dahdi do you use?
12:52.39tzafrirsorry: of asterisk?
12:54.05bidossessifwiw i installed asterisknow but upgraded asterisk to 1.6
12:54.31tzafrirbidossessi, are you on the same system with xtrac020 ?
12:54.35bidossessisince it seems we have a very similar problem, i might as well pitch in
12:54.53xtrac020@tzafrir asterisknow 1.5 with the latest updates on it
12:55.08tzafrirwhat version of asterisk?
12:55.11tzafrirtry: asterisk -V
12:55.16bidossessitzafrir, well, same system, but i upgraded asterisk to asterisk16
12:55.43xtrac020core on my system is 2.5.2.1
12:55.52tzafrirbidossessi, so for you the following minimal file should work:
12:56.10xtrac020sorry that freepbx is 2.5.2.1
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12:56.40littleballhello, anyone know echo cancel algorithm?
12:56.41tzafrirecho '[channels]' >/etc/asterisk/chan_dahdi.conf; echo 'channel => 1' >>/etc/asterisk/chan_dahdi.conf
12:56.48xtrac020Connected to Asterisk 1.4.28 currently running on locfpbx (pid = 2490)
12:56.53tzafrir(assuming you have just a single channel)
12:57.06xtrac020@ tzafrir yes, single chan
12:57.19tzafrirxtrac020, this was for bidossessi
12:57.44bidossessitzafrir, i have 2 channels but i only use one, so would it be a problem?
12:59.34coppicelittleball: yes, thanks
12:59.40tzafrirxtrac020, in your case you'll also need to set the signalling
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13:00.08tzafrirxtrac020, the simplest thing would be to run'  dahdi_genconf chandahdi
13:00.08xtrac020okay, its currently set to fxs_ks
13:00.30tzafrirand read /etc/asterisk/dahdi-channels.conf
13:00.47littleballcoppice, where can I get a detail algorithm for echo cancel
13:00.48littleball?
13:02.09coppicelittleball: if you want the basics you use google. if you want the details you need to work them out for yourself
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13:03.58Kealvoipmonk needs to seriously give up on attempting to assist me.
13:04.05xtrac020@tzafrir thanks, i've run that and i created the file, rebooting now to see how it goes
13:04.40beekKeal: You're beyond help?
13:04.40KealThe kind of stuff I do with voip lands most people in solitary confinement.
13:05.29KealStuff like SAC auto-dialers from hell.
13:05.37bidossessitzafrir, no love
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13:06.59bidossessitzafrir, chan_dahdi.so still fails to load
13:07.00xtrac020@tzafrir no joy here either :-( I get the all circuits are busy now message when dialingout
13:07.04KealI keep telling the people on the http://telephreak.org/ bridge they are numbnuts for wanting to waste a line-in on something like a scanner rather than using an internalized soundcard emulation using a modular ADC.
13:07.36bidossessiactually dahdi only loads if i nuke the chan_dahdi.conf and leanve the freepbx template
13:07.49KealI keep telling them that simply adding more soundcards, you will just keep wasting downtime updating the hardware.
13:08.19*** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net)
13:08.54KealPlus doing it that way is expensive to redonkulous levels.
13:10.10KealI am thinking of getting a closed doors license to continue producing Gatorade Tiger under a relabel.
13:11.48KealI'd call it something like Ligerade
13:12.23KealEither that or Ligerine.
13:12.25bidossessiconfirmed: i nuked chan_dahdi.conf and dahdi loaded in asterisk
13:12.52bochwhat is the best H323 channel for using with asterisk ?
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13:12.57*** mode/#asterisk [+o leifmadsen] by ChanServ
13:13.49KealOne of these days I am going to have to nuke 100% of the Asterisk source and create a reformulation that doesn't have all the line artifacts.
13:15.25KealStuff like detecting a tone in only 12 samples in-order to not have the clicking sounds when people push buttons, etc.
13:16.57creativxwho needs tones
13:20.01KealGoDaddy doesn't even offer dedicated servers with directly attached landlines.
13:21.37KealI told them I wanted to geographical remote from one another dedicated servers each with their own 800 landline directly connected and they shat their pants.
13:21.40Kealtwo*
13:21.55tzafrirbidossessi, if you don't have chan_dahdi.conf, chan_dahdi will not load (or ratheR: will fail to load)
13:22.16tzafrirnow, start by creating a proper one (or one you suspect might be proper)
13:22.25tzafrirand run:  module unload chan_dahdi.so
13:22.30tzafrirmodule load chan_dahdi.so
13:22.38*** part/#asterisk benngard (n=benngard@213.88.138.230)
13:22.47KealApparently this was the first time in history of the GoDaddy company that they could offer a hosting solution,
13:22.55Kealcouldn't*
13:23.02tzafrirWhat errors do you get (if any)? What do you have in your chan_dahdi.conf
13:23.03tzafrir?
13:23.46bidossessitzafrir, the exact opposite happens, right now, i have no chan_dahdi.conf and dahdi loads. the moment i create, even a minimal chan_dahdi.conf, dahdi fails to load in asterisk (no such command)
13:26.07*** join/#asterisk lftsy (n=mlr@install.deckpoint.ch)
13:26.44bidossessii just finished a yum upgrade so i'm rebooting the machine. i'll try again in a few seconds
13:27.43*** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net)
13:28.38*** join/#asterisk ickmund (n=magnus@ada-bcn-fw01.adamoeurope.com)
13:32.10*** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147)
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13:36.09coppicetzafrir: did OSLEC actually make it into the mainline Linux kernel?
13:36.20tzafrircoppice, sort of
13:36.28tzafrirIt's in the staging tree
13:36.34Skeeter-about speech recogniztion, is it only available in english??
13:36.37coppicestill?
13:36.59tzafrirI guess it will remain there until it will have a proper interface,
13:37.07tzafrirwhich can only happen if something uses it
13:37.40coppicetzafrir: so the dadhi RPMS/DEBS/etc with the major distros still don't use it?
13:38.00coppiceso, what is lacking in its interface?
13:38.21tzafrirDebian debs include it. but as part of dahdi :-(
13:38.50tzafrirright now its interface is based on the dahdi echo-canceller one, I guess
13:39.21coppiceso, what is needed to move it forward?
13:39.56tzafrirI guess that if some other driver started using it
13:40.06tzafrirmisdn looks like the best candidate
13:40.28coppiceif you rely on misdn you really are in sad territory :-)
13:41.05tzafrirOr the existing "telephony" driver (PhoneJACK - never looked at it)
13:41.21coppiceoh, that's dead and buried.
13:42.50*** join/#asterisk m0e (n=eX@41.196.147.115)
13:43.48voipmonki still have one
13:43.56voipmonksomewhere -
13:44.52dmastg'morning all
13:46.19voipmonkg'morning
13:46.42sun28moin
13:52.38Skeeter-is speech recogniztion worth it with asterisk ??
13:52.51voipmonkyes
13:52.54voipmonkvery much so
13:53.10Skeeter-which engine to do use?
13:53.14voipmonkLumenvox
13:54.12Skeeter-ok, i have checked the demo on the website, im not sure to understand how license works
13:54.33tzafrirSkeeter-, with speech recognition of today you don't actually get the exact text from audio
13:54.40voipmonklicenses are based on simultaneous use
13:54.48Skeeter-ok
13:55.18tzafrirrather, you give the speech regonition engine a number of potential words ("dictionary"), and it tells you which of those was used
13:55.20voipmonkso if u want 5 ppl to use your speech recognition, lumenvox says you need to buy licenses to enable 5 ports
13:55.44Skeeter-oh, that grows up fast
13:55.54tzafrirPart of the problem with writing one is how to translate words in the language to something the internals of the engine understand
13:55.55voipmonkwell.....  you can get creative
13:56.35Skeeter-is there anyway to remake all the language
13:59.49*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
14:00.00Skeeter-nvm, sounds like it is offered in 6 languages
14:09.58beekmorning jaytee
14:11.02*** join/#asterisk joako (n=ston3d@opensuse/member/joak0)
14:12.38Kattybroke a nail this morning :<
14:12.46beekCALL 911!
14:12.56beekMorning Katty
14:13.29*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-poaxapjoenrasnie)
14:13.35*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
14:13.35*** mode/#asterisk [+o angler] by ChanServ
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14:13.52Kattywell i wouldn't say it's 911 worthy, but it sure does /hurt/
14:14.13Skeeter-wouldnt a french manicure solve the problem
14:14.38Skeeter-im not talking about the pain here, hahaha
14:15.43Skeeter-im currently using spectralink phones wireless, is there any better/affotable wireless phones on the market
14:16.14Kealvoipmonk, why do you keep trying to help me with stuff that I never asked help for?
14:16.55jayteemorning beek
14:17.17beekHow are the animals faring in the cold?
14:17.31voipmonkwhat?
14:18.16*** join/#asterisk FabiOne (n=ovfuser@151.13.190.14)
14:18.21FabiOnehi all
14:18.59*** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163)
14:19.02FabiOnesomeone can where i can ask for help about asteriskgui 2.0?
14:19.07KealMy goal is to completely rewrite Asterisk and release that new item under GPL 2.0, not get help using it.
14:19.07FabiOne*tell
14:19.10*** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net)
14:19.15[TK]D-FenderFabiOne: #asterisk-gui
14:19.18FabiOnetnx
14:19.24[TK]D-FenderFabiOne: Just like the topic tells you
14:19.40FabiOnelol, sorry
14:23.58*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
14:27.04FabiOnethe only problem's that noone answer me...
14:28.02voipmonkyou cant force anyone to answer you
14:28.47Chainsawvoipmonk: Well with a kind word and a gun, you get more done then with a kind word alone.
14:29.12ManxPower-workFabiOne: If you cannot get support for the GUI you use, maybe you should rethink using that GUI.
14:30.00*** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net)
14:30.45FabiOneManxPower-work: sorry
14:31.00*** join/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net)
14:31.22LemensTSwhat ata do you guys suggest? just need 2 port devices, i was looking at SPA2102's
14:31.41ManxPower-workLemensTS: I guess it depends on what you need.
14:31.41voipmonkI use the PAP2T-NA
14:31.55ManxPower-workYou didn't bother to tell us, but I assume you need "two FXS ports"
14:32.22[TK]D-FenderManxPower-work: Safe assumption givent he name kinda implies it
14:33.08ManxPower-work[TK]D-Fender: he could be confused and looking for a 2-port FXO device.
14:33.21FabiOnei've used linksys spa2102, pap2t and grandstream ht-486
14:33.34FabiOneimho linksys are better
14:33.51LemensTSManx: Yea 2 fxs ports for analog phones. Its for people outside the network of the asterisk.
14:33.59FabiOneif you don't need router capabilities use pap2t
14:34.08ManxPower-workSIPURA/Linksys is what you want then
14:34.09KattySkeeter-: errr, french manicure?
14:34.14KattySkeeter-: do you mean French Tips?
14:34.17LemensTSYea I dont need router
14:34.26*** join/#asterisk porter (n=terdon@unaffiliated/porter/x-000001)
14:34.54KattySkeeter-: my nail split, and it broke so far that it bled a bit when i ripped the rest of the nail off.
14:35.14KattySkeeter-: my nail isn't long enough to do a french tip, unless they were going to put an acrylic nail on..
14:35.26*** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca)
14:35.36KattySkeeter-: and it's highly unlikely they'd put an acrylic nail over something which has the potential to get infected.
14:35.46FabiOneso the only advantage of 2102 is to bridge lan and wan port to make a passthru ethernet point
14:37.08ManxPower-workHaving the router can be very helpful.
14:37.27FabiOnei agree with ManxPower-work
14:37.56Kattymanx is usually very agreeable.
14:37.58ManxPower-worki.e. a hotel that only allows one device per port.  Use the NAT features of the router to get your laptop and your SPA online.
14:38.59*** part/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net)
14:39.29FabiOnei work in a wisp, i use 2102 as bridge to avoid problem with nat to client's pcs
14:40.51Skeeter-Katty, ok, french tips, i didnt know how it was called in english
14:41.15KattySkeeter-: (=
14:41.40KattySkeeter-: not a big fan of the french tip, personally
14:41.40*** join/#asterisk ruyo (n=psantos@195.23.253.223)
14:41.48*** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com)
14:42.09KattySkeeter-: i usually just keep a top coat and cuticle cream on my nails.
14:42.10gr0mithey FabiOne i am working on a wisp project here
14:42.10Skeeter-me neither
14:42.18Skeeter-Katty, u lost me there
14:42.50*** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com)
14:42.53Kattygets pictures
14:44.12FabiOnegr0mit: lol
14:44.36KattySkeeter-: http://www.backtoschooluk.com/wp-content/uploads/2009/09/seche_vite.jpg
14:44.42FabiOnegr0mit: go in private?
14:44.48gr0mitsure
14:44.48KattySkeeter-: clear nail polish.
14:45.17Skeeter-oh i see
14:45.33KattySkeeter-: there are two kinds. BASE goes on the bottom to help keep the nail polish attached to your nail. TOP goes OVER the nail polish to help dry it quickly and keep it there. also protects against chiping, fading, etc.
14:45.51KattySkeeter-: do you know what a cuticle is?
14:45.52ManxPower-work.part #asterisk-drag-queens
14:45.55ManxPower-workoops
14:46.04Skeeter-Katty, ya, same word in french
14:46.10Skeeter-cuticule
14:46.24Kattywell they make cuticle cream which keeps the cuticles soft and helps prevent hang nails.
14:46.31coppicecute tickle
14:46.51Kattyhttp://upload.wikimedia.org/wikipedia/commons/1/18/Hangnail_on_left_hand_pinkie_01.jpg <- hangnails are painful.
14:49.46*** join/#asterisk smooth_penguin (n=smoove@59.95.34.16)
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14:56.12*** join/#asterisk mocker (n=user@206.55.118.85)
14:56.13KattyManxPower-work: i'm about as far away from drag queen as they come.
14:56.43KattyManxPower-work: and there's nothing wrong with a guy taking care of their nails.
14:56.48ManxPower-workKatty: I don't believe that for a min.  Not based on your personality on the channel.
14:56.51*** join/#asterisk jesse0987651 (n=Administ@198.179.137.22)
14:57.31mockerI'm looking for SIP providers in the US for a business, anyone have experience with the SIP offerings from places like global crossing, xo, or att?
14:57.33KattyManxPower-work: no, i'd say i'm your typical woman.
14:58.24ManxPower-work~itsp
14:58.25infobot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
14:58.29Kattyin fact, now that i think about it...i can't think of a single friend of mine (female) that doesn't take care of their nails.
14:58.32Kattyeven my mom does.
14:59.00mocker~itsplist-us
14:59.01infobotrumour has it, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
14:59.02xtrac020okay folks, i'm back now after some more digging, and i'm getting the CHANUNAVAIL inside in asterisk, but dahdi show channels shows both pseudo and 1 being present....
14:59.09KattyManxPower-work: how is that drag queen?
14:59.16mockerManxPower-work: I know about most of those, but they seem like smaller players.
14:59.55ManxPower-workmocker: unless you are willing to commit to significant monthly mins and costs you'll go with a smaller carrier.
15:00.24mockerManxPower-work: My company is willing, just wanted to get some recommendations
15:00.40Kattymocker: we use bandwidth.com here.
15:00.48Kattymocker: probably about...10 channels.
15:01.00mockerManxPower-work: We were looking at McLeod, but they require you to drop a T1 in.
15:01.02ManxPower-workmocker: Well Verizon Wholesale are hell to work with, hell to get working, but they seem to be pretty stable.
15:01.06Kattymocker: had good luck so far (=
15:01.15mockerI'd rather have it all be SIP based, but still with a reputable company. :)
15:01.54Kattymocker: it's certainly nice when it works.
15:02.03mockerhaha
15:02.19ManxPower-workWhy do you want the less reliable SIP service over a PRI?
15:02.30ManxPower-work....rather then over a PRI
15:02.41Kattyprobably cost.
15:02.45mockerManxPower-work: My boss has a fear of being tied to a copper pair coming in.
15:02.45coppicebecause its the in thing
15:02.45Kattythat'd be my guess.
15:02.50mockerWe have two PRIs right now.
15:02.53Kattyyes. coppice
15:02.57Kattyit's FASHIONABLE to be sip.
15:03.01Kattyand wear ugg boots.
15:03.21coppiceyou mean its hip to be SIP. keep it slogany
15:03.43Kattyi associate HIP with a brand which stands for High Intensity Pigments.
15:04.00Kattywhich is just...weird, in the same sentance.
15:04.11Kattybut hey, it's cool.
15:04.19Kattyinfobot: sip
15:04.20infobotwell, sip is Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261)
15:04.31Kattyinfobot: forget sip
15:04.31infoboti forgot sip, Katty
15:04.37ManxPower-workmocker: Just so he understands that a copper pair will be more reliable
15:04.37coppiceinfobot: slurp
15:04.38infobotWatches as any program is installed and turns it into an RPM. URL: http://students.vassar.edu/~jajohnst/slurp/
15:05.07Kattyinfobot: Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP!
15:05.20Kattyinfobot: sip is Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP!
15:05.20infobotKatty: okay
15:05.21KealHigh Intensity Pigment Session Intensity Protocol, used by queer hairstylists everywhere!
15:05.21mockerManxPower-work: Well, our two locations are data centers.
15:05.38mockerSo hopefully our bandwidth will be ok. :P
15:05.42mockerknocks on wood, etc..
15:06.01KattyKeal: i don't think HIP carries hair color.
15:06.06Keal:|
15:06.25diatonicSo great to watch my school win the Fiesta Bowl last night.
15:06.26Kattyby the way, HIP is a line of L'Oreal
15:06.32ManxPower-workmocker: My (limited) info indicates Verizon wants you to commit to about 1 million mins /month.
15:06.38Kattyand L'Oreal does have hair color
15:06.41KealKatty I want some. :|
15:07.06KealKatty do they have magnetic pigment?
15:07.13KattyKeal: well if you really want some, i'd recommend the metalic duo.
15:07.14ManxPower-workThis number is NOT verified, just extrapolated from what we pay Verizon (we have a special deal with Verizon)
15:07.19KattyKeal: no.
15:07.20KealI want to be able to magnadoodle my hair.
15:07.25KattyLOL
15:07.25coppiceManPower-work: you can fulfil that with 2 teenage daughters
15:07.50ManxPower-workcoppice: I am fortunate that I won't ever have teenage daughters.
15:08.27coppicewhether that is fortunate largely depends on the reason
15:12.51smooth_penguinwhats a magnadoodle
15:12.56*** part/#asterisk Omorika (n=omorika@193.198.31.85)
15:13.12KattyKeal: may i quote you?
15:14.24Kobazhmm
15:14.34Kobaztrying to figure out how to set up multi-server registration on polycoms
15:15.11KattyKobaz: they have two servers available for every extension
15:15.37KattyKobaz: phone ip -> Lines -> server 2
15:15.58Kobazyeah
15:16.02Kobazthat's what i'm trying
15:16.07Kattyor could use line 2.
15:16.10Kobazit doesn't save
15:16.15Kattyit doesn't... save?
15:16.48Kattythe polycom doesn't save the settings?
15:16.49Kobaznope
15:16.57Kobaztrying a second time
15:17.01Kattydo you have cfg files in an ftp directory?
15:17.08Kattythey will overrule a polycom's config if changed manually.
15:17.19Kobazi edited the config on the ftp, and it didn't use the new config
15:17.25Kobazso i edited it on the website
15:17.28Kobazand that didn't save either
15:17.36*** join/#asterisk mrbnet_ (n=mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net)
15:17.43[TK]D-FenderKobaz: Web overrides configs
15:17.46Kobazi might just do a redo, just reset local/device and try to make it use the config
15:17.49mockerManxPower-work: I tried to make the point that for SLAs and reliability we should go w/ someone like PAETEC, didn't win. :)
15:17.50Kobazyeah it's supposed to
15:17.52[TK]D-FenderKobaz: Remind me to shoot youl ater
15:17.55Kobazheh
15:17.58*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
15:18.03Kobazi'm just trying to make it work for now
15:18.04mockerPAETEC drop in dedicated circuit, run SIP across that.
15:18.07Kobazand then i'll monkey the configs
15:18.24Kattyget rid of your mac.cfg file for now. move it somewhere
15:18.26Kobazi just edited server2 for line1 on the website
15:18.30ManxPower-workmocker: a dedicated circuit could be as reliable as PRI -- depends on many things
15:18.34Kattyand then reset the phone
15:18.34Kobazand saved, rebooted, and now it's gone
15:18.36Kattyredo your setup
15:18.48Kobazthe mac.cfg is just a pointer to the phone<exten>.cfg
15:18.58ManxPower-workBut only a moron would send his business phone traffic over the internet and expect it to be as reliable as standard telephone service.
15:18.59Kobazthat's gonna be the same
15:19.04mockerManxPower-work: Scared of the fact it's one circuit coming in instead of going across blended bandwidth.
15:19.32KattyKobaz: well you can do it the easy route and just delete the ftp info temporarily
15:19.39KattyKobaz: so the phone doesn't check ftp
15:19.46ManxPower-workmocker: Exactly!  So you don't have to compete with porn for bandwidth
15:19.47Kobazoh, yeah
15:20.03mockerManxPower-work: I know. :(
15:20.15Kattypokes Keal
15:20.16ManxPower-workmocker: don't worry.  Once you have the service for a while I'm sure your boss will either change their mind or fire you for letting them make bad decisions.
15:20.22mockerI'd rather have two circuits, one at each facility and do HA
15:20.23Kobazhaha
15:20.23mockerhaha
15:20.35*** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf)
15:20.37Kobazi vote for fired
15:20.59Kattyi doubt that.
15:21.07ManxPower-workI vote for boss admitting they are wrong, but that doesn't happen very often.
15:21.16Kattyif someone is hired to manage and setup that stuff...
15:21.17ManxPower-workmocker: whatever you do get your recommendations put in writing.
15:21.19Kattyand they make a recommendation
15:21.32Kattythen it would be retarded to not listen to the recommendation.
15:21.46Kattyin writing would be best.
15:22.08ManxPower-workKatty: how do you think the boss became the boss?  Chances are he was a better bullshitter than the rest of the canidates.
15:22.29Kattymy boss used to work on wallstreet
15:22.36Kattyhe just got the company dumped on him when his dad died.
15:22.43ManxPower-workI'm lucky in that my boss almost never does "stupid boss like things".
15:23.09ManxPower-work(I can't actually think of any "stupid bosslike things" he's done)
15:23.16Kattymy boss constantly tells me to tell him when he's gone and snickerdoodled something up
15:23.19*** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org)
15:23.47Kobazooooooh
15:23.51Kobazi already have a line 2 configured
15:23.59Kobazpolycom 320/330 only support two unique registrations
15:24.10Kattyyep
15:24.15Kobazaughh
15:24.21Kattythe 501/500 is 3
15:24.24Kobazyeah
15:24.33ManxPower-workThe 550 is 4
15:24.38Kobazheh
15:24.44Kobazand the 650 is 6!
15:24.58*** join/#asterisk sun28 (n=light@sun28.ipfw.su)
15:25.03Katty5 golden rings!
15:25.14Kobaztime to rearchitect
15:26.38Kattytime to check on the squirrels.
15:26.57*** join/#asterisk Raden (n=Raden@66-168-15-19.dhcp.stpt.wi.charter.com)
15:27.10Kattyoh
15:27.12Kattyit's raden
15:27.15Raden:(
15:27.23Radenhi Katty
15:27.23Kattyi haven't seen you in a bit.
15:27.25Kattyhugs Raden
15:27.31Radenbeen working on the road alot
15:27.33*** join/#asterisk trumee (n=trumee@117.241.144.141)
15:27.35Katty:<
15:27.38Radenhugs Katty
15:27.39Kattyin this weather?
15:27.46Radenyup
15:27.48Katty:<<<
15:27.50trumeeguys, i have setup snmp for asterisk. How can i query for the active calls in * using snmpwalk?
15:27.51Radenblizzards and all
15:27.56Kattyyuck.
15:29.01minotaur01trumee: i dont use snmpwalk but can it run console commands?
15:29.29Kattyecho -e "foo" | snmpwalk
15:29.49mockerI think all the big players I was sent to also require a dedicated circuit for their SIP offerings.
15:29.56mockerSo far Global Crossing and XO seem to need it.
15:30.06trumeethis is the output of snmpwalk http://pastebin.ca/1738878
15:30.09Katty'dedicated circuit'?
15:30.18trumeeminotaur01, no my * box is on a remote server.
15:30.49Kattywhat does that have to do with snmpwalk's ability to understand console commands.
15:31.08minotaur01lol katty is right
15:31.12Kattypretty sure it does the same thing here or in china
15:31.18trumeeKatty, what field in the snmpwalk gives the active sip calls?
15:31.20leifmadsenthat's how the internet works :)
15:31.28Kattyi don't know.
15:31.33minotaur01lmao
15:31.33Kattyi have no idea what snmpwalk is
15:31.39Kattybut you're not answering the question
15:31.40mockerKatty: They drop a line in
15:31.46Kattydoes it, or does it now, support console commands.
15:32.04Kattys/now/not/
15:32.35trumeeKatty, what do you mean? i can ssh into the box, and run asterisk commands
15:32.42Kattyfacepalms.
15:32.51minotaur01snmpwalk is an SNMP application that uses SNMP GETNEXT requests to query a network entity for a tree of information
15:32.52Kattyjust...nevermind.
15:32.58*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
15:33.03Kattyhi Naikrovek!
15:33.13Naikrovekheya
15:33.33Naikrovek55F in the office today
15:33.37Kattyewwwww
15:33.38trumeei have a mythtv on a local machine and a * box on a remote server. i want to query for active calls from the myth machine and stop the TV if there are any calls.
15:33.45Naikroveklove my cheapskate building manager
15:33.52Kattyi bet you do.
15:33.59minotaur01its a snmp client not a snmp server
15:34.31KattyNaikrovek: i hope you at least have mittens.
15:34.34trumeethe snmp server and * box are running on the same machine.
15:34.42NaikrovekKatty: mittenless.
15:34.47KattyNaikrovek: oh dear :<
15:34.50*** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca)
15:34.50Naikrovekyeah
15:34.52mockerHuh, I don't know if Asterisk has any SNMP support.
15:35.08Kattyas far as i know, just the phones do.
15:35.11Naikrovekmocker: res_smnp.so
15:35.14Naikroveki think...
15:35.16mockerNaikrovek: Ahh, nice.
15:35.18voipmonkits there
15:35.18ManxPower-workYou would want to use the Manager Interface, not some unsupported-nobody-knows-about-SMNP-hack.
15:35.18Kattyand the asterisk software relies on what the system time is
15:35.22voipmonku can add it and tweak it
15:35.23xtrac020right, trixbox is now downloading, hopefully will be easier to configure with an X100P card.
15:35.31NaikrovekManxPower-work has a point there
15:35.32trumeei followed this tutorial to setup snmp in *, http://www.voipphreak.ca/2008/10/28/asterisk-snmp-with-cacti-howto-upgraded-for-asterisk-16-and-ubuntu/
15:35.39ManxPower-workxtrac020: You would have to ask on #FreePBX
15:35.41tzafrirregarding a question from yesterday: if you want to have a '/' in the username part of a 'register =>' keyword in sip.conf,
15:35.52Kobazokay so
15:36.02Kattyso?
15:36.03ManxPower-work~trixbox
15:36.04infobot[trixbox] a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
15:36.11Kattysew! a needle pulling thread!
15:36.21tzafriryou must use an explicit transport. e.g.: register => udp://foo/bar:password@domain
15:36.24[TK]D-FenderKatty: NO SOUND OF MUSIC
15:36.28tzafrirNote the 'udp://' part
15:36.31Kobazdo i do reg1.server1 and reg1.server2.... or do i do reg1.server1 and reg2.server2
15:36.31*** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be)
15:36.33[TK]D-Fender</julieandrews>
15:36.37trumeeso nobody used snmp here?
15:36.48[TK]D-Fendertrumee: Nobody uses that module.
15:36.56Katty[TK]D-Fender: THE HILLS ARE ALIVE>
15:37.00Katty[TK]D-Fender: WITH THE SOUND OF MUSIC
15:37.07[TK]D-Fendertrumee: Welcome to the path of most resistance
15:37.16Katty[TK]D-Fender: DAaaa da da da
15:37.20*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
15:37.20*** mode/#asterisk [+o Deeewayne] by ChanServ
15:37.26Naikroveklol most resistance
15:37.26Kattyhi Deeewayne
15:37.27tzafrirxtrac020, configuring that card should be simple. But you should follow up a bit on questions
15:37.46DeeewayneKatty, hello :-)
15:37.46trumeethats a shame. there was somebody who runs mythtv and *, and introduced me to this idea of snmp. need to check mythtv irc logs
15:38.03*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
15:38.04xtrac020@tzafrir  I guessed it should be simple! but its eluding me atm
15:38.23tzafrirxtrac020, what's the output of:  lsdahdi
15:38.24mockerKatty: Who do you use again?
15:38.29mockerbandwidth.com?
15:38.34xtrac020might put the CD in and start from scratch again, i dont know how many things Ive broken now!
15:38.35Kattymocker: yes.
15:38.42mockerKatty: Experience has been good?
15:38.46Kattymocker: yes.
15:39.09tzafrirxtrac020, mind following up a bit?
15:39.11minotaur01trumee: http://www.voip-info.org/wiki/view/Asterisk+monitoring
15:39.19xtrac020### Span  1: WCFXO/0 "Wildcard X100P Board 1" (MASTER)
15:39.20xtrac020<PROTECTED>
15:39.45tzafrirgreat. asterisk uses it
15:39.56tzafrirnow, what do you see in:  dahdi show channels
15:40.01tzafrir(in the asterisk CLI)
15:40.05xtrac020yeah, and I sees the channels aswell, both pseudo and 1
15:40.23tzafrirgood. To what context is it set to go?
15:40.28xtrac020<PROTECTED>
15:40.29xtrac020<PROTECTED>
15:40.29xtrac020<PROTECTED>
15:40.45*** join/#asterisk dkirker-openmobl (n=dkirker@pcp065264pcs.wireless.calpoly.edu)
15:40.48tzafrirthe context is 'default'
15:40.51xtrac020ahh, should '1' be to a different context?
15:41.12tzafririncoming calls will go to the extension 's' in the context 'default'
15:41.23tzafrirsee also:   dialplan show s@default
15:41.27tzafrirdialplan show default
15:42.21xtrac020locfpbx*CLI> dialplan show s@default
15:42.22xtrac020[ Context 'default' created by 'pbx_config' ]
15:42.22xtrac020<PROTECTED>
15:42.22xtrac020<PROTECTED>
15:42.22xtrac020locfpbx*CLI>
15:42.23xtrac020-= 1 extension (2 priorities) in 1 context. =-
15:42.23tzafrirIf you have extensions.conf from the sample configuration, this should land you into a reasonable demo IVR
15:42.33tzafrirno, please don't paste it here
15:42.36[TK]D-Fenderxtrac020: PASTEBIN.  Stop flooding the channel
15:42.38[TK]D-Fender~pb
15:42.39infobot[~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
15:42.41[TK]D-Fender^^^^^^
15:42.49xtrac020apologies to all
15:43.19[TK]D-Fenderxtrac020: And your device seems fine.
15:43.31tzafririn this case it just plays you a file and hangs up. Which is not so useful
15:43.51*** join/#asterisk raden (n=Raden@66-168-15-19.dhcp.stpt.wi.charter.com)
15:43.55xtrac020i should be able to call out now ?
15:44.04[TK]D-Fenderxtrac020: that has NOTHING to do with calling out
15:44.21tzafrirxtrac020, call out from what phone?
15:44.22[TK]D-Fenderxtrac020: Did you set up an extension that dials out that device somewhere?
15:44.46*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
15:44.46*** mode/#asterisk [+o angler] by ChanServ
15:44.48*** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-tbgsnobiwxhkfbso)
15:44.49xtrac020i've got 2x spa942's both registered, i.e. they can call each other
15:44.57[TK]D-Fenderxtrac020: meaningless
15:45.04*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-elnxzeepojzvtpsb)
15:45.04ManxPower-workxtrac020: none of this had to do with outgoing calls.
15:45.21[TK]D-Fenderxtrac020: Did you set up an extension that dials out your device?
15:45.41tzafrirtry in the asterisk CLI:   originate SIP/peername  DAHDI/1/number_to_dial
15:45.42xtrac020I dont know
15:45.49voipmonk"no"
15:45.56[TK]D-Fenderxtrac020: #freepbx <---
15:45.57tzafrir(fill in 'peername' and 'number_to_dial')
15:46.02[TK]D-Fenderxtrac020: Not supported here.
15:46.13*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
15:46.13*** mode/#asterisk [+o malcolmd] by ChanServ
15:48.29*** join/#asterisk Buklov (n=buklov@213.138.71.254)
15:48.33Kattyhi malcolmd
15:48.57Kattyhas anyone seen dave?
15:48.58Kattyeppigy: ping
15:49.07malcolmdhowdy
15:49.12*** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel)
15:49.36eppigyKatty: pong
15:49.49Kattygood morning!
15:49.51Kattyhugs eppigy
15:51.01*** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl)
15:52.42*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
15:53.36*** part/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
15:54.15*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
15:54.33*** join/#asterisk high-freq (n=hfreq@99.188.120.31)
15:54.38*** join/#asterisk bidossessi (n=bidosses@41.221.23.177)
15:54.49EmleyMoorI have a problem with a Monitor I am trying to do not actually creating a file...
15:56.07*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:56.18EmleyMoorNo error shows in the console, but no files appear either. Other Monitors I do work, though - so I am wondering what stops it...
15:56.48*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
15:56.48*** mode/#asterisk [+o putnopvut] by ChanServ
15:58.06*** join/#asterisk ik_5 (n=ik@85.64.59.61.dynamic.barak-online.net)
15:58.29EmleyMoorHmmm... Monitor getting run twice... could that create such an odd result?
15:58.30eppigyhuggles
15:58.35eppigyhuggles Katty
15:59.21ik_5hello, i'm using asterisk 1.4.28. when I try to use bindaddr to a specific ip address asterisk does not bind the sip address at all unless it's 0.0.0.0 or 127.0.0.1, and it does not report any error. does anyone knows how can I debug the problem ?
15:59.56sbrathWhen I egress a call over DAHDI to a Merlin PBX, does anyone know if I have to do anything special to sieze the oubound trunk on the other side?
16:00.22sbrathIf anyone has a setup like that, I'd love a copy of a dialplan or a print of the config on the Merlin... I'd pay $$
16:01.43[TK]D-Fenderik_5: You can't bind to 2
16:01.47[TK]D-Fenderik_5: 1 or all.
16:02.06ik_5[TK]D-Fender, what do you mean ?
16:02.29ik_5[TK]D-Fender, I can't make it listen to a specific address ?!
16:02.31*** join/#asterisk lynxsys (n=thelynx@82-71-19-61.dsl.in-addr.zen.co.uk)
16:02.35*** join/#asterisk Tim_Toady (n=moi@77.49.184.114.dsl.dyn.forthnet.gr)
16:02.39[TK]D-Fenderik_5: One yes, multiple specific, no.
16:02.46[TK]D-Fenderik_5: Show us your attempt to do so
16:02.49[TK]D-Fender~pb
16:02.50infobot[~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
16:02.51[TK]D-Fender^^^^^^^^
16:03.18lynxsysHey all
16:03.24ik_5bindaddr=212.199.157.154
16:03.40ik_5[TK]D-Fender, thats all my rules regarding binding
16:03.48[TK]D-Fenderik_5: PASteBIn your configs, along with ifconfig and netstat dumps from your server
16:03.51lynxsysanyone know a good resource for cisco 7941 and asterisk
16:03.58[TK]D-Fenderlynxsys: ...
16:04.00[TK]D-Fender~wikis
16:04.01infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
16:04.44lynxsysthanks any others?
16:04.53EmleyMoorHmmm... yes, double Monitor doesn't
16:05.01[TK]D-Fenderlynxsys: www.google.com
16:09.18*** join/#asterisk moy (n=moy@74.12.129.52)
16:11.16*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
16:11.17Naikrovekcommpartners
16:11.32Naikrovekhuh
16:11.41Naikrovekthey're my sip provider
16:11.45Naikrovekand they run that wiki?
16:11.47[TK]D-FenderNaikrovek: pwned
16:11.50Naikrovektotally
16:12.09Naikrovekthey are good though
16:12.23Naikrovekno problems with them, and they have unlimited style trunks
16:12.34Naikrovekunlimited calls to US & Canada at least
16:12.36ik_5[TK]D-Fender, added debug to console gave me: WARNING[2371]: chan_sip.c:18770 reload_config: Failed to bind to 212.199.157.154:5060: Cannot assign requested address
16:12.39Naikrovek$33/mo/channel
16:12.41Naikroveki'm done :)
16:13.12[TK]D-Fenderik_5: Where's the pastebin of what i requested?
16:13.14*** join/#asterisk Akiraa (n=Akiraaaa@79.112.35.81)
16:14.06ik_5[TK]D-Fender, http://pastebin.com/d6d15bca8
16:14.52[TK]D-Fenderik_5: bindaddr=bindaddr=212.199.157.154 <-- whats with this nesting nonsense?
16:15.27ik_5[TK]D-Fender, it pasted me twice it's not like it in the conf files
16:15.55[TK]D-Fenderik_5: Also that address doesnt's eem to be that of your eth devices
16:16.11ik_5[TK]D-Fender, yes
16:16.28*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
16:16.41[TK]D-Fenderik_5: host=212.199.157.154 <-- why is this peer pointing to what you seem to claim is a LOCAL IP for the server?
16:16.51[TK]D-Fenderik_5: [012-out] <---
16:17.02[TK]D-Fenderik_5: this is some seriously screwed up stuff...
16:18.07ik_5[TK]D-Fender, it's not a local ip, it's an external ip of a sip provider
16:18.16[TK]D-Fenderik_5: You don't seemt o get it...
16:18.43[TK]D-Fenderik_5: You bind * to LISTEN to a specific INTERFACE on your server.  this is not a REMOTE address
16:18.55[TK]D-Fenderik_5: You seem to be lacking basic networking knowledge
16:19.09Skeeter-Anyone knows if Data plan only exist for iphones??
16:19.24[TK]D-Fenderik_5: If I have 5 phone numbers and you ask me for one to call me at, I shouldn't be telling you one of YOUR numbers.
16:19.35[TK]D-FenderSkeeter-: Go ask your telco
16:19.47[TK]D-FenderSkeeter-: this has nothing to do with the phone itself
16:20.01*** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net)
16:20.04Skeeter-iphone got their own data and voice plan??
16:20.19[TK]D-FenderSkeeter-: iPhone = a phone.  Data plan = carrier
16:21.06Skeeter-[TK]D-Fender, aight then, i rly though they were in a seperate categorie
16:21.51[TK]D-FenderSkeeter-: does my Nokia phone has a magical special plan.  Good friggen grief...  Phone = HARDWARE, not SERVICE
16:22.52Skeeter-iPhone are intented not to be unlocked which is why they have specified plans arnt they??
16:22.56ik_5[TK]D-Fender, I get it now thanks
16:23.18coppiceSkeeter-: all iphones are unlocked in some markets
16:23.27[TK]D-FenderSkeeter-: ... iPhone's get unlocked all the time and aren't on just one carrier.  Please find a clue, fast...
16:23.30ChainsawCorrect, Italy for example.
16:23.35ChainsawWhich is where I imported mine from.
16:24.24*** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
16:25.10Kattyhi farkus
16:32.47*** join/#asterisk jpcansa (n=jpbenavi@200.91.99.148)
16:33.18*** part/#asterisk mocker (n=user@206.55.118.85)
16:33.28*** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net)
16:33.50jpcansahi, how can i play a ROL when all g1 channels are busy?   http://pastebin.com/m34fede8d
16:33.55Skeeter-voipmonk, have succesfully register ur iphone with acrobits along with asterisk??
16:34.22voipmonkyes
16:34.28voipmonkyou've been reading my blog
16:34.42Skeeter-sure, im suscribed
16:34.43Kattyare there ultrasounds there too?
16:34.53voipmonkno :)
16:34.54voipmonknot yet
16:34.57Katty:<
16:35.06voipmonkits like $65 for the 3d one
16:35.15voipmonkI really want the 3d one :)
16:35.21Kattyyou should get it.
16:35.30Kattyyou will never have the opportunity, ever again.
16:35.32*** join/#asterisk Chodorenko (n=chodoren@mm-206-239-84-93.leased.line.mgts.by)
16:35.37voipmonkright
16:35.47*** join/#asterisk cadillackid (n=root@rrcs-70-60-34-73.central.biz.rr.com)
16:36.14Skeeter-voipmonk, would u mind pointing me how u accomplished that, i tried many softphone for iphone, all worked, but this one seems interesting
16:36.18voipmonkSkeeter-: I was using Siax for a while - because it does sip and iax..... but then when i found "Softphone" from acrobits I fell in love
16:36.38defsdoorhow does a softphone on iphone receive incoming calls ?
16:36.39voipmonku can record calls on the phone
16:36.46voipmonkwifi, defsdoor
16:36.54voipmonkand a did sip provider
16:36.57defsdooryeah yeah but it doesnt multi task?
16:37.03voipmonkyes it does
16:37.06voipmonkwith backgrounder
16:37.08Kattyinfobot: acrobits
16:37.09Skeeter-ya
16:37.12Skeeter-backgrounder
16:37.15Skeeter-lovely app
16:37.29defsdoordoesnt require jailbroken ?
16:37.34voipmonkyes it does
16:37.34cadillackidhey there im trying to set up a voicemail box with no password on a 1.4.21 system that is utilizing users.conf is this possible?
16:38.00Kattyaww. just iphone and itouch )=
16:38.01voipmonksoftphone can record calls on the phone
16:38.16voipmonkKatty: there's a mic attachment u an buy
16:38.22voipmonkif u have an itouch
16:38.35voipmonkand u can use bluetooth
16:38.35Kattyi carry a blackberry.
16:38.40Kattyso gonna be a no go there.
16:38.46defsdooruses nokia n95
16:38.58Kattyfender uses a banana phone.
16:39.20beekuses my shoe phone.
16:39.31voipmonkdoesnt the itouch have bluetooth?
16:39.39Skeeter-voipmonk, how did u set it  with asterisk??
16:39.48Skeeter-voipmonk, maybe the 3rd gen
16:39.50ManxPower-workhugs his $30 Walmart cell phone
16:39.54voipmonkits an extension that hangs off my asterisk system
16:40.01*** join/#asterisk cadillackid (n=root@rrcs-70-60-34-73.central.biz.rr.com)
16:40.05KattyManxPower-work: hey whatever works, right? (=
16:40.09voipmonki point my google voice number to my asterisk system
16:40.11p3nguinitouch?  Maybe you mean My Touch?  or iPod Touch?
16:40.13Skeeter-voipmonk, i have a 1st gen itouch and back in days we could make phones calls on 1.4 rofl
16:40.24voipmonkand when im on wifi i usually run the softphone to receive my calls
16:40.37*** join/#asterisk sebbl (n=Momofu@109.192.162.148)
16:40.43voipmonku can run asterisk on your itouch nowadays
16:40.58p3nguinitouch?  Maybe you mean My Touch?  or iPod Touch?
16:41.10NuggetBAD TOUCH!  BAD TOUCH!
16:41.16voipmonkyou can touch whatever you want on your own
16:41.22Kattytouches Nugget
16:41.25cadillackidasterisk runs on itouch?
16:41.32voipmonk"Good not mine"  ( quote from cooking mama)
16:41.34p3nguinWTF IS AN ITOUCH?
16:41.42Kattyit's a little pda like an iphone
16:41.42cadillackidIpod touch
16:41.44Kattybut with no phone
16:41.51cadillackidiphone without the phone
16:41.55Nuggethttp://www.apple.com/ipodtouch/
16:42.07p3nguinThat's an iPod Touch.
16:42.53cadillackidso anyway is there a way to create a VM box with no password?
16:42.59cadillackidin 1.4
16:43.18Nuggetnot that I know of, but it's certainly possible to make it behave that way as far as the user is concerned.
16:43.25Nuggetin the dialplan
16:43.28p3nguinNot really, but you can allow entry to the mail box without entry of a password.
16:44.00Nugget<PROTECTED>
16:44.06Nuggetcore show application VoiceMailMain
16:44.26Kattyso what happens if you leave the pin number out of voicemail.conf?
16:44.28Kattywhat does it actually do
16:44.38Kattyjust spaz on every pin number entered?
16:44.56cadillackidit wants the extension number when you leave the pin out
16:45.08Kattywoah.
16:45.09Kattyguys.
16:45.16Naikrovekbad juju
16:45.16Kattyfirst ever occurance, two squirrels SHARING the tray
16:45.20Kattyinfobot: crittercam
16:45.21infobotwell, crittercam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d
16:45.23[TK]D-Fendercadillackid: So tell it not to ask for one
16:45.28Kattythis is amazing.
16:45.46Naikrovekare they usually territorial or something
16:45.49Kattyvery
16:45.51KattyEXTREMELY
16:45.52*** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net)
16:46.03Kattythey get cranky if one even gets up on the porch
16:46.06Kattymuch less on the tray
16:46.10cadillackidthanks guys works perfectly with the s option
16:46.14Naikrovekmaybe they're relatives.  that's how it was when i used to feed squirrels
16:46.15p3nguinIf you enter the mail box based on caller ID and you don't ask for the password, anyone that can change their own caller ID could get into the mail box, right?
16:46.27KattyNaikrovek: yeah id on't know.
16:46.34KattyNaikrovek: but this is the first time i've seen it happen!
16:46.51Naikrovekneat huh
16:46.59cadillackidim using it for hotel mauilboxes so people wont change the caller ID
16:47.09Naikroveksoon mothers will be bringing their babies to the food and showing them how to eat it
16:47.10p3nguinSo what would be the more secure way to authenticate to the mail box without a password?
16:47.27Kobazp3nguin: using black magic
16:47.48p3nguinkobaz: I thought about that, but then I figured that was the silly way.
16:47.54KattyNaikrovek: i thought squirrels only mated in the spring.
16:48.16Skeeter-voipmonk, google voice, invitation only??
16:48.20Kobaz'no security token' and 'secure', are a contradiction in terms
16:48.32Kattygoogles.
16:48.45NaikrovekKatty: well "soon" is a loose term
16:48.49[TK]D-Fenderp3nguin: MANY cell company VM systems have been compromised like that.
16:49.00voipmonkyes p3nguin
16:49.13Kattygoogle says squirrel mating season starts in late winter.
16:49.20p3nguin[tk]d-fender: Oh, I know.  AT&T still allows you right into the VM as long as CID matches.
16:49.26KattyNaikrovek: January to March
16:49.28cadillackidOcTel was the worst for hacking via call-ID
16:49.52Kobazthere are still companies that let you turn off your voicemail passwor and auth with callerid
16:49.52p3nguinLuckily Verizon doesn't do that by default.
16:49.56KattyNaikrovek: gestation is about 33 days or so
16:50.09KattyNaikrovek: so i'd say you're right! they're going to start showing up in a few months!
16:50.10cadillackidverizon uses OcTel so they had to securei t down
16:50.22KattyNaikrovek: baby squirrels :> how exciting.
16:50.27coppiceoctel is a name I haven't heard for a while
16:50.36Kattyi knew an Octel once.
16:50.41Kattyhe was on slashnet.org
16:50.42p3nguinWhat does AT&T/Cingular use?
16:50.43cadillackidits an oldie but still lots of cell companies using it
16:50.59KattyNugget: do you remember Octel?
16:51.01*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
16:51.04NaikrovekKatty: the moms will bring their babies and defend them against anything that comes close
16:51.09NaikrovekKatty: it'll be awesome to watch
16:51.23KattyNaikrovek: there aren't many dangers in this neighboorhood.
16:51.25cadillackidI think AT&T uses it too.  id have to get my iphone out and check the voicemail..  I havent checked the voicemail in the 2 years ive had it
16:51.27KattyNaikrovek: cept cars.
16:51.37Naikrovekyeh
16:51.38KattyNaikrovek: they're generally smart enough to stay out of the road.
16:51.44Naikrovekyes
16:51.46Naikrovekmore or less
16:51.53Kattyand i always slow down for them
16:51.54*** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com)
16:52.29cadillackidI levelled a squirrel other day... it was either me and my Jeep in the ditch for a flat squirrel
16:52.35coppiceoctel was absorbed by avaya, was it not?
16:52.50Nuggetnope, never heard of octel
16:52.52cadillackidback when it was Lucent they bought OcTel
16:53.02KattyNugget: hrmm.
16:53.04*** join/#asterisk jo8330 (i=d04149c9@gateway/web/freenode/x-dsvcrjuqiqlenxqc)
16:53.14cadillackidNortel used to partner with octel for it on meridian / SL1 systems
16:53.18KattyNugget: maybe he was an afda regular. in fact i think he was. along with Bucko
16:53.25cadillackidthen Lucent bought octel and meridian mail was born
16:53.36jo8330hi there. i was just curious how people's experience with 1.6 has been for production systems.  i'm wondering if I should stay at 1.4 or move to 1.6
16:53.40cadillackidas nortel didnt want to push an AT&T product
16:53.59Kattyjo8330: i haven't upgraded any of my production systems yet. haven't really had the need to.
16:54.13cadillackidim on 1.4 for all my live systems
16:54.26cadillackidI did replace zaptel with dahdi on everything though
16:54.31jo8330ok, i think i'll stay at 1.4 then.
16:54.49Kattyjo8330: unless you have a good reason to upgrade, or your building a new box, there's not much reason (=
16:54.56jo8330it is a new box
16:55.01jo8330but it will go into production in a month or so
16:55.08cadillackidim building a few new boxes as we speak and am stayin 1.4
16:55.10Kattyi've heard many good things about 1.6
16:55.15jo8330but 1.4 does everything i need
16:55.16Naikrovekas have i
16:55.21jo8330which is really fastagi mostly
16:55.30coppicea lot of cellular systems use glenayre voicemail, which must have kept them afloat when the paging business died
16:55.33Naikrovekjo8330: well 1.4 is LTS until 1.8 comes out, so I'd stick with it probably
16:55.33Kattyi'm not familiar with the agi changes.
16:55.40Chainsawjust migrated kit from 1.6.1.12 to 1.6.2.0
16:55.52cadillackidisnt that what sprint / nextel has always used?
16:56.01jo8330thanks all, i'll stick to 1.4 for now.
16:56.45Kattyand if anyone was interested, i've almost gone through the 75lbs of squirrel feed.
16:56.56Kattycomes out roughly to 2lbs of squirrel feed a day
16:57.01cadillackidwhats your OS of choice? I put mine on centos or fedora
16:57.09Katty2.5lbs
16:57.16Kattycadillackid: i'm a debian fan.
16:57.49cadillackidits debian more efficient with the kit than redhat variants?
16:57.58Kattydunno.
16:58.00Naikroveki think it is
16:58.03Naikrovekcertainly smaller
16:58.05Naikrovekless stuff
16:58.10Naikrovekusually means faster
16:58.12Naikrovekusually
16:58.22*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
16:58.55cadillackidive been using AstLinux for some of my embedded boxes but havent tried it on a hiosted server yet
16:59.00*** join/#asterisk CunningPike (n=CunningP@204.239.8.157)
16:59.02jo8330i like debian too
16:59.16jo8330i used to use ubuntu but i hda some insane audio issues at one point
16:59.23jo8330which debian didn't have
16:59.25jo8330oddly
16:59.41cadillackidubuntu seems to be a resource hog
17:00.03cadillackidI like a lean mean machine..  no X, no GUI, no extra hardware
17:00.07jo8330is AsteriskNOW any good? what distro does it use?
17:00.28cadillackidI didnt like *now
17:00.46p3nguincadillackid: So why not use Ubuntu Server?
17:00.51cadillackidI found if you wanted the gui it was easy enough just to build ab ox and add the * gui
17:01.21cadillackidI may try that here in the lab
17:01.46p3nguinIf you like Ubuntu, the Server edition may be for you.
17:02.30cadillackidI was never aware of the server..  I think its worth a build up on an extra server..  will it run multi-core processors?
17:02.40p3nguinof course
17:02.43*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
17:03.25benngardlikes debian
17:03.53p3nguinPersonally, I can't stand Debian or anything that looks like Debian (which includes Ubuntu).
17:04.04jo8330anyone using 64-bit asterisk? good stuff?
17:04.06benngardwhy not?
17:04.28Chainsawjo8330: Running Asterisk 1.6.2.0 on non-multilib hardened AMD64 kit. It works well.
17:04.34p3nguin/usr/sbin/asterisk: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.18, stripped
17:05.05Chainsaw/usr/sbin/asterisk: ELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.9, stripped
17:05.49p3nguinjo8330: Coincidentally, the 64-bit one runs the same as the 32-bit one.  :)
17:05.59jo8330cool
17:06.01cadillackidisnt ubuntu based on debian?
17:06.06jo8330yes it is
17:06.27p3nguinWell, it used to be.
17:06.29benngardwith a lot of stuff u dont need
17:06.29cadillackidp3: what do you run on?
17:06.39jo8330i find ubuntu's a bit bloated
17:06.47jo8330it's more for desktop environments
17:06.50p3nguinIt has evolved so much that it's not really as Debian-esque as it was in the beginning.
17:06.58p3nguincadillackid: Arch
17:07.00jo8330freebsd used to be my fav, but debian kind of won me over
17:07.08benngardsame here
17:07.33benngardif u are used to fbsd it is easy to go to debian
17:07.58p3nguins/debian/any\ linux/
17:08.10cadillackidis arch meant for tuntime only or can you compile / develop on it?
17:08.13jo8330yeah. i found debian's cleaner and more organized than bsd
17:08.27benngardand u will very fast learn to type "aptitude install xyz" :)
17:08.34p3nguincadillackid: hmm... what?
17:08.35jo8330ports with bsd were a pain
17:08.41jo8330espcially if you had out of date ports after a few years
17:08.52p3nguinjo8330: It's not THAT hard to type "make install distclean"
17:09.07jo8330i've gotten into states that were a pain to fix
17:09.12jo8330possible, but still a pain
17:09.12cadillackidsorry meant to say is arch meant for runtime or are their compilers, etc for it?
17:09.31p3nguincadillackid: It's a Linux system, so you can do whatever you want with it.
17:09.47jpcansahi, how can i play a recording when all g1 channels are busy?   http://pastebin.com/m34fede8d
17:09.52cadillackiddidnt know..  Astlinux is a linux system too but its for runtime only
17:09.55*** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net)
17:10.18p3nguinThat really doesn't make any sense to me.  You can do ANYTHING you want with it.
17:10.36p3nguinThat's what OSS is all about.
17:10.44ManxPower-workjpcansa: The same way you handle all other calling errors, check the value of HANGUPCAUSE or DIALSTATUS and do stuff based on that.
17:10.45cadillackidit makes sense for astlinux because it is lean and mean and designed JUST to run asterisk
17:11.08p3nguinIf you don't install anything on Arch except for Asterisk, it'll be exactly the same way.
17:11.09cadillackidyou cross compile everything on a development machine
17:11.29cadillackidp3: thats what im lookin for
17:11.31*** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com)
17:11.38cadillackidso I dont have to cross compile all the time
17:11.53cadillackidthanks guys.. im outtie for now
17:11.59p3nguincadillackid: Arch provides a "core" install.  From there, you add whatever you want and take out anything you don't want.
17:12.01cadillackiddinner time
17:12.33jo8330asterisk with centos seems popular as well
17:12.35benngardfun, was just listening to weather forecast for sweden, - 20 celcius, and 10-20 cm snow :(
17:13.17Naikrovekjo8330: ubuntu server is pretty slim, but debian is very slim.  someone in here wrote an article on their site for compiling and setting up asterisk on a bare server using debian not long ago
17:13.21Tim_Toadyif you care about serious use of asterisk you go either with debian stable or centos, i personally prefer centos cause of the loger support time (7 years)
17:13.59Tim_Toadyrolling distros like arch is a joke when it comes to servers
17:14.17p3nguinOnly if you don't know how to admin a box.
17:14.43Tim_Toadythe box must work with the possible minimun administration
17:14.52Tim_Toadya rolling distro cannot provide this
17:15.05p3nguinI disagree.
17:15.14drumkilladistro wars are borrrrrrrrrring
17:15.39Naikroveki wouldn't say boring, sometimes educational, but played surely
17:15.40p3nguinNothing forces you to update every day.  If you're spending lots of time on it doing updates, you're probably bored and need to go find something else to work on.
17:15.43Naikrovekplayed out i mean
17:15.50drumkillafair enough.
17:16.42Tim_Toadyp3nguin secutiry forces you to have ur system up to date, and when the packages version is not frozen (rollin) nasty bugs and incompability can occur on every update
17:17.11NaikrovekTim_Toady: in that case you should install security updates only
17:17.19Naikroveknot regular run of the mill software updates
17:17.24p3nguinIf the box only runs one or two services on it, then you need to update those services.
17:17.25Naikrovekand certainly not kernel updates
17:17.43Tim_ToadyNaikrovek this is not always an option in rolling ditros
17:17.53Tim_Toadyand in the long tern it leads to problems
17:17.56Tim_Toadyterm*
17:18.04NaikrovekTim_Toady: never once heard the term rolling distro before.
17:18.27p3nguinThe term is "rolling release"
17:18.30Naikrovekbut you're saying to stick to a release, and don't upgrade because "here there be dragons" and i agree
17:18.36*** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net)
17:19.31p3nguinArch doesn't have a version like 10 or 5.2 or something like that.
17:20.08p3nguinAnd since it is source-based, you can install your own packages at any time you want with ease.  You don't have to update to all the new packages that came out today.
17:21.20Tim_Toadythats fine for a home box if u want to play and spend ur time creativly but not for production :)
17:22.05p3nguinQuite honestly, I'm kinda glad you don't want to use Arch.  It doesn't need anyone using who doesn't understand it nor desire to understand it.
17:22.57Tim_Toadymate im not arguing about arch, im just pointing that all distros are not suitable for any workload
17:23.14Tim_Toadyits not about arch
17:23.23*** join/#asterisk oej (n=olle@ns.webway.se)
17:24.06p3nguinI suppose you feel that way about any distro that hasn't been pushed as being a "server distro," too.
17:24.24p3nguinI'm sure you think that Slackware isn't server material, as well.
17:25.01Tim_Toadyp3nguin my home pbx runs slackware with 2.4 kernel :P
17:25.13[TK]D-FenderI ran a Slackware Samba server here for several years before switching it to centos
17:25.15p3nguinWhat about FreeBSD?  There is new software (including updates of old software) for it EVERY DAY.  I bet you think it's not good for a server, too.
17:25.15Tim_Toadybut as i said thats a home box for fun and play
17:25.37[TK]D-FenderI miss Slack 9.0 when it all fit on 1 CD.
17:25.42Tim_Toadyheh
17:26.00Tim_Toady[TK]D-Fender it still do if you dont want gui :P
17:26.24[TK]D-FenderTim_Toady: Sorry, I expect that too :)  I could accept if it could only fit ONE to start though :)
17:26.27p3nguinAll of these systems of well-suited for servers.
17:26.28Kobazso, anyone know how to do multi-server single-extension registration with polycom
17:26.50[TK]D-FenderKobaz: Huh?
17:26.54NaikrovekKobaz: multi server single extension?  WTF
17:27.04Naikrovekphones and registrations are a 1:1 ratio
17:27.10Kobazregister the same extension on multiple servers
17:27.16NaikrovekKobaz: you don't want to do that
17:27.19Kobazit's for redundancy
17:27.24p3nguinYou don't register extensions.
17:27.30Kobazyeah you do
17:27.35p3nguinno, you don't.
17:27.36Kobazsip peer registrations
17:27.38Kobazyes you do
17:27.40p3nguinno, you don't.
17:27.43Kobazuh, okay
17:27.54QwellYou register devices.  Extensions have nothing to do with devices.
17:27.59Naikrovekdevices are registered, and extensions point to devices
17:27.59Kobazwell whatever
17:28.04Naikrovekit's not a trivial point
17:28.11Kobazanyways
17:28.14p3nguinYou can register SIP peers, but certainly not extensions.
17:28.21Kobazyou know what i mean
17:28.29p3nguinno, YOU know what you meant.
17:28.32Kobazheh
17:28.32[TK]D-FenderNaikrovek: Don't argue with him.... he's already using the web interface to configure those phones... he's too far gone :p
17:28.34*** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com)
17:28.41Naikrovekoh wow
17:28.49Kobaz[TK]D-Fender: well because using the cfg files on the ftp wasn't working
17:29.04[TK]D-FenderKobaz: because you OVERRODE shit in the manual configs
17:29.10Kobazno
17:29.16Kobazthat was before playing with the web interface
17:29.25[TK]D-FenderKobaz: It works, you failed, and its all in the past now.
17:29.28Kobazheh
17:29.42Kobazi reset all the phone local configs so i'm back on the ftp cfg files
17:29.47*** join/#asterisk dkirker-openmobl (n=dkirker@pcp065264pcs.wireless.calpoly.edu)
17:29.54Naikrovekif you want a redundant distributed system, I have a PDF somewhere that you'd likely want to read.
17:29.57[TK]D-FenderKobaz: So moving on.... iit should not reg to a 2nd server until it fails refresh or perhaps if a call is placed to the primary and fails
17:30.00Naikrovekif i can find it though... another matter
17:30.09[TK]D-FenderKobaz: barring that, prepare to take up another line appearance
17:30.13Kobaz[TK]D-Fender: yeah, that's what i would want
17:30.22KobazNaikrovek: yeah that sounds good
17:30.31NaikrovekKobaz: http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf
17:30.35Kobaznifty
17:30.53Naikrovekit's a few years old so may not be fully applicable with modern versions but the concepts are there and still apply
17:32.23Kobazhmm, this isn't exactly what i was looking for, but it's interesting
17:33.46Naikrovekwell if you want any single phone to be available when and if one of a pair or more registration servers goes down this is one way to do it.
17:36.39Kobazit seems to assume that each phone is registereing to a different server
17:37.59Naikrovekin that example, 1001, and 1006 are registered to the same server
17:38.22Naikrovekthe example is a huge system distilled down to a minimum for educational purposes
17:39.52ManxPower-workhttp://www.fnords.org/~ewieling/polycom-config-examples/ for those that are interested
17:40.10Naikrovek404 for me, ManxPower-work
17:40.39Kattyalso 404 :<
17:41.09QwellManxPower-work: /~eric/
17:41.13QwellURI fail :p
17:41.29Kattyoah, there we go
17:42.29ManxPower-workSorry, http://www.fnords.org/~eric/polycom-config-examples/
17:42.34Kattyalright. time to get groceries.
17:42.42Kattypoofs
17:45.52*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
17:45.55*** join/#asterisk ruben23 (n=AGENT@122.55.48.243)
17:50.42jo8330Naikrovek: do you have a reference to the debian/asterisk article?
17:50.50jo8330i found installation was very straight forward
17:50.55Naikrovekhttp://thinkdebian.org/ I believe
17:51.05Naikrovekyeah it was cake
17:51.08Naikrovekis*
17:51.08jo8330cool thanks.
17:51.21jo8330anyone using python in their solutions?
17:51.34Naikroveki need beginner help with the dialplan setup though.
17:51.49Naikroveknto right now i mean
17:51.53Naikrovekto set up a system from scratch
17:54.15*** join/#asterisk corretico (n=laguilar@201.201.46.106)
17:59.38Godfather_i've a little echo calling to the pstn with my spa3102, how i could fix it?
17:59.42*** join/#asterisk Ad-Hoc (n=nimbus@62.1.231.93.dsl.dyn.forthnet.gr)
18:00.34ChannelZwelcome to the SPA3102
18:00.48Godfather_:-)
18:01.24voipmonktry  220+820||120nF
18:01.29voipmonkin the regional tab
18:01.36*** part/#asterisk Chodorenko (n=chodoren@mm-206-239-84-93.leased.line.mgts.by)
18:01.40voipmonkfxs input to  -1
18:01.47voipmonkoutput to -1
18:02.05voipmonkand if u have any pstn gateways try SPA To PSTN Gain: 3
18:02.14voipmonkand PSTN To SPA Gain: 3
18:02.22benngarddepends pretty much where in the world u are
18:02.53Godfather_voipmonk, in the regional tab in wich field?
18:03.14voipmonkimpedence
18:03.34benngarddeep down to the right on that tab
18:03.40Godfather_i see.
18:03.56jo8330spa3102 runs asterisk?
18:04.12benngardsure why shouldnt it?
18:04.24jo8330linksys :p
18:04.32Godfather_voipmonk, done, and fxs input output where?
18:04.46ruben23hi there guys, im getting like this on logs..any idea on this...? http://pastebin.com/mb43c8ce
18:04.57voipmonku have to be in admin and advanced
18:05.16Godfather_admin/advanced/reginoal?
18:05.25ChannelZI think you want the impedance setting on the PSTN tab, the one on Regional is just for the FXS port..?
18:05.41Godfather_i see
18:05.53Godfather_FXS Port Input Gain:FXS Port Output Gain:
18:05.56Godfather_those values?
18:05.59voipmonkyep
18:06.13benngardmake sure u chose "the" impedance value for your region
18:06.16voipmonkthose should be  -1
18:06.22voipmonkthen set the fxo gains
18:06.23voipmonkto 3
18:06.29voipmonksee how that works for you
18:06.51*** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca)
18:06.54ChannelZruben23: I think your log contains your answer on line 1?
18:07.19Godfather_benngard, i dont know the impedance value from my region, i test voipmonk settings
18:07.23Godfather_one sec
18:07.43benngardGodfather_: could work
18:08.12ruben23ChannelZ:yes i get declined form my voip provider is this an issue with the voip telco..? or on my server..?
18:09.06ChannelZwell THEY are the ones rejecting it so you'd have to ask them
18:09.52Godfather_LOL
18:09.59Godfather_no echo
18:09.59Godfather_xd
18:10.06Godfather_ty voipmonk
18:10.40voipmonk:)
18:10.42*** join/#asterisk Alagar (n=Administ@122.164.104.155)
18:10.54ChannelZGodfather_: Are you using an analog phone hooked up to the 3102 or SIP or something on the other side of * through it?
18:11.22*** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com)
18:11.28Godfather_i'm ussing a spa504 in other side
18:11.55*** join/#asterisk nny (n=scott@64.203.239.83)
18:11.58*** join/#asterisk Alagar (n=Administ@122.164.104.155)
18:12.44Naikrovekruben23: what is SIP/delridge - is that your ITSP?
18:12.51ChannelZLOL - look up "spa 504" on Google, first result
18:12.54nnyPre-planning a system with a PRI card and an additional card with 2 FXS ports. One will have a DID over the pri to a fax machine and the other is for a credit card machine. Do you think I want/need EC on the FXS card?
18:13.19nnyChannelZ: LOL
18:13.28nnyChannelZ: hawt
18:13.33seanbrightso is there no way to just block dialplan execution waiting for a digit, and then continuing when the digit is pressed?
18:13.37seanbrightexten => 1234,n,WaitForDigitAndThenJustGoToTheNextPriority()
18:13.37ChannelZWho knew disposable panties had model numbers
18:13.50*** join/#asterisk wierdo (n=jimmy@77.78.3.197)
18:13.51ruben23<PROTECTED>
18:13.51seanbrightit's probably something simple that i am not thinking of
18:13.56Godfather_hahaha
18:14.01Godfather_n1 ChannelZ
18:14.12Godfather_its a new model of the spa
18:14.17Godfather_:-)
18:14.18Naikrovekruben23: that means your voice provider is giving you the congestion signal.  you'll have to increase your channels or something
18:14.43Naikrovekruben23: or maybe THEY are congested
18:15.47ruben23Naikrovek: no relation to having high load average to my asterisk server..? im getting 10.030 on load average for asterisk calls
18:16.37Naikrovekruben23: a) get a new server or something, b) your provider is giving your system the error, so they're where the congestion signal is originating from, as i read it.
18:16.45ChannelZseanbright: Read()
18:17.07Naikrovekruben23: what else does your asterisk box do
18:17.17seanbrightChannelZ: thanks.  that'll work.
18:17.43*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:18.47ruben23Naikrovek:it do predictive dialing..
18:18.49ChannelZor there is WaitForNoise, you could let them scream into the phone to continue
18:18.51ManxPower-workGot SIP response 503 "Service Unavailable" back from 74.80.203.194
18:19.43ManxPower-workThat usually means something other than "you are using too many channels"  Contact your ITSP to find out why they are sending that.  Since we don't see the ACTUAL DIAL in that pastebin we can't know if you are sending a stupid number to your carrier
18:20.01ChannelZMaybe he forgot to pay his bill
18:20.06nnyif I have a PRI card going to an FXS port for faxing, do I want EC on the FXS card?
18:20.38ManxPower-worknny: It doesn't matter.  EC is disabled when the fax tone is detected
18:20.47nnyManxPower roger thanks
18:21.28Skeeter-how long to received the google voice invitation
18:21.39ManxPower-workruben23: you're not using stuff from Asteria, are you?
18:22.45ruben23ManxPower-work:im not familliar with it, im using a predictive dialing for my asterisk..
18:23.37ManxPower-workruben23: Asteria Solutions Group sells dialers
18:24.48ruben23ManxPower-work:ok ive search it, not its not im on vicidial.
18:25.34*** join/#asterisk vma (n=vma@pas75-4-88-176-16-205.fbx.proxad.net)
18:25.54vmaHello asterisk people !
18:26.24vmaIs it possible to have g729 passthru mode in iax2 like in sip ?
18:28.17[TK]D-Fendervma: Yes
18:29.51vma[TK]D-Fender: so I do not need to buy the licence if I convert iax to sip with the same codec ?
18:30.06[TK]D-Fendervma: Correct
18:30.13[TK]D-Fendervma: Should simply get repackaged
18:30.57vma[TK]D-Fender: thanks, you saved my day ;-)
18:31.05ManxPower-workvma: all the standard issues with G729 passthru still apply.  No MeetMe, not transcoding, no one-touch-recording, no ChanSpy, you need all your audio files in G729 format, etc.
18:31.16ManxPower-workno DTMF based transfers
18:31.34Naikrovekyeah pretty limited
18:31.39[TK]D-FenderManxPower-work: umm.... not quite right on that last one
18:31.50vmaManxPower-work: I need it only for a iax2sip gateway, so it is ok for me...
18:31.58Naikrovekdmtf is out of band usually so they can escape that limitation but the rest are valid, no
18:32.32ManxPower-workNaikrovek: Asterisk has historically considered DTMF to be part of the audio stream even when it's out of band
18:32.40Naikrovekrealy
18:33.01Naikrovekbut it's not encoded into the audio stream, is it
18:33.11ManxPower-workTurn on T/t on Dial and see if the call reinvites
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18:33.30ManxPower-workNaikrovek: I did NOT say it was logical, or even good.
18:33.35Naikrovekhehe
18:33.55Naikroveki would say "totally unexpected"
18:34.07mcargileI need to do host based authentication with 30+ servers all with the same carrier, is there a way in 1.4 to define multiple ip addresses in sip.conf for one account entry?
18:35.08mcargilesomething like
18:35.09mcargilehost=192.168.1.1,192.168.1.2,192.168.1.3,...,192.168.1.200
18:35.41ManxPower-workmcargile: host= defines DESTINATIONS for calls TO that peer.
18:35.54voipmonkyou may be interested in utilizing mysql for that :)
18:36.00ManxPower-workI think you mean permit=/deny= and you'll see examples of that in sip.conf.sample
18:36.17ManxPower-workpermit/deny control INCOMING calls to asterisk
18:36.37mcargilecool thanks
18:36.47ManxPower-workmcargile: you might want to READ sip.conf.sample
18:37.43p3nguinWith IAX2, is DTMF considered inband or out-of-band?
18:38.01ManxPower-workp3nguin: IAX2 does not support INBAND DTMF/
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18:44.48Naikroveknum simultaneous calls / 32 = how many megabits of bandwidth needed for g729 calls, right?
18:45.23Naikrovektalking SIP "trunk" here, no iax2 benefits
18:45.39Kattyhi.
18:46.17ManxPower-workThere are plenty of bandwidth calculators on the internet, one is even listed in voip-info.org
18:46.32Tim_Toadyi think for g729 and sip is about 30 calls per Mbit
18:46.35p3nguinDo most people use rfc2833 for DTMF on SIP?
18:46.38Naikrovekyeah i'll check my work there.  tyring to come up with a good rule of thumb
18:46.56NaikrovekTim_Toady: 32 by my math, and 32 calls * 32kbps each = 1024 kbps = 1mbit
18:47.02nnyany sangoma nuts here?
18:47.02Tim_Toadyyes might be
18:47.10Naikrovekso num calls/32 = how much bandwidth needed in kpbs
18:47.18Tim_Toadythe number 30 was off the top of my head
18:47.25Naikrovekexcellent guess
18:47.56moynny: depends on what Sangoma nuts means?
18:48.10nnywondering if the best solution for a PRI + 2 FXS is a A101D + a second card, A20100
18:48.10ManxPower-worknny: Nobody that uses Sangoma is nuts.  What Sangoma question did you have?
18:48.14nnyhahaha
18:48.21Kattynny: why don't you call them?
18:48.25Kattynny: they're very friendly.
18:48.26nnyi meant rather, a proponent
18:48.34Naikrovekenthusiasts
18:48.35ManxPower-worknny: do you need fax or modem on the FXS?
18:48.36nnyKatty: I did, must be a holiday there :D
18:48.50Kattynny: then why aren't you celebrating?
18:48.50nnyManxPower one line is a dial out for a CC machine, the other FAX
18:48.53Kattynny: off with you!
18:48.54nnyhehehe
18:49.00ManxPower-worknny: I would not even run those thru Asterisk
18:49.07moynny: is not ... I work there
18:49.22nnymoy: oh heh, well, left a message :D
18:49.22ManxPower-workthen you can just avoid 1) the analog card cost and 2) problems with Fax or CC thru Asterisk.
18:49.51moyI'm not from Support, but, I can help sometimes with easy questions :)
18:50.10ManxPower-workWhoo!  Whoo!  XO *actually* fixed something today!
18:50.18Kattygoes through sangoma's directory looking for moy's extension
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18:50.45ManxPower-worknny: If you run the fax or CC thru Asterisk the telco will blame Asterisk for any problem.
18:50.49nnyManxPower part of the sales proposition for our client is pushing those two lines over to a PRI (they have 11 analog + the fax/ cc right now). You're saying even if it's "route DID from PRI to FXS1 or FXS2, there exists potential for issues?
18:50.57moyx128 ... not at the office now though, will be back in about a week
18:51.07nnyahah!
18:51.15Kattyi will leave you eleventy billion voicemails.
18:51.15nnyit's an extended holiday!
18:51.19ManxPower-worknny: Fewer issues with Sangoma, but it's just another point of failure.
18:51.34ManxPower-workYou could also plug a phone into each of the lines for when you are totally out of service.
18:51.47moyjust working from home for a few days ... it's warmer here than in Canada :)
18:52.06Kattyi just bring home to work
18:52.20nnyManxPower I agree with the sentiment, is it known that trying to do what I mentioned will fail normally? Or do people do this and it works.
18:53.01nnynormally I keep CC/FAX and * separate, but if the client can spend 100 less a month on the two, it helps with the transition
18:53.41ManxPower-worknny: Back in "the old days" (2002-2004)  I had NOTHING but problems with PRI->Asterisk->FXS->Fax using Digium cards.  Maybe things have improved since then, but I'd not test the idea.  I just want stuff to work.
18:54.32nnyManxPower gotcha, i'll see if I can find any examples of sangoma + latest 1.4
18:55.06nnyhttp://www.voip-info.org/wiki/view/Asterisk+fax#Sangomacardsandfax is what I am looking at right now
18:55.16ManxPower-worknny: Sangoma seems to have better card timing than Digium.
18:56.04ManxPower-workBut trust me, when their credit card machine goes out you'll wish you had either a flame thrower to keep the angry villiagers with pitchforks away or wish you had installed an analog line for the CC machine.
18:56.27ManxPower-worknny: where is the client located?
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18:57.32nnyhere on hilton head, the person who is dealing with the client is an ex employee of the telco heh
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18:58.32nnyso it looks like with the proper sangoma cards, the FXS card can recieve timing from the PRI card
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19:02.04[TK]D-Fendernny: there is a sync cable to connect between the 2
19:02.08jo8330does anyone have suggested references for setting up a T1 PRI with asterisk?
19:02.18jo8330i've done lots of POTS systems, but this is my first time dealing with PRI
19:03.35nny[TK]D-Fender: yeah
19:04.39nnywould something of this nature solve the majority of issues with fax/cc through asterisk? There is no SIP to ATA or anything, just PRI -> ASTERISK -> FXS -> FAX/CC
19:05.18bmoracanny, it's been my experience (having tried what you are doing) that ATAs are both cheaper and work better.
19:05.34nnybmoraca: ATA? from a PRI?
19:05.38bmoracanny, it's a nice idea to be able to do that, but asterisk just doesn't do that well
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19:06.06bmoracanny, yep.  i attempted to implement the same thing you are and had some problems.  it worked about 90% of the time, but those other 10% just killed me with people complaining
19:06.23nnybmoraca: did you use the sangoma cards?
19:06.26bmoracanny, in fact, I could usually get a 36.6k dialup data connection out of it
19:06.31bmoracanny, yep, with EC
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19:06.52bmoracanny, it just doesn't work well.  on the other hand, though, i've never had a problem faxing on a local LAN with an ATA
19:06.55tzafrirThough the EC is disabled at the time of a call
19:06.59nnybmoraca: so an ATA worked 100%
19:07.01nny>
19:07.02nny?
19:07.04nnysorry heh
19:07.46bmoracanny, it does work...just not 100%.  and office workers are NOT tolerant of not working, even if it's just once per day.
19:08.44nnybmoraca: so you're saying neither that ATA or Sangoma solution worked 100%?
19:08.57bmoracanny, no, the ATA solution has worked 100%
19:09.14nnybmoraca: oh ok gotcha
19:09.17bmoracait's just limited to about 9600baud faxes.  no 14.4 or other higher speed.
19:09.44nnybmoraca: I assume most fax machines just auto negotiate the data rate, I have seen a dell that was set manually before
19:09.50bmoracabut the FXS ports on the PBX had problems.  even using the timing cable.  for whatever reason, it just wasn't reliable.
19:10.07bmoracanny, most do, yes.  if they fail at a higher speed, they'll try other ones.
19:10.34nnybmoraca: any ATA you suggest in particular? Did you have to set up the ATA and asterisk to play nice together?
19:10.43nnyer play nice in regards to faxing*
19:12.06bmoracanny, Linksys PAP2Ts work well.  just set the codec to alaw or ulaw (depending on your locality).  if you need many ports or might be doing voice calls, too, the Adtran TA900s are really nice, though not quite so easy to configure...
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19:13.05nnyI wonder if an ATA + CC machine is as reliable
19:14.05nnyI am starting to think ManxPower-work's solution is best ha, I usuallytry to avoid this because it's like evaporating water into steam, and carrying it in a baloon/condesnsing it into a cup when you are thirsty :)
19:14.53bmoracanny, CC machines typically operate at 2400baud...never had a problem with one of them, either
19:15.15nnyok this is enough info to delve further i think
19:15.30nnylet me find out and discuss it with the client
19:15.39bmoracanny, i usually recommend my customers keep one or two analog lines for alarm/fax/cc
19:15.52ManxPower-workSee!  I'm not crazy!@
19:15.57nnyyeah as do I 90% of the time heh
19:15.59bmoracanny, they tend to be more reliable and work without power.  especially for alarm lines
19:16.02ManxPower-workHmm..that sounded better in my head.
19:16.06voipmonknot at this moment, ManxPower-work
19:16.12nnyManxPower-work: hehe
19:16.17voipmonkits early yet
19:17.11bmoracanny, i did have one customer that wanted all faxes to be digital...so what we did was set up Hylafax to do fax-to-email and then a couple ATAs for outbound faxing
19:17.16bmoracait worked very well
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19:17.54nnybmoraca: yeah I use send2fax (a company here) for about the same thing at the office
19:18.01ManxPower-worknny: Another option would be similar to bmoraca.  Give everyone a DID, have asterisk detect fax tone on DID and send to RxFax.  Keep the analog fax line for when the PRI fax does not work
19:18.45nnyManxPower-work: yeah but 90% of the time, as everyone is saying is bothersome
19:18.50ManxPower-workI did that at a longtime customer.  Worked very well.  Not as reliable as analog, but users always had a direct telco line + fax for when they had problems with the fax->PRI-> e-mail system
19:19.27bmoracamy last experiences with RxFax didn't work very well, heh.  hylafax and iaxmodem have been fairly reliable, though.  plus, avantfax gives a nice UI for them to archive the faxes
19:20.00ManxPower-workbmoraca: I had problems with RxFax until about 2005.  After that things seemed to work very well
19:20.16ManxPower-workI just sent the fax to the user's e-mail address as a PDF.
19:20.36bmoracai haven't tried it recently.  i've heard good things about the 0.6 version of spandsp, though.
19:22.49ManxPower-workHere's an AGI that did the conversion and e-maoling
19:22.58ManxPower-workhttp://www.fnords.org/~eric/fax2email.txt
19:23.32bmoracaeww perl.
19:23.34bmoracadid you write that?
19:23.41Naikrovekeww?
19:24.16ManxPower-workbmoraca: Yes, but don't tell anyone!
19:24.21bmoracalol
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19:27.17nnybmoraca: how long ago did you try the sangoma t1 -> fxs btw?
19:27.29bmoracaabout 4 months ago
19:27.32bmoracamaybe 6
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19:40.18bmoracahrm...$529 for Google's new N1 phone
19:44.26etfonhomeynny, have you seen this part from Sangoma: http://www.sangoma.com/products/hardware_products/digital_analog_hybrids/b601.html
19:46.08nnyetfonhomey: yes, just need 1 more FXS :D
19:46.30Kattyomnomnomnomnoms popcorn
19:46.46etfonhomeyAh.  in what I read earlier, I saw you wanted 2 FXS.  Both for fax?
19:46.52etfonhomeyfax/cc right?
19:47.02nnyetfonhomey: the latter
19:47.03etfonhomeySorry, catching up to the discussion.
19:47.13nnyfax/cc
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19:49.53nnyetfonhomey: right now the client has an MICS with 11 lines and a bunch of phones, it's a mini call center for rentals.
19:50.05nnyetfonhomey: the pricing for a T1 is less than the 11 lines
19:50.19nnyetfonhomey: er 11 lines +fax/ccc
19:51.18nnyetfonhomey: so the ideal path would be using a T1, and since they don't need all 23 channels (and the telco doesn't do any kind of partial) to move the fax/cc to the t1. However if it's a matter of reliability I would rather not get myself into a position where they aren't happy
19:51.33etfonhomeynny, gotcha.
19:51.59nnyetfonhomey: it's been suggested that the syncing two cards thing is not reliable, and that an ATA is
19:52.36nnyetfonhomey: generally I have alsways avoided fax through *, so I was just pinging around seeing if anyone else had experiences that would work just as well if done right
19:53.14ManxPower-worknny: Just make sure your "T-1" is a PRI and not a "CAS T-1"
19:54.26nnyManxPower-work: should be a proper PRI , but I will confirm with the representative
19:58.44Skeeter-is there any official release date for asterisk 1.8??
19:58.58[TK]D-FenderSkeeter-: "when it's done"
19:59.46Skeeter-1st quarter,2nd ... ??
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20:01.09NaikrovekSkeeter-: is there something you're waiting on 1.8 for
20:02.18Kattybrr.
20:02.52beekis looking for the enhanced calendaring applications in 1.8
20:03.16tzafrirManxPower-work, does it really matter (PRI vs. CAS) if the issue is sync?
20:03.56Skeeter-Naikrovek, cant remember what was new it 1.8, but i remember it sounded interesting
20:04.05Naikrovekah
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20:06.21donnibhi
20:06.38donnibanyone using PyGoogleVoice together with Gizmo ?
20:06.43*** join/#asterisk oej (n=olle@ns.webway.se)
20:06.57donnibi am having trouble using it together. i don't get the ringback from GV.
20:09.12ManxPower-worktzafrir: no, but he'd really regret it if he accidently ordered a CAS T-1
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20:10.06ManxPower-worknny: ALSO check with other carriers, many will do partial PRI
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20:10.59ManxPower-workI used to work with a carrier that would bring a T-1 into the building, run it into an adtran channel bank, hand us off PRI with 8 - 22 B-channels, a D-channel, and hand us analog channels off the channel bank for fax.  Worked like a charm
20:12.04nnyManxPower-work: I will look into that as well
20:13.00ManxPower-worknny: where did you say the customer was located? (just the city/state)_
20:13.46AkiraaWhat's wrong when a remote SIP extension can receive sounds from the PBX, but it can't send them to other extensions behind the PBX?
20:14.01ManxPower-workAkiraa: usually NAT not configured correctly
20:14.14AkiraaManxPower-work: on which side?
20:14.42ManxPower-workAkiraa: Your symptoms don't help determine which side, but most people screw up the NAT setup on Asterisk
20:14.57ManxPower-workYou've been here long enough, you know what to look for
20:14.59ManxPower-work~answers
20:15.00infobotanswers is probably 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
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20:18.34mcargileahh Uniplay
20:18.34mcargilehttp://corp.vicidial.com/otrs/index.pl?Action=AgentTicketZoom&TicketID=4202&ArticleID=30916&QueueID=4
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20:22.09beekinfobot tell me about thebook
20:24.10etfonhomeynny, I have a client with the similar setup as ManxPower-work said.  T1 data to the building, with a couple analog lines split out.  This telco does it with a Cisco 2400 series IAD.  I believe they actually do SIP/RTP over a data T1 IAD with Cisco fax/modem relay between the IAD and CO.
20:24.38ManxPower-worketfonhomey: In my setup the carrier did a real T-1
20:24.55etfonhomeyManxPower-work, that's why I said similar, not exactly the same.
20:25.51ManxPower-workI don't trust "PRI over VoIP".  If you're going to use VoIP as the transport just hand me a SIP account.
20:25.52etfonhomeyThe telco advertises this product as a "dynamic T1".  Because you get to use the data reserved for the voice channels when no calls are taking place.
20:26.14bochim spying channels using ChanSpy() with the 'r' option for recording, but it is not recording anything, what could be the problem ?
20:26.16AkiraaManzPower-work: hm... would it work to set up a remote phone as a DMZ host?
20:26.52AkiraaI think the problem is remote because voice can be transmitted from the PBX to the phone, but not from the phone back to the PBX
20:27.06voipmonkthe advantage with what etfonhomey is talking about is the leased line and not having your traffic go over the public internet......  ( lies... even tho it does anyway on the far end most of the time )
20:27.29ManxPower-workAkiraa: you should never have any special NAT anything on the phone side.  In fact if you set the phone up for NAT it won't work with Asterisk's nat=yes you have set for that peer
20:28.10ManxPower-workAkiraa: Is Asterisk behind NAT?
20:28.32ecraneYeah.. if your telco provides the VoIP and the T1 and they do QOS for voip across their network this is more reliable then voice going over the public internet.
20:28.32AkiraaManxPower-work: yes, with ports forwarded
20:28.57ManxPower-workAkiraa: WHICH specific ports forwarded.  What is your externip= and localnet= settings in sip.conf?
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20:30.47AkiraaManxPower-work: 5060, 10000-20000 forwarded to the PBX box for SIP
20:31.00AkiraaI have a bunch of remote IAX2 phones as well, which work fine
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20:34.43p3nguinGotta love NAT crap.
20:39.01ManxPower-workAkiraa: and the answer to my other question?
20:39.16ManxPower-workp3nguin: NAT is simple.  It's people that make it complicated.
20:39.30Akiraasip.conf is auto-generated by FreePBX (the interface I use)
20:39.48p3nguinIt's often defective devices that attempt to do good NAT that make it complicated.
20:40.35ManxPower-workAkiraa: you'll never make it work then if you can't add the correct REQUIRED nat options to Asterisk's setup
20:41.06ManxPower-workAkiraa: Either configure it correctly or stop wasting our time.
20:41.29[TK]D-FenderAkiraa: Go follow the guide in their channel : #freepbx
20:41.46[TK]D-FenderAkiraa: do "?? ports" , "?? nat" when you arrive
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20:42.03ManxPower-workAkiraa: Unless you can get localnet= and externip= set correctly Asterisk will use the wrong IP address for the NAT'd client.
20:42.33ManxPower-work[TK]D-Fender: Good thing I don't have ops or I'd /kick him for wasting everyone's time.
20:45.06[TK]D-FenderManxPower-work: s/everyone/your/
20:45.16Naikroveklol
20:45.31ManxPower-work[TK]D-Fender: No, everyone that spent time reading his messages.
20:46.36[TK]D-FenderManxPower-work: Hundreds of idlers can't be wrong!
20:46.54ManxPower-workIf he was a total n00d, then I'd understand, but he's been here, he knows we don't support FreePBX or other GUIs
20:47.31[TK]D-FenderManxPower-work: And has serious trouble with even cut&paste instructions, but that's besides the point...
20:47.44ManxPower-workWelcome to Burger King, can I take your order?  Yes, I'd like a Big Mac.
20:48.09bmoracaAkiraa, look to the sip_nat_custom.conf file or something like that.  additionally, sip.conf is NOT autogenerated by freepbx and you can modify it if you need to.
20:50.20*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
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21:00.05SuPrSluGanyone using nagios to monitor?
21:00.33AkiraaManxPower-work: aha, progress
21:00.42Akiraasip_nat.conf, thank you
21:00.58SuPrSluGI'd like to do a test call to check whether asterisk can actually process a call. Doable?
21:02.06Akiraaexternhost, localnet, externrefresh -- to use for dynamic DNS
21:04.48leifmadsennice, someone joined via IPv6 :)
21:05.52bmoracai've been thinking about setting up an IPv6 tunnel with HE...but cisco routers are too loud to have in my bedroom and my fanless PIX 501 doesn't support IPv6, so i haven't bothered
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21:15.41ManxPower-workAkiraa: I am finished helping you.
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21:22.55beekSuPrSluG: I use nagios to monitor my Asterisk box.
21:24.49SuPrSluGbeek: can you get it to make a call. we can check if sip is running. but would like to have it do something like  a call file. to be sure peers and * are routing the calls. ever hear of anyone doing that?
21:25.55beekSuPrSluG: no, I haven't.   You could write a script for Nagios to call that would do that and return the results.
21:27.05donnibreally no one is using Pygooglevoice ?
21:27.12SuPrSluGbeek: tnx . just wondering if it was possible.
21:27.39Kattybrr.
21:28.00beekSuPrSluG: Definitely possible.  It depends on how much effort you want to put forth.  Wouldn't using qualify and then checking the peer status do the same thing for you?
21:30.29SuPrSluGbeek: possibly. sometimes qualify will  look good but the call may fail.
21:42.47*** part/#asterisk shafu (n=giany@83.169.0.238)
21:43.44p3nguinleifmadsen: What?  I've been on IPv6 (and #IPv6, if that's what you meant) almost every time I am here.
21:44.00leifmadsenp3nguin: and how awful of me to have not noticed you
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21:44.09p3nguinYou are forgiven.
21:44.15leifmadsenp3nguin: that, or I've never seen you join
21:45.41p3nguinAh well.  I just didn't want someone else to win the first IPv6 User prize undeservingly.
21:47.10p3nguinAnyone know if it's okay to charge a TomTom from your computer's USB instead of the cigarette lighter USB adapter?
21:47.44leifmadsenp3nguin: I'd think it should be fine
21:48.01nnyhmm this will be fun
21:48.02leifmadsencheck google :)
21:50.06p3nguinI was thinking of installing a cigarette lighter jack into the front of my computer just for charging it, but then I realized that is is probably a standard-wired USB cable.
21:50.08voipmonkp3nguin: its fine
21:51.59etfonhomeyp3nguin, does the TomTom also do data through the USB?
21:52.05carrarWhat good is a laptop without a cigarette lighter in it
21:52.30p3nguinetfonhomey: Yes.
21:53.32p3nguinI just wasn't sure if it would charge via computer's USB.  Some cell phones have funky wiring so that certain things do or do not work the same when connected to a data cable as compared to a charging cable.
21:53.46nnyuhhh
21:53.49nnythis is odd
21:54.14p3nguinI think my answer is going to be "I'll try it and see if it charges up, blows up, or starts a fire in my computer."
21:54.17nnyeven after a reboot my system has a channel open
21:54.23nnygonna pastebin one sec
21:54.30etfonhomeyp3nguin, Just remember.  Duck and cover.
21:55.06p3nguinI looked in the user guide, but it does not address charging whatsoever.
21:56.08p3nguinThe only thing it says is step 1 connect the charging cable to the device and plug it into the cigarette lighter -- doesn't suggest to disconnect after charging, doesn't say leave it plugged in forever, etc.
22:01.48*** join/#asterisk chocoz (n=chocoz@pool-173-52-163-137.nycmny.east.verizon.net)
22:07.27chocozhi, i'm trying to use sipp to test our asterisk box. but im unable to get a response when the sipp client and asterisk are on different machines
22:19.35Kattystares at clock
22:20.30Kattysighs
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22:24.41voipmonkyou starin at my clock?
22:27.08Kattyi don't know who owns it.
22:27.20Kattybut i wish it would hurry up and move along
22:28.10chocozyou can help me!
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22:31.41voipmonkheh
22:32.06voipmonkchocoz: 'what da prolem is?'
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22:34.08Kattyhttp://gudhak.org/img/howtomakecoupons.jpg
22:34.56Kattythat's awesome.
22:35.59chocozI'm unable to get asterisk to respond to sipp when they are on separate machines. Here is a chunk of the debug log: http://pastebin.com/m537ac50e
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22:37.53dlynesKatty: thief!
22:38.36dlynesI always knew you were kinda squirrelly ;(
22:41.15Kattyit was posted on reddit.
22:41.27Kattyand we all know reddit is squirrely.
22:46.23benngardi was reading asterisk-users, do they realy charge you for "renting" a number?
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22:53.01jblack"Reality show bug crash in california, causing major slut spill"
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23:34.50dzupsome one knows a software for call-center? that works out of clint browser and manages all operations of a call center with integrate support in pbx asterisk ?
23:36.31ChannelZFOP?
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23:39.06jaytee"I don't want FOP, I'm a Dapper Dan man!"
23:40.23dzupFOP? is that for me?
23:40.32dzupChannelZ: ?
23:40.37ChannelZYes
23:40.40ChannelZ~fop
23:40.41infobotAn XSL formatter written in Java that outputs PDF. URL: http://www.jtauber.com/fop/, or the Flash Operator Panel
23:41.06ChannelZok ignore _that_link
23:41.20ChannelZhttp://www.asternic.org/
23:41.41ChannelZand/or http://www.fop2.com
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23:47.19dzupChannelZ: nice
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23:50.23AkiraaThere is a setting in my SIP gateway (a rather old 4AFXS by Netgate): Under "SIP Information" there are 3 "run modes": peer-2-peer, proxy and gateway; now, I know peer-to-peer can be used to make direct phone-to-phone calls without an intervening PBX, what would be the difference between the other two?
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23:56.49coppiceAkiraa: its poorly documented, but proxy mode makes all the ports share a number, which gateway keeps them all as separate extension numbers. I'm not sure what else it changes

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