IRC log for #asterisk on 20101203

00:08.29*** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-gngwxjgpltstsnlr)
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00:21.00tessier_Is it normal for ITSPs to charge for inbound minutes? Teliax is charging for inbound minutes and it bugs me.
00:21.20tessier_It completely blows the economics of switching to them from POTS lines for us.
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00:32.26jdoetessier_: the ones I've dealt with, yes.
00:35.06jdoetessier_: a lot will also offer unlimited whatever for $X/mo, but with a cap on channels to prevent... well, prevent you from using them :P
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00:40.49tessier_jdoe: I really like teliax for the most part, although their website is slow and the billing/reporting is confusing and a pain to use.
00:41.12tessier_jdoe: But we are sometimes having call quality problems which I cannot duplicate and the boss is seeing cheaper deals out there and is itching to dump teliax.
00:41.24tessier_I'm pretty sure teliax is not the cause of the quality problems but I can't prove it.
00:41.40tessier_He's the type to waste 10 hours of my time troubleshooting problems to save $10 on a bill.
00:41.46tessier_And my time isn't cheap.
00:42.36jdoetessier_: no experience with teliax at all, so don't let me steer you at someone else or away from them.
00:43.06tessier_I have been using them for over 3 years now with great results.
00:43.15tessier_I'm so glad I got out of the VOIP consulting biz.
00:43.19jdoeall I'm saying is that charging for inbound is (in my experience) standard. I mean either you pay less for dids/channels and pay minutely, or you pay more for dids, get fewer channels, and get unlimited...
00:43.29tessier_In 2005 or so I was selling/installing custom asterisk boxes. It is a horrible business.
00:43.33jdoehaha.
00:45.05tessier_jdoe: This guy also used to be in the telecom business. A partner in a company which he sold for good money. So he thinks he knows all about telecom. And he says they get inbound for free and it makes no sense that they charge us for it. I have tried explaining about how our call is using a port no matter what (on their DS-3 or whatever) and that costs money but he doesn't buy it.
00:47.31jdoetessier_: best of luck :P
00:47.59jdoethankfully my employers generally only fight me on things when I'm actually wrong ;)
00:49.40DeHackEdI was asking ealier and got cut off. I have a pretty basic SIP phone line and would like to autodetect if the other end is a fax machine. I'm using AGI right now but am willing to entertain alternate strategies.
00:51.05tessier_jdoe: My employers fight me often it seems. Not that I'm wrong. I think it is just the kind of people I pick to work for.
00:51.32jdoeDeHackEd: no idea, sorry
00:51.40tessier_Actually, it's only been the last couple that are occasionally problematic. Prior to that everyone was pretty reasonable.
00:51.54jdoetessier_: call center?
00:52.48tessier_jdoe: No, we aren't a call center. Although I did set one up though. With asterisk even. They were very reasonable as an employer and I generally got what I asked for.
00:53.04vimpulsep3nguin:  _Corey_:  all:  I looked into VoIP some more and it looks like it isn't worth switching right now for this four-employee business.  The cost savings would be small and the effort is large.
00:53.04vimpulseI'm going to recommend that they keep their old Norstar 6x16 with DR5 instead, or that they upgrade to a newer Nortel PBX if they prefer.  But thank you very much for the info you provided:  you corrected my misguided notion that reliable VoIP required a T1.  :)
00:53.10tessier_The call queing was occasionally screwy but overall worked well. We used Teliax even.
00:53.32jdoetessier_: nah, I was just guessing based on what you'd want mass quantities of inbound for.
00:53.52jdoeI haven't had problems with asterisk for ours, but our 'call center' is *very* small.
00:55.13tessier_jdoe: Oh. We don't really have mass quantities. Maybe $50 worth.
00:55.27tessier_But a bill which we were hopeing would be $40/mo is now $100/mo or more.
00:57.33DocfxitI'm getting an error loading the sip.conf  the error is  parse error: No category context for line 1 of /etc/asterisk/sip.conf
01:00.37DocfxitThis is what is in sip.conf http://pastebin.com/DmeE9niv
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01:23.17*** join/#asterisk FuriousGeorge (~chatzilla@ool-4354d13c.dyn.optonline.net)
01:23.23FuriousGeorgehey all...
01:24.37FuriousGeorgei have a phone in a noisy location.  no one can hear it ring.  for analog phones there are strobe lights that can be installed in-line, but this is a sip phone...  i was wondering if there is anything that could interface with the rj11 port used for headsets
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01:39.14WIMPyFuriousGeorge: There must be many ways to do it, but connecting to the headset port is not one of them.
01:41.53FuriousGeorgeWIMPy: [TK]D-Fender suggested an ATA and a strobe
01:42.27WIMPyThat might be the easiest one.
01:43.58WIMPyOr if the phone has an optical ringing indicator, you could glue a photography slave flash to that.
01:44.20WIMPyOr if your Asterisk is near ... endless possibilities.
01:44.36*** join/#asterisk hydester (4a672fe6@gateway/web/freenode/ip.74.103.47.230)
01:45.23hydesterhi.  i'm trying  to use gtalk via asterisk 1.8.  i just get ringing on the phone, but it never really connects and i get this in the log: Dec  2 20:41:52] NOTICE[7647] chan_gtalk.c: Remote peer reported an error, trying to establish the call anyway
01:46.00hydesteri know my password is correct.  is there anything obvious i'm doing wrong from this message?
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02:07.45superm1hydester, known issue: https://issues.asterisk.org/view.php?id=18412
02:08.00superm1just started a day or two ago, no root cause yet
02:08.20hydestersuperm1: i actually just came across that on your blog ;)
02:08.43hydesterthanks for the tutorial!
02:09.03hydesteri'm using ubuntu, so i'm not sure how to go about the AGI route.  hmmm
02:10.05superm1it should still be pretty much the same on ubuntu
02:10.20superm1just agi support is built into the package
02:10.28superm1on openwrt it's split out to save space for people who dont need it
02:11.31hydesterah.  i'll give it a shot.
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02:20.55dmzhowdy y'all, if i wanted to plug a pstn line into an fxo (it's fxo right?) port and receive a fax using asterisk_fax would that work or do i need a fax modem?  what if i wanted to have a t1 interface of 24 lines of faxes coming in? would i be better avoiding * totally and just dealing with fax only software?
02:22.40WIMPyYou can do it with Asterisk.
02:23.02WIMPyBut if that's the only thing you want maybe not the most obvious choice.
02:24.41coppiceWIMPy what would you consider the obvious choice?
02:25.18WIMPyA dedicated fax server solution.
02:25.31WIMPyBut as I said: You can build one yourself with Asterisk.
02:25.35coppicethat would be far more expensive
02:25.48coppiceand offer no benefit
02:26.13WIMPyMaybe not for you :-)
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02:28.12coppicewell, it might give you V.34 FAXing
02:29.16hydestersuperm1: inbound seems to work on 101 udp, but i think not on 102 tcp.  but i suppose this is just because of how the config was defined in your example.  but outbound  still has the remote peer error.  your outbound example has replacing _1 with _+1, besides the AGI difference.  did you really mean replace both the _1 and _+ with AGI?  although i tried that too
02:29.44superm1hydester, if you are still having an error with outbound, you're not using AGI
02:29.44WIMPyBelonging to the rest of the world, I'd see G4 fax as the interesting difference.
02:29.50superm1run asterisk in -vvvr mode
02:29.57superm1and watch the logs as it goes
02:30.06superm1and did you make sure to reload your dialplan (dialplan reload)
02:30.17coppicenobody uses G4 FAX
02:30.21hydesteri just did service asterisk restart
02:30.32*** join/#asterisk micols (~mio@rlogin.dk)
02:30.53WIMPycoppice: Maybe not where you are.
02:31.07superm1if it doesn't stand out in the -vvvr logs, would you mind sharing your dialplan (feel free to scrub anything personal if you want)?
02:31.26coppicenot anywhere. even in .de nobody fas seen a G4 fax macine
02:32.05coppicethere were small numbers made 20 years ago, but they didn't catch on
02:32.47WIMPycoppice: Office copiers do G4, but usually ppl use software and that will do G4.
02:33.20coppicename an office copier that does G4
02:33.30hydestersuperm1: how do you run it in -vvvr mode?  i tried "sudo -u asterisk  /usr/sbin/asterisk -p -U asterisk -vvvr" and get "unable to connect" error
02:33.45WIMPyI'm not into copiers.
02:33.47superm1hydester, have the service running, when you run asterisk -rvvv it's a remote console
02:34.00superm1so still sudo start asterisk first
02:34.06superm1and then run sudo asterisk -rvvv
02:34.25WIMPyBut if you really want to know I guess I could find out. But then I guess so could you.
02:34.48coppiceI have researched this before
02:35.13WIMPyIt's all software anyway.
02:35.37coppiceno its not. G4 requires an ISDN interface
02:36.23WIMPySure
02:36.30WIMPySo what?
02:36.53coppiceseen a fax machine with an ISDN port?
02:37.44hydestersuperm1: https://gist.github.com/726500
02:38.08superm1so that looks correct
02:38.26superm1at least that it's using the AGI that is
02:38.30WIMPyI already said that I'm not into copiers, but actually I have ssen one anyway nad I'm sure it wasn't special.
02:38.41superm1are you sure that you have Google Talk enabled in google.com/voice currently?
02:38.53WIMPyIn fact it looked like a rather cheap one.
02:39.01hydestersuperm1: inbound works, not outbound
02:39.17superm1but that's not at all the same error you were getting originally before using AGI
02:39.19hydestersuperm1: i accessed it from the US and have used it for outbound via the web page successfully
02:40.59superm1hydester, let me show you my log from a successful call
02:41.29superm1http://paste.ubuntu.com/539252/
02:41.43superm1see the AGI script callback portion isn't happening for you
02:41.57superm1so you are either missing something on your dial plan
02:42.03superm1or your password was put wrong in that AGI script
02:42.07superm1or it's not marked executable
02:49.04hydestersuperm1: it is marked as executable.  how does it know to look in the agi-bin directory?  should it be owned by root?  i can post my dialplan (extensions.conf)
02:49.30superm1well that's the default place for it to be looking
02:49.58superm1you can do a little bit of hackery to run it by hand if you want too
02:50.16superm1just run /usr/lib/asterisk/agi-bin/google-voice-dialout.cgi and it will have a blank prompt
02:50.24superm1agi_channel:"xyz"
02:50.35superm1agi_dnid:"5555551212"
02:50.41superm1or whatever number you want to test dialing
02:50.43superm1and then hit enter a few times
02:50.54superm1and it will manually run so you can see if there is some sort of problem there
02:53.14*** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7)
02:55.24cmnky"VoIP Connections ... Over the next few days we will be filling in the remainder of these tutorials. Sorry for any inconvenience. Please check back soon."
02:55.37cmnkyany guess how long thats been on the digium website ?
02:56.07coppiceWIMPy: sagem makes a low cost DECT + FAX machine with BRI and IP ports. They don't even bother with G4 support in that. Its T.38 + G3. The BRI port is just to get added features for DECT
02:56.52hydestersuperm1: i guess youj meant .agi?  i did channel 102 and a number and got "answer" then "Getting login page".  not that i could hear anything since it was on the server.  i'll get my config
02:57.38superm1it should have rang one of your phones (whatever the default one is in your dial plan)
02:59.04WIMPycoppice: That hardly sounds like an office product. More like a home one.
02:59.07hydestersuperm1: https://gist.github.com/32eb6749056ec1699e6d
02:59.33coppiceWIMPy: well it one of the rare cases of a box with FAX and a BRI port
02:59.38WIMPyBut it's almost PC only anyway.
02:59.42superm1hydester, your google-in is a little messed up
02:59.44*** join/#asterisk Defraz (~Defraz@gump.fuzecore.com)
03:00.04superm1you want exten => myid@gmail.com, 1, GotoIf(${DB_EXISTS(gv_dialout/channel)}?bridged) at the start of it
03:00.18superm1and then you want to set exten => myid@gmail.com, 1, NoOp(Callerid  ${CALLERID(name)}) to exten => myid@gmail.com, n, NoOp(Callerid  ${CALLERID(name)})
03:00.20WIMPyBut actually fax is a thing where voip is a good thing. Fax usually doesn't work there any more so hopefully if finally dies.
03:00.31WIMPyit
03:00.40superm1like mine: http://www.arctangent.net/~superm1/gv_configs/extensions.conf
03:01.51coppiceWIMPy: not really. the importance of FAX, and problems with FoIP, just slows down VoIP migration
03:01.57*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
03:02.47hydestersuperm1: ah, so i supposed to add it to the top, not the bottom.  funny how in worked, and out didn't
03:02.55WIMPyOver here you will have a hard time to stay away from voip, even if you don't see it: keyword NGN
03:04.36coppicethe term NGN is mostly applied to things which are opaque to the user, so it really doesn't matter to them
03:06.26WIMPyIt does make a difference. Most features don't work any more. And some even managed to do it so badly that fax won't work.
03:07.20hydestersuperm1: what about my gv-outbound?  i was less sure about that
03:08.00coppiceNGN pretty much demands that everything works as before. its not hard for the network provider to make FAX and other modems run smoothly across NGN, because they have end to end control
03:09.50WIMPycoppice: In an ideal world maybe, but telcos are far away from ideal. You can't expect the same quality from a 4.99 IAD that you got from a several thousand CO line card.
03:10.17coppicea CO line card is about $10
03:10.19WIMPyAnd as it's all SIP based, many features are simply impossible.
03:10.33WIMPyPer port?
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03:12.07WIMPyToday you can get whole COs on ebay, but that's not the price the telcos paid for the stuff.
03:12.42coppicetelcos paid $10 for line cards
03:13.09superm1hydester, you need to fix your exten => _1NXXNXXXXXX,1,Dial(Gtalk/myid/${EXTEN}@voice.google.com) line still
03:13.12superm1it needs to be made AGI'ed
03:13.14WIMPyThe real problem is that they went sip instead of some of the ISDN over IP things that should be able to offer the same service as before.
03:14.07coppicethey started with ISDN over IP (H.323) and dropped it. that wasn't a very smart move, really
03:14.31WIMPyI couldn't agree more.
03:15.05WIMPyBut there are other options like L1oIP that should work at least as well.
03:15.21WIMPyBut I have never seen any of them in use.
03:18.57hydestersuperm1: does this mean it is my client?  https://gist.github.com/5be973f05066bf30c461
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03:20.46*** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net)
03:21.10neurosyshas anyone ever overloaded a poe switch with phones? what are the usual symptoms?
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03:22.42WIMPyneurosys: That surely depends on the quality and probably ranges from ports being shut down and SNMP traps being sent to smoke signals.
03:23.16superm1hydester, intersting
03:23.26neurosysWIMPy:  Do you think call quality may sporatically degrade?
03:23.45WIMPyThat sounds unlikely to me.
03:24.00WIMPyBut not impossible.
03:24.06superm1do you not have alwaw and ulaw both available?
03:24.23hydestersuperm1: i think 101 may be sent for outbound only?  your example had 102 for both directions?
03:24.45superm1oh wait were you dialing out from 102?
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03:25.13hydesterhydester: that was from 101.  if i do 102 then i get the gtalk_parser remote peer error
03:25.22hydesterheh.  whoops.  that was to you superm1 ;)
03:25.43superm1dont worry about getting 102 working until you have 101 for both ways
03:25.57*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
03:26.23superm1so in your sip.conf you might need to force the codecs that are allowed im thinking for 101
03:26.29superm1if you are getting errors like that
03:26.48hydestersuperm1: FYI - i'm using sipdroid on an android
03:27.00superm1hydester, yeah i saw those issues with sipdroid on android
03:27.07superm1i'm actually sure then you need to fix the codecs that are allowed
03:27.17superm1look at my 102 set of rules for how i force g722 as a priority
03:27.22superm1http://www.arctangent.net/~superm1/gv_configs/sip.conf
03:27.29superm1apply that same set of stuff to 101
03:27.48superm1and you might consider looking at csipsimple instead, i get much better performance out of it than sipdroid
03:28.38hydestersuperm1: now it rings, according to debug
03:29.22hydestersuperm1: but not for real, since i have the other phone next to me.  the chan_gtalk.c remote peer error
03:29.52superm1can you post your (updated) dialplan?
03:30.00superm1and i'm assuming you restarted asterisk or reloaded the dialplan right?
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03:35.52hydestersuperm1: ah, i figured out what i did wrong this time.  sheesh.  i never updated the agi script with my account info.  man, i feel like such a script kiddie
03:36.04hydestersuperm1: so out now works.  but it calls back after hanging up
03:36.17superm1how quickly are you hanging up?
03:36.24superm1does it happen if you wait for the call to connect still?
03:36.28hydestersuperm1: quick.  not much to say to myself
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03:36.46superm1try waiting for 20 seconds or so
03:36.59superm1because you're racing with the AGI script otherwise i think
03:37.55hydestersuperm1: yes, waited for connection.  but if it is a quirk with the script that is fixing  this in the first place, i can live with it for now
03:38.48superm1yeah this is all just temporary until the real solution is in place
03:38.52hydestersuperm1: but after all of this,  i did notice that the callerid on incoming calling calls now say asterisk instead of the caller's number, which i thought it original did
03:38.53joeyjonesis asterisk running on qnx stable?
03:38.54superm1(not using the AGI script anymore)
03:39.28drmessanoQNX is still around?
03:39.56superm1hydester, well that's weird if that came back
03:40.21joeyjonesdrmessano: qnx is massive, in its market
03:40.50superm1hydester, you might have messed up the callerid stuff in your dialplan maybe
03:40.52drmessanoSo are pork flavored candy bars
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03:41.50joeyjonesdrmessano: qnx is a real-time OS, and i think it would be quite useful for asterisk
03:43.00cmnkyjoeyjones, did you stop using linux ... shame on you joey
03:43.19joeyjonescmnky: qnx is a linux derivative, and no
03:43.21cmnkyjoeyjones, i leave you alone for a year ... and your all covered in QNX fluids now ... tisk tisk ;)
03:43.37cmnkyjoeyjones, its not a linux derivative ;)
03:43.43joeyjones*unix
03:43.49joeyjonescmnky: are you headmonkey?
03:44.02cmnkyjoeyjones, yes
03:44.07joeyjonesomfg
03:44.36joeyjoneswe thought you got shot in a mugging or went into massive debt and got sent to forced hard labour or something
03:44.42drmessanoOMG
03:44.52drmessano<-- chunkymonkey
03:44.56drmessanoI MISSED U GUYS
03:44.58cmnkyjoeyjones, its the latter not the former
03:45.34joeyjonescmnky: so, there was something going on then
03:45.38joeyjonesyou sorta disappeared
03:45.50cmnkyyeah ... ive been working ;)
03:46.23hydestersuperm1: thanks a lot for your help.  i really appreciate it.
03:46.33cmnkyi didn't know i was breaking barriers by working hard ... when do i get my magazine cover spot ?
03:46.54superm1hydester, sure no problem.  feel free to leave some comments on the particular areas you ran into issues on my blog, i'm sure  others will run into the same types of things
03:46.57joeyjonescmnky: you should come back to the chan
03:47.00joeyjonesit's changed a lot
03:47.09cmnkyimin ##linux
03:47.51joeyjonesnot good enough :p
03:48.12cmnkywhich channel you tlkin bout ?
03:48.47hydestersuperm1: np.  you are an ubuntu contributor?
03:48.50joeyjonescoders
03:50.16joeyjoneshow can i check my debian kernel timer?
03:50.40superm1hydester, yup, i'm an ubuntu core-dev
03:51.48drmessanoOh, so you're one of the guys I need to complain to about...
03:51.56hydestersuperm1: i'm using http://ppa.launchpad.net/dajhorn/pkg-voip/ubuntu for my asterisk install
03:52.40superm1oh cool, he backports from the pkg voip team in debian
03:53.27hydestersuperm1: apparently it works great! at least with your instructions and help too, but that is because i am an asterisk novice.  only just started playing with it last night
03:54.39superm1cool, make sure to read up on all the stuff that is in your dialplan now, it's easy to copy and paste stuff and have no idea what it's up to with asterisk configs
03:54.58hydestersuperm1: so would it be correct to say the SIP call to asterisk is not encrypted, and then the google voice part is?
03:55.16superm1i'm not sure if google voice encrypts the call tbh
03:56.32hydestersuperm1: heh.  i'll check it out.  this all started because i wanted to have a data plan on vacation where voice is $2/min, and wanted to have a way to be able to use my mobile as a phone too
03:56.56hydestersuperm1: but now i am much more interested in this for more than that
03:57.11superm1yeah it is way more powerful than you initially realize
03:58.43hydester<PROTECTED>
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04:11.32hydestersuperm1: commenting out those two google-in lines that i moved to top, that you suggested restored caller id and inbound calls still seem to work.
04:20.46greezmunkey<PROTECTED>
04:26.55ChannelZnot that I'm aware of
04:27.48greezmunkeyChannelZ: hmm, well there is one way to find out...brb
04:28.11FuriousGeorgegreezmunkey: your raid array has degraded, and as a result you are seeing an older copy of these files which is out of sync with the failed drive (?)
04:28.17FuriousGeorgecat /proc/mdstat
04:30.27greezmunkeyFuriousGeorge: Say it anit so...
04:30.32greezmunkeyaint
04:31.22greezmunkeyIt's restarting now, I don't have an array, but lvm is installed, so who knows.
04:34.26greezmunkeyhmm, the drives are fine, and the files didn't change. I've been staying up way too late, lately.
04:34.56FuriousGeorgeyou're probably just hallucinating then.  it happens
04:35.29greezmunkeylcrackx
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04:38.11ChannelZI recall you were monkeying with some packages last night with the docs, perhaps something got re-installed.
04:41.13greezmunkeyChannelZ: Thinking back, I think I made those changes prior to doing all of that... Sorry for the bother.
04:41.56ChannelZNo bother, just offering possible explanations
04:42.51trollasaurusMan, this system never goes idle... Even at this hour it's still doing 480MBytes/sec
04:42.57trollasaurus(read+write)
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04:55.16ChannelZThatsalottaporn
04:55.57carrarhell yeah
04:56.24greezmunkeythat's funny...
05:13.59trollasaurusThat system has about 7TB of storage from the SAN
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06:02.31oDeskhello all
06:03.25oDeski want to code a script that starts before the first ring into the dialplan
06:03.47oDeskand i guess the first ring shows this line "Starting simple switch on 'DAHDI/1-1'" into asterisk CLI
06:05.06oDeskbut this line actually come after i hear the first ring when dialing the into Asterisk, thus i'm losing the CallerID "i guess"
06:06.12oDeskwhat i want to code is.  a script that monitor the line using dahdi_monitor and detect the DTMF number using multimon script then return back the callerid to the dialplan
06:06.15oDeskanyone ?
06:09.42coppicewhy not let asterisk deal with the caller ID for you?
06:11.07oDeskcoppice: because i can't get the caller id through asterisk, but i can get the caller ID manually by recording the signal then decoding the DTMF wav file
06:11.35coppiceyou just haven't configured it properly to decode the caller ID
06:12.07oDeskcoppice: i would be glad if you can help me do it the correct way
06:12.29coppicebesides, multimon is a really bad choice for doing anything, though. the DTMF decoder in asterisk is far far more reliable
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06:15.28oDeskcoppice: ok, lets go step by step.. asterisk feel the call after i hear the first ring and the callerID comes before the first ring.. how possible it is to get it detected into asterisk
06:16.02coppicewhere are you? that sounds like the weird brazilian caller ID
06:16.18oDeskcoppice: actually SaudiArabia CallerID
06:17.07coppicestrange. I thought saudi caller ID came between rings, like most caller ID.
06:18.05oDeskcoppice: no actually the monitor says that it comes before the first ring, because i can capture it before the first ring and i've aslo asked telecom personal
06:19.06user1324anybody looking into gtalk/gv initiate error?
06:19.53coppicethat's a dumb design that brazil also uses, because it means a phone has to be mains powered. anyway the brazil caller ID mode should work... unless they do something else weird in formatting the DTMF string.
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06:57.48oDeskcoppice: not even brazil works
06:58.14oDeskcoppice: do you have any article that greatly help is such cases  ?
07:17.49greezmunkey<PROTECTED>
07:18.10greezmunkeyoDesk: THere are others.
07:21.15ChannelZso cidsignalling=dtmf / cidstart=polarity is not the magic combo?
07:22.05greezmunkeyChannelZ: if he has a Zap...I didn't see any mention of that.
07:23.41ChannelZhe's using DAHDI but Zap has the same settings IIRC.  Whether or not they're right or not, dunno..  I would assume there is a polarity shift or some other signal before the DTMF callerid
07:25.30greezmunkeyChannelZ: yeah, I found a site that partially discusses how cid works in saudi, but the link to the detailed description is broken!
07:26.07ChannelZbørken
07:26.16greezmunkey.
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07:38.13dan__tIs there a function that I can use to just blurt some information in the log inside of an extension?
07:38.16dan__tFor debugging etc etc.
07:41.50ChannelZVerbose()
07:41.56dan__tThank you.
07:41.58greezmunkeyexten => _X.,n,Verbose(2, ${variable})
07:42.06greezmunkeyas an example
07:42.23ChannelZyou must be logging verbose though
07:42.24dan__tPerfect.  Didn't see it on the voip-info.org site under Asterisk Functions.
07:42.34dan__tIs there a better documentation resource?
07:42.37ChannelZIt's an app, not a function
07:42.41dan__tOh, right.
07:42.43dan__tSorry.
07:42.49dan__tHence ()
07:43.00ChannelZ(if you were looking at the 'function' section that'd be why you didn't see it :)
07:43.03greezmunkeyApendix E of _Asterisk: the future of telephony_
07:43.15ChannelZwell functions have () too actually
07:43.18kaldemardan__t: asterisk itself is the best. you'll always see what your version has. "core show functions" in CLI. "core show applications" shows all apps.
07:43.27dan__tPerfect, thank you much.
07:43.42dan__tI'll download it again, thanks, greezmunkey.
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07:48.59greezmunkeyOutside of the obvious problems, like overwriting dialplan, etc; it seems that * happily accepts multiple connections to the cli. I haven't found any discussion regarding this. Is this by design, and are there any deep dark downsides to having more than one cli connection at a time?
07:50.05ChannelZIt's by design
07:50.56ChannelZIt's not really a problem
07:51.45greezmunkeyChannelZ: I can see that being happy from a remote support perspective, like being able to interact with a local user. Nice.
07:52.38greezmunkeyHappy, er, handy
07:52.52ChannelZyou can fart around with cli_permissions.conf in 1.8 if needed
07:53.37ChannelZor I guess 1.6 has it too.. 1.4 probably but I dunno
07:54.18greezmunkeyChannelZ: I loaded 1.6 last night. I'm doing my home work on upgrading, but it really doesn;t look like it's too difficult. At least not with simple systems.
07:54.25ChannelZhmm maybe not looking at an old source dir
07:54.33dan__tkaldemar, any way I can query * for all variables that are available?
07:55.08greezmunkeythat conf file is here.
07:55.17dan__tI want to know which DID a call originated on.
07:55.35joeyjonesChannelZ: i got an asterisk server running, any suggestions on tutorial/reading besides the book?
07:55.35dan__tI thought DIALEDPEERNUMBER was it
07:57.03ChannelZlook in doc/text/channelvariables.tex
07:57.14ChannelZI thought it was ${DNID} or something
07:57.42dan__tLooks like its ${EXTEN} i think..
07:57.53ChannelZwell ${EXTEN} is the dialed exten
07:58.14dan__tRight, sorry - that DID essentially is an extension I guess.
07:58.16ChannelZif that happens to be your DID like from a SIP channel from an ITSP...
07:58.21dan__tIt is.
07:58.39dan__tProbably not technically the one variable that I'm looking for but for now I think that will work.
07:58.42ChannelZok.  It's actually the CALLERID(dnid) function but I think that mainly has relevance on PRIs
07:58.51dan__t${DNID} * Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)})
07:59.00dan__tyea you're right.  i'll try that, seems liek that's proper.
07:59.05jsjcI wonder if in Skype for asterisk (hopefully somebody uses it) is there any way to check if a user in buddy list is connected or not.
07:59.14ChannelZjsjc: yes
07:59.27greezmunkeyChannelZ: qualify ??
07:59.38dan__tYep that looks correct, thanks for clarifying, ChannelZ.
07:59.47ChannelZgreezmunkey: do what now
07:59.49ChannelZ?
08:00.44greezmunkeyChannelZ: to check if a "buddy" is available, use the qualify=yes deal? I'm taking a swing here.
08:01.10ChannelZoh no.  There's an app for getting the buddy list that includes their status
08:01.17greezmunkeyo
08:02.13jsjcqualify?
08:02.25jsjcyes I know I think I can get full buddy list
08:02.34jsjcI dont know if comes with status or not.
08:03.02p3nguinqualify is typically used for NAT keepalive and/or portable UAs where Asterisk needs to know it has gone away.
08:03.05ChannelZIt does as I said
08:03.12greezmunkeyjsjc: don't take any advice from me, I'm on two weeks into this so far.
08:04.11jsjcI will paly around.
08:04.22jsjcI think i might be able with getting full buddy list.
08:04.39ChannelZSKYPE_BUDDY_FETCH()
08:05.20ChannelZYes you have to parse the list yourself, I don't think there is a function to request a specific single buddy.  You request the buddy list for an account, and then sift through that array to find what you want
08:05.53ChannelZalthough there is a config option to turn on and off presence..  uhm..
08:06.18ChannelZbuddy_presence=yes
08:08.53jsjcthat I think is for presence of the user logged in with asterisk
08:09.31ChannelZwell no it controls whether or not it will subscribe to presence updates for that user's buddy list
08:09.39jsjcI think I will try with it.
08:12.09ChannelZalthough I want to say there was a function or something for controlling your own presence..
08:12.53ChannelZSKYPE_ACCOUNT_PROPERTY(skypeaccount,status)
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08:13.32ChannelZThere's a list in the SFA Administration Manual PDF
08:14.13ChannelZhttp://www.digium.com/en/products/software/skypeforasterisk.php#documentation
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08:22.47dan__tSo, playing with the MYSQL application..... with "SELECT somedata FROM table WHERE ... ... ... ... LIMIT 1", am I correct in understanding that I should be able to use ${var1}, or literally use ${somedata}?  I believe the documentation might be
08:22.51dan__tambiguous.
08:23.08dan__tI know a piece of data is returned.
08:25.13ChannelZif it works like most SQL implementations it probably returns a handle or token to the result, which you then pass to another function to fetch each row
08:26.34dan__tPbttht.  You're right.  Could have sworn I did try that.  Must have been a typo.
08:27.46dan__tThank you.
08:28.07ChannelZI don't use SQL in * so I don't know the syntax
08:28.25ChannelZI should probably build it and play with it at some point
08:29.00dan__tYep no worries, you were correct.
08:29.26dan__tI know of RealTime, but before I can use that I need to know some other information taken from sql etc etc
08:30.31ChannelZhave fun.  Bed time for this bozo
08:30.42dan__tHave a good one.  Thanks gain.
08:30.44dan__tagain, too.
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08:31.16ChannelZno prob
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08:46.14Godfather_i want to disable some of the columns of cdr_mysql, how should i erase it? removing it directly from the table?
08:48.56Godfather_for example i dont want to store lastapp or the amaflags
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08:55.25Godfather_Standard columns may also be excluded now, simply by dropping the column name from the table, renaming the column, or aliasing the cdrname from a nonexistent variable (e.g. foo => amaflags).
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09:45.23Godfather_how can i cath the execution if the caller hang up before the timeout? i got exten => s,1,dial(sip/blah,45); s,n,noop(${dialstatus}); h,1,noop(${dialstatus});  but it works if the called party hangup or the timeout is finished, when the caller hangup i get exited non-zero, any ideas?
09:46.03Godfather_i tried to setting autofalltrough yes and no with the same result
09:47.54ectospasmGodfather_: whenever there's a hangup, it always exits nonzero
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09:49.02sam_affablehi
09:49.18sam_affableim using asterisk 1.8
09:49.23Godfather_ectospasm, then the question is, how can i cath the execution if the caller hangs up before the called party ?
09:49.58sam_affableit comes with jabber and gtalk right ? coz im getting No such command 'jabber' (type 'core show help jabber' for other possible commands)
09:50.09sam_affablealso core show help jabber
09:50.09sam_affableNo such command 'jabber'
09:50.20Godfather_sam_affable, did you compile it with the jabber dependences?
09:51.11Godfather_ectospasm, do you understand my point? i mean, the next execution after a dial is not executed if the caller hangs up
09:51.20ectospasmGodfather_: I don't think you'll be able to do that with DIALSTATUS, since Asterisk doesn't distinguish whether the near end or far end hangs up
09:51.57sam_affableGodfather_: aah thanks  even /usr/lib/asterisk/modules dont have res_jabber and res_gtalk
09:52.11sam_affablethanks will configure and install again
09:52.17Godfather_ectospasm, i just want to see ${DIALSTATUS} = NOANSWER.. i dont want to know you hangup
09:52.51ectospasmGodfather_: if Dial() is never executed, DIALSTATUS won't be populated for the channel.
09:52.52Godfather_the problem is that the next execution after a dial is not executed if the channel is not briged, i'm sure i can explain it better.. sorry for my english
09:53.14Godfather_ectospasm, ? i execute de command Dial()
09:53.35Godfather_but then, if the caller party hangup after for example 10 seconds and the called party doesnt cath the call
09:53.41ectospasmGodfather_: but if they hangup before Dial() is reached, it's not executed
09:53.43Godfather_then i cant be able to get the execution
09:54.54Godfather_ectospasm, aha, then, theres no posiblity to catch the execution if the caller hangs up before dial() is reached?
09:55.16ectospasmwhat do you mean "catch" the execution?
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09:55.49Godfather_ectospasm, continue in the dialplan
09:56.03Godfather_ectospasm, go to the next execution
09:56.31ectospasmGodfather_: it should hit the hangup ('h') extension when they hangup before Dial(), but DIALSTATUS won't be populated
09:57.09Godfather_ectospasm, like i said, i dont go to the h extension
09:57.16Godfather_*it dont
09:57.16sam_affableGodfather_: i followed this guide http://www.voiptoday.org/index.php?option=com_content&view=article&id=546:asterisk-18-how-to-configure-asterisk-to-googlevoice   and i see jabber.conf and gtalk.conf  in /etc/asterisk but in CLI jabber command says command not found
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09:57.38Godfather_just go to the h extension is the channel is bringed
09:58.28ectospasmit should execute the h extension when the channel is hung up, regardless of where
09:58.36ectospasmGodfather_: what version of Asterisk?
09:58.40Godfather_ectospasm, see that little dialplan http://pastebin.com/PhLqGLXT
09:59.05sam_affableGodfather_: that site dosent say anything about jabber dependencies
09:59.09ectospasmGodfather_: you have no h extension
09:59.32Godfather_ectospasm, i forget to paste it sorry
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09:59.58Godfather_ectospasm, http://pastebin.com/NB28DTes
10:00.32Godfather_sam_affable, after you do ./confnigure do a make menuselect and see inside if you have the dependences for jabber
10:00.45Godfather_ectospasm, i'm using 1.6.2
10:00.48ectospasmGodfather_: still doesn't look like it does anything
10:01.19Godfather_ectospasm, i got another context called hangup...
10:01.22ectospasmGodfather_: are you hanging up before or after the macro begins execution?
10:01.39Godfather_No...
10:01.54Godfather_the Dial is executed, i see ringing the called party
10:01.54ectospasmI'd think the h extension needs to go in [internal], not the macro
10:01.57sam_affableGodfather_: i dont see jabber option in make menuselect
10:02.24Godfather_ectospasm, let me try it
10:03.21Godfather_ectospasm, you are right
10:04.06Godfather_the problem was the macro return the execution to internal, thx
10:04.20ectospasmno problem
10:04.20kaldemarsam_affable: res_jabber in "resource modules" in the menu. depends on iksemel and openssl.
10:05.23sam_affablekaldemar: oh okie ty will check that
10:05.24Godfather_sam_affable, see inside channel driver, chan_gtalk
10:05.27Godfather_or something like that
10:05.35sam_affableokie will check that
10:06.51Godfather_sam_affable, dont forget to do a ./configure again
10:07.04ectospasmyeah, when you try to build a resource or channel driver module, if you don't have all dependencies met it will tell you what you need when you highlight it in menuselect
10:07.25sbszuluYes.
10:08.03ectospasmnote that your Linux distribution may name the dependencies different than Asterisk menuselect
10:09.35sam_affableGodfather_: yeah doing it again , ty
10:22.14Godfather_ectospasm, the dialstatus is populed to 'CANCEL' if the caller hangs up before the channel is reached
10:22.27Godfather_at least in 1.6.2
10:22.35Godfather_i see it now
10:22.50ectospasmActually, that makes sense now that I think of it.
10:23.24Godfather_ok :)
10:23.48WIMPyHa, quite some time ago wehad been puzzling about what CANCEL actually means. Now I know that.
10:24.22Godfather_WIMPy, lol :-)
10:25.11Godfather_CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up. (voip-info)
10:26.20WIMPyOh, something useful on voip-info. Wonder if that's documented that way elsewhere.
10:27.24drmessano~lawnmowers
10:27.27drmessano~lawnmower
10:27.33drmessano~bosch
10:27.33infobothttp://www.voip-info.org/wiki/view/Bosch+lawnmowers .. "Same to you, buddy!"
10:27.41drmessanoThere's your "useful" lol
10:28.17drmessanoMy fav bit of internet randomness
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10:41.55garymcHi anyone help me out. My Telco are telling me my PBX is messing up the ISDN30e. I have a sangoma A101D card using AsteriskNow on the server thats connected to it. ITs been fine all year a nd now im haveing problems.
10:42.05garymcIve just ran command "wanpipemon -i w1g1 -c Ta" and here is the output in a pastebin. http://pastebin.ca/2009373
10:42.15garymcHere is the output of "wanrouter hwprobe" http://pastebin.ca/2009376
10:42.31garymcAlso im using a SIP trunk as an emergency at the minute and its all working fine.
10:42.41garymcAlso I had a Sangoma techy SSH to my system on Tuesday the lovley lady in sangoma said it was my telco and not my equipment. What can I say to my Telco provider to make them understand?
10:44.03WIMPy"messing up" in what way? What's happening?
10:44.19kaldemarask them what they changed and then start the argument.
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11:07.44DelphiWorldhi
11:07.54DelphiWorlddoe someone called beerpages  pasted an marketing URL here?
11:11.12ectospasmDelphiWorld: I haven't seen it in this channel, but another channel earlier (about an hour ago)
11:12.00DelphiWorldectospasm: join please #freenode and complain
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11:55.06Godfather_is there any way to wait few seconds during execution when the call is not briged? in the h extension (like wait(), but this doesnt work in the h extension)
11:56.01WIMPyWhy do you want to wait when there's noone left to wait for?
11:56.47Godfather_WIMPy,
11:57.40Godfather_cause i need to convert a file using system (lame file.wav file.mp3)
11:58.08Godfather_and until the h extension is execution the script who mix both audio (in out)
11:58.13Godfather_doesnt start
11:58.41Godfather_in other words, when the channel is hangup sox mix the monitor(...,m)
11:58.59Godfather_and then i can do a System(lame file.wav)
11:59.00WIMPyBut you still don't know when it's finished.
11:59.24Godfather_WIMPy, yes, i dont know when sox finish it, but maybe waiting 5 seconds or so
12:00.02WIMPyWouldn't it be better to check for the files going into sox disappearing?
12:00.15Godfather_but sox doesnt start mixing it until the channel is hangup
12:00.30Godfather_then i need to do the System command into the h extension, and here i have to wait...
12:01.02Godfather_i can predict it seeing the time duration or so
12:01.33WIMPyWhen it goes in to the h extension or when it leaves the h extension?
12:01.42Godfather_the  point is that i do a h,1,lame(file.wav) and sox have not finished mixing the file (in out)
12:01.57ectospasmSystem blocks until the command completes
12:02.03WIMPyBut it has started?
12:02.40Godfather_ectospasm, but the problem its with sox, not with system
12:02.51Godfather_the system command is after the sox mix
12:03.02WIMPyMaybe it could make sens not to use mixmonmitor and do the mixing and encoding in one go?
12:03.45Godfather_WIMPy, ectospasm, http://pastebin.com/NnEBJ8vR
12:03.49ectospasmmixmonitor does it both in the command, but that can put undue load on the system.  If you use monitor, you can call sox separately
12:04.23Amnesiais there any possibilility to disable the default asterisk ivr when incoming calls come from pstn or w/e?
12:04.41Godfather_MixMonitor? let me try it
12:04.43ectospasmAmnesia: default IVR?  What do you mean?
12:05.03eduzimrshi guys, need hel, i got a YEALINK TI with its module, i want to monitor the SIP PEERS, i did that, but when a UNREGISTER a peer from * server the BLF continues "GREEN" how can i do when the peers get unregister to turn the light as "busy" ???
12:05.07WIMPyGodfather_: I'm pretty sure, BTW that you want to pass some options to lame.
12:06.03WIMPyeduzimrs: Tell your phone what colour you want in what state.
12:06.28Godfather_WIMPy, lame file.wav <- this convert file.wav to file.wav.mp3
12:06.50Godfather_i know theres other options, like setting the filename or the bitrate.
12:06.57Amnesiaectospasm: when I call to the number the pbx is connected to, I hear a woman saying congratiolations,you've configured blablabla
12:07.36ectospasmAmnesia: that's the demo
12:07.41eduzimrsWIMPy it doesnt have this options
12:07.59Amnesiayeah
12:08.01ectospasmAmnesia: for the channels those calls are coming into, you'll need to set it to some other context
12:08.06Amnesiahow can I disable it and let the phones ring?
12:08.19WIMPyGodfather_: I guess you use mp3 to save space and you can optimise that with some advanced options.
12:08.26eduzimrsWIMPy its a YEALINK T26P
12:08.37ectospasmAmnesia: it's gotta go somewhere, you have to define that in the channel driver config file, and also in the dialplan (extensions.conf et al.)
12:08.57Amnesiahm..
12:09.18ectospasmAmnesia: exactly where it goes is 100% configurable
12:09.21WIMPyeduzimrs: Heard of them, never seen one. But if you can't change it, you can't change it.
12:09.22Godfather_WIMPy, im sure of it, but i need first to write it into mp3 xD
12:09.25ectospasmAmnesia: ...in dialplan
12:09.32Amnesiauhu
12:10.16Amnesiausing exten => _0X.,1,SetCallerID(0344622033)
12:10.37Godfather_WIMPy, ectospasm , mixmonitor worked!
12:10.47Godfather_it mixes it on the go
12:10.48Godfather_:D
12:10.52ectospasmyep!
12:11.20ectospasmAmnesia: you should use Set(CALLERID(num)=0344622033) instead
12:11.35eduzimrswhen i unregister sip peer it get the state "UNAVAIBLE" but to the phone this state is like "ONLINE" understand, i need a "busy" statment when a unregister a sip peer to turn the light "red"
12:12.02Amnesiahm thx
12:14.02WIMPyeduzimrs: You could use custom device states that you set manually when the real phone state changes. Or you could get a phone that does what you want.
12:19.25Amnesiaectospasm: so I have to edit dahdi-channels.conf?
12:19.28ectospasmAmnesia: no
12:19.36ectospasmAmnesia: well, maybe
12:19.51AmnesiaI've finally got outgoing working
12:19.57ectospasmAmnesia: you need to set the context= lines to match your desired IVR in extensions.conf
12:20.09AmnesiaI havent got a IVR
12:20.19Amnesiait should just ring the phones
12:20.19ectospasm...then you need to match the incoming DID or the 's' extension, as necessary
12:20.48Amnesiahttp://pastebin.com/ngGvn5HD
12:20.56Amnesiathat'd my dahdi-channels.conf
12:21.24ectospasmAmnesia: change "context=default" to something else
12:21.42Amnesiabut channel 1 is used
12:21.45Amnesiaand not 2/3/4
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12:22.22ectospasmAmnesia: OK, you'll have to redefine [from-pstn], not [default]
12:22.31ectospasmI didn't look closely
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12:22.55Amnesiato some context I made in extensions?
12:23.02ectospasmsend calls that hit [from-pstn] to your ringroup
12:23.06ectospasm...or whatever
12:23.24ectospasmAmnesia: it's a bit old, but it still has a lot of useful content
12:23.32ectospasm~thebook | Amnesia
12:23.40ectospasm~thebook
12:23.40infobotfrom memory, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com
12:23.47ectospasmAmnesia: ^
12:23.54Amnesiahmkay
12:23.56Amnesiathx I guess
12:24.57eduzimrsWIMPy i was thinking to change the statment to "busy" in chan_sip.c when the peer get unregisterd
12:25.11eduzimrsWIMPy possible ?
12:25.29WIMPysure
12:26.16ectospasmthat may have unintended consequences, but that's the beauty of open source!
12:28.20eduzimrsectospasm u know the consequences?
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12:29.16ectospasmeduzimrs: like when you really want the UNAVAILABLE state rather than BUSY
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12:29.47ectospasmI'd assume in the unavailable case you'd want the lamp to go dim, not lit at all
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12:36.18eduzimrsWIMPy doing this, is it possible to habe problems in code ?
12:42.37Godfather_WIMPy, ectospasm, its very strange but the lame comand inside the System command removes 1 or 2 at the end of file, for example, file.wav has 20s, file.mp3 18s, isnt strange?
12:43.03Godfather_i do the same on the console (lame file.wav file2.mp3) with the same file.wav, and it has 20s
12:43.06ectospasmGodfather_: remember that lame is essentially a compression tool (-;
12:43.14WIMPyeduzimrs: You wil have to find out.
12:43.27Godfather_ectospasm, jajaja, it compress the seconds, lol
12:43.28Godfather_:D
12:45.24ectospasmGodfather_: also remember that running lame in System() vs. in the CLI will have different environments.
12:45.24Godfather_ectospasm, well, in fact, im pretty sure cause MixMonitor has not finished
12:45.36ectospasmcould be
12:45.50ectospasmMay want to put a Wait(5) afterwards
12:45.57Godfather_ectospasm, i cant
12:46.19Godfather_ectospasm, the channel is hang up, wait is not allowed here
12:47.21Godfather_h,1,Wait(5) <- not allowed
12:47.45ectospasmGodfather_: so execute lame as part of MixMonitor, rather than in a System() call
12:48.00ectospasmGodfather_: see "core show application MixMonitor" in the Asterisk CLI for details
12:48.48Godfather_ectospasm, i will use WaitUntil(epoch)
12:49.01Godfather_ty
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12:54.10eduzimrsWIMPy do u know some T.I device that supports the identification of the codes from asterisk
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13:14.16eduzimrs[TK]D-Fender hi, im going to get trouble if i change the code in chan_sip.c. I Want to change the state to "BUSY" when a i unregister a peer
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13:23.54z00daxhi guys
13:25.28z00daxI'm running asterisk-1.6.2; have a 'conf => 5000,1234,4321' in meetme.conf and have exten => 100,1,MeetMe(5000,cdMpPrs,) in extensions.conf
13:25.55z00daxwhen i call into 100, i get thrown straight into a new conference, the pin is never requested.
13:26.06z00daxis there anything else that needs doing ?
13:30.05glamz00dax: try to use "exten => 100,1,MeetMe(${EXTEN},MsI)
13:30.48kaldemarglam: what on earth was that?
13:32.28leifmadsenglam: he defined the conference room as 5000 but dialed 100, so ${EXTEN} is incorrect there
13:33.27leifmadsenz00dax: glam has the right idea though, and you should simplify your arguments to MeetMe(). Start with one at a time and continue to add them until you find the reason the pin doesn't work. Then you can re-evaluate the flag you added to see if perhaps it is causes the pin to no longer be required (or perhaps a bug that should be filed)
13:33.50glamz00dax: did it work?
13:33.54z00daxno
13:34.01leifmadsenI wouldn't have expected it to
13:34.05leifmadsenas I outlined above
13:34.06z00daxlet me drop the flags into the room
13:34.27leifmadsentaps the mic
13:36.07glamleifmadsen: mine worked.
13:36.22kaldemarglam: your configuration is different.
13:36.24leifmadsenglam: of course it did, because you probably have conference room 100 defined
13:36.35leifmadsenglam: he does not have conference room 100 defined -- he has room 5000
13:36.46leifmadsenwhich he is calling from extension 100
13:37.48z00daxleifmadsen: result.
13:38.27z00daxremoving all flags from the exten => call into MeetMe bring up the PIN request
13:38.56kaldemarz00dax: it's the d option.
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13:39.25z00daxkaldemar: the dynamic option ??
13:39.29kaldemarz00dax: "Dynamically add conference." - that makes asterisk ignore what you have in meetme.conf.
13:39.57leifmadsenkaldemar: +1
13:40.33leifmadsen'd' makes it so you don't need to define conference rooms in meetme.conf (dynamic). There are flags to still allow a pin to be requested I believe
13:40.42kaldemaractually, looks like you can't have PINs in dynamic meetme conferences at all, even if given as an application option.
13:40.44z00daxkaldemar: interesting. however, moving the PIN to MeetME(5000,cdMpPrs,1234) does not bring up pin request either
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13:41.14z00daxkaldemar: worth noting here is that it worked for me as is in 1.6.0 and its the u/g to 1.6.2 that blew it.
13:41.30leifmadsenkaldemar: interesting, I thought you could, but it's been a while. The best way then is to use Read() and use some prompts to request a pin.
13:41.30z00daxon the other hand, its always possible that 1.6.0 didnt do what it was meant to, so the bug *was* fixed :)
13:41.52leifmadsenMeetMe() doesn't necessarily have to handle the PIN stuff
13:41.57leifmadsenuse the other tools ;)
13:42.13kaldemarleifmadsen: the application documentation has no mention of it. kinda illogical.
13:42.18z00daxleifmadsen: so with read, move to confBridge ?
13:42.25leifmadsenz00dax: I didn't say that
13:42.29leifmadsenbut you could
13:43.05kaldemarignoring the PIN option in dynamic conferences sounds more like bug than using it.
13:43.42z00daxyeah, if its on the config line, use it; sounds logical - or break saying 'not supported'
13:43.59z00daxleifmadsen: looking for more of a recommendation than anything else
13:44.36leifmadsenz00dax: use Read() and MeetMe() or ConfBridge()
13:44.49z00daxta
13:44.52leifmadsenanyone is welcome to file a bug if they find an issue :)
13:45.01leifmadsenif they're not filed, they can't be fixed
13:46.46z00daxtrue
13:46.49carrarword!
13:47.51Godfather_leifmadsen, how can i wait 3 seconds in the h extension? i can't use wait or waituntil cause both arent allowd if the channel inst active
13:48.28leifmadsenGodfather_: really? I don'[t know...... ${SHELL(sleep 3)} ?
13:48.53leifmadsenNoOp(${SHELL(sleep 3)})
13:49.14Godfather_ok, i'll try it
13:49.16leifmadsenseems odd Wait() couldn't be used...
13:49.25Godfather_leifmadsen, this is the problem, http://pastebin.com/7rYnPTun
13:49.50leifmadsenthat doesn't show me a problem
13:49.59leifmadsenjust add...
13:50.02leifmadsensleep 3 ;
13:50.02Godfather_it seems MixMonitor hasnt finished to mix both files, and always i lost a few seconds at the end
13:50.08leifmadsenin your SYstem() line
13:50.25Godfather_1 sec, i try it
13:50.26leifmadsenyou're already calling the shell, so add a sleep 3
13:50.32leifmadsenI suggst you just use an AGI() at that point
13:54.14Godfather_leifmadsen, yours work
13:56.30Kattygoooooooooooood morning beautifuls!
13:56.33Kattyit's FRIDAY!!!!!!!!!!!
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13:56.48WIMPyGodfather_: Now you have to make a 3 hour call and see if 3 seconds are still enough :-)
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13:59.14Godfather_WIMPy, yes, they are enough cause mixmonitor do it on the go, and when the system(lame..) is executed alway reamins 1 or 2 seconds (since my experience from 1h ago)
13:59.32Godfather_1 or 2 seconds to mix
14:00.05leifmadsendoes anyone here have experience in configuring SElinux for a non-root Asterisk install?
14:00.06Godfather_*i'm not pretty sure about that last but i have to believe it
14:00.30russellbleifmadsen: the Fedora package of Asterisk probably does that
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14:01.31Kattyhugs leifmadsen
14:01.33Kattyhugs russellb
14:03.24leifmadsenrussellb: hmmm, wonder if I could find that info in the RPM script....
14:04.09kaldemarz00dax: btw, you can replace the d option with D if you want to use dynamic conferences, it is supposed to ask the PIN.
14:04.34eduzimrs[TK]D-Fender Thanks for helping!!! i´ll enjoy your hints
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14:08.13waKKu~devstate
14:08.13infobotfunc_devstate is a module included with Asterisk 1.6 and above.  A 1.4 backport is available here: http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c
14:08.26waKKuty infobot ;)
14:09.38z00daxkaldemar: thanks
14:09.51eduzimrs~devstate
14:09.51infobotfunc_devstate is a module included with Asterisk 1.6 and above.  A 1.4 backport is available here: http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c
14:10.03z00daxi went with a read() and confBridge option, lets see how this works out
14:18.18kaldemarz00dax: between 1.6.0 and 1.6.1 meetme stopped asking for the PIN with dP. D works though.
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14:30.28cipherpunkhey folks!
14:30.39cipherpunkis there a way to use an external script in the dialplan to determine the number to call?
14:30.56cipherpunkif yes, can you give a small hint what to look for in the docs?
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14:34.04pabelangerCan you access ASTETCDIR and ASTVARLIB from the dialplan?
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14:49.44z00daxthanks for the help earier guys.
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14:56.01kaldemarcipherpunk: core show function SHELL.
14:58.12kaldemarpabelanger: func AST_CONFIG. if that was what you meant.
15:07.31cipherpunkkaldemar: shell is not included in the asterisk i have to use here, which is 1.4
15:07.55cipherpunkbut [TK]D-Fender showed me an even better way to accomplish what i wanted! thank you very much [TK]D-Fender!
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15:20.17RickCogleyhello, new to asterisk, using 1.6.2.14 via asterisknow install. Wondering, is 1.8 "better" - should I upgrade? I am connecting via SIP to a gateway device, and the basics are working fine, but, there seem to be some bugs. Using latest freepbx too.
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15:25.33SuPrSluGRickCogley: using 1.8 with FreeBPX may break a lot of things. Wait for them on that or use asterisk 1.8 w/o gui.
15:26.27RickCogleyhmm, ok, thanks SuPrSluG
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15:47.46nunnersUsing queues, is there a way of "queueing" the call without actually picking it up until a user is ready?  At the moment, the system picks up the call, then puts it in the queue, and plays a phone
15:48.09nunnerswhoops... pressed return before finished... it plays a phone sound in the background... is there another way?
15:48.17leifmadsennunners: phone sound?
15:48.32nunnersliefmadsen: yep...
15:48.44leifmadsenyour answer still doesn't help me understand what you mean by that
15:48.52leifmadsennunners: phone sound?
15:49.27nunnersleifmadsen: it plays the sound of a phone ringing...
15:50.29leifmadsenthen it must be trying to connect to a phone
15:50.44leifmadsenwhich means the queue thinks the phone is available
15:51.01leifmadsenhow are you connecting to the phone? Local channel? Agent channel? SIP channel?
15:51.43nunnersoh... ok I'm with you.... and actually yes, I don't think I've had a circumstance yet that has an extension unavailable... so yes that makes sense...
15:52.12nunnersthanks
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15:58.07Godfather_why the dialstatus of dahdi  is answered if the called party doesnt take the call? there is any way to know if the call was answer or no?
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16:00.28ChannelZdepends on the type of line.
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16:00.58SuPrSluGa call can be answered before a phone ever rings
16:01.06ChannelZPOTS doesn't really have useful call progress (you can use ring and busy detection and such but it's flakey)
16:01.19pabelangerSuPrSluG: Yes, Progress()
16:02.30Godfather_ChannelZ, pstn
16:03.23ChannelZYah.  It's not ideal
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16:04.29Godfather_ChannelZ, how should i use ring/busy detection?
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16:06.19ChannelZcallprogress in chan_dahdi.conf
16:06.35ChannelZand busydetect
16:06.45ChannelZYMMV
16:09.40russellbyour mileage will most likely lead to randomly dropped calls
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16:11.51Godfather_russellb, during a call?
16:12.02russellbyes?
16:18.43eduzimrs~devstate
16:18.43infobotfunc_devstate is a module included with Asterisk 1.6 and above.  A 1.4 backport is available here: http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c
16:18.59Godfather_seems working,  i will lead with it.
16:19.51Godfather_eduzimrs, this applies to me?
16:22.01nevermind_quacksuperm1: trying your gv workaround on a remote machine ... how do I call your agi from the command line / queue a call to it?
16:22.35superm1nevermind_quack, if you call it from command line, you'll need to input a few things to stdin that normally are inputted by asterisk
16:22.39superm1agi_channel:'xyz'
16:22.47superm1agi_dnid:'number'
16:22.51superm1number is the number you are dialing
16:22.59superm1agi channel just needs to be there for the script to not fail
16:23.10superm1hit enter a few times and it should do a call back
16:25.52nevermind_quackk thx
16:28.54simplydrewcan anyone help me with a streaming music on hold issue I'm having? I'm trying to setup mpg123, but for some reason trixbox won't untar with the command tar –zxvf mpg123-0.59r.tar.gz with the guide that I'm following
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16:40.02timholumI am trying to configure a polycom 335 to be used in remote locations ( through nat ) my asterisk server is not behind a nat. it works for maybe a minute or two when the phone first starts up, but then it goes offline I have tryed setting an expires setting in the phone, ext.., but no luck. anyone have any idea's?
16:41.27SuPrSluGyou can try using qualify. that may help. it may be the router/firewall it's behind closing the connection though
16:43.52timholumSuPrSluG: ill give that a shot, thanks
16:44.03_Corey_timholum: configure the phone to re-register at a shorter interval
16:44.19_Corey_timholum: quality probably won't help too much in that situation, as you want the phone to keep punching out
16:44.30nevermind_quacksuperm1: i will be out for the rest of the day probably. thanks for that workaround, I'll be checking the bug and probably popping in here if I have any great new ideas. ;)
16:44.43_Corey_timholum: Check the registration expire interval in your polycom config... i'd try 60 or 90 secs
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16:44.52timholumany idea where I tell the phone to re-register?
16:44.59superm1nevermind_quack, , okay cool. i'll idle here too
16:45.43*** join/#asterisk QubeZ (~nkasu@97.104.194.44)
16:45.44QubeZhello all
16:46.31QubeZI have a client from the outside registering over SIP that is having an issue I cannot resolve.  It shows "SIP client x.x.x.x registered", then immediately after that it shows "Unregistered SIP <extension>"
16:46.44QubeZi tried to register from the outside using a soft phone with same credentials and it works
16:47.12_Corey_timholum: Are you configuring with an xml template or using the phone's web GUI?
16:47.19QubeZsip debug on their ip shows 401: Unauthorized.  I'm not sure why it would when the configurations for the polycoms are pulled from our web server and I've validated the information is correct ie: user / pass
16:47.23timholumphones web GUI
16:47.52timholumI think i just figured my issue out
16:47.57LinuturkWhat are some common strategies in dealing with faxes?
16:48.14timholumI had Expires set under outbound proxy, not the phones line1
16:48.18Linuturkand, what can asterisk do with a fax it receives?
16:48.27*** join/#asterisk timahvo1 (~rogue@41.223.57.75)
16:48.51_Corey_timholum: We find that usually fixes remote phones becoming unreachable, good luck
16:49.12timholumthanks
16:50.45*** join/#asterisk Jasnejac (kvirc@81.91.107.235)
16:51.15JerJerLinuturk:  anything you want
16:53.01JerJermy fax solution copies the TIFF file to an NFS mount, then notifies a web service about the fax
16:53.40JerJerthen the web service handles the fax based on how the customer has it configured
16:54.51dan__tWell this is badass, my first real project with *.  Used a lot of that MYSQL application, a lot of dialplan logic.... feels good heh.  Little victories.
16:55.19cmnkydan__t, congrats
16:55.19JerJerdan__t:   mysql in the dialplan?  I seriously hope not
16:55.21LinuturkJerJer: did you write aGI scrips to do that?
16:55.22*** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein)
16:55.27leifmadsendan__t: you should use func_odbc instead of MYSQL()
16:55.42dan__tMYSQL app in the dialplan, yes.
16:55.47dan__tI'll check it out, leifmadsen.
16:56.18leifmadsendan__t: http://ofps.oreilly.com/titles/9780596517342/ch15.html#Database_id287624
16:56.20dan__tAfter all, you're a trusted resource.  haha.
16:56.28leifmadsenI'm no such thing
16:56.40dan__tI know, they were lies.
16:56.51JerJeractually he should think a bit more out of the box
16:56.57leifmadsenMYSQL() causes ugly dialplan code, is not as stable as func_odbc, and is less flexible.
16:57.20leifmadsenI need lunch
16:57.42dan__tIt is ugly as hell.
16:57.56dan__tI like func_odbc, I'll refactor using that.
16:58.13dan__thah, I didn't know about same => either.
16:58.49leifmadsen<3 same =>
16:59.08dan__tLunch?  Sounds like a good idea.  Its on me next time yer in Phoenix.
16:59.13JerJerI like using 'middleware'  - creates very simple dialplan and can scale
16:59.28leifmadsenexten => _[n]a[n]cy-NXXNXX-[n]oc,1,NoOp()
16:59.33leifmadsensame => n,NoOp()
16:59.42dan__theh
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17:04.53dan__trad thanks again.
17:10.39Corydon76-homeloves when leifmadsen loves stuff he wrote... it's as if leifmadsen were loving another man... ;-)
17:11.11JerJertmi
17:11.23cmnkythanks for that imagery
17:11.25russellbCorydon76-home: really?  in #asterisk?
17:11.38Corydon76-homeHeh
17:11.46Corydon76-homeI kid.
17:12.20carrarAs opposed to a different channel name...
17:13.03Corydon76-homeYes, but leifmadsen wasn't loving my contributions in a different channel name...
17:13.11gslima1980hello, i'd like to know if it's possible to use asterisk with the above configuration:
17:13.11gslima19802 lines for receiving calls
17:13.11gslima19802 analog phones connected to a single ramal
17:13.11gslima1980autoreceiving calls
17:13.11gslima1980to program asterisk to pass calls from known numbers to this ramal
17:13.13gslima1980provides a 5 sec delay before hanging-up for unkown numbers, if they don't provide a code (the ramal number).
17:13.16gslima1980I live in Brazil, I don't know if callerId works the same way USA's
17:13.51Corydon76-homegslima1980: what's a ramal?
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17:14.49gslima1980branch
17:14.56gslima1980is the same thing
17:15.54dan__tYeah, that was a pretty disgusting visual.
17:16.07Corydon76-homegslima1980: based upon callerid?
17:16.48Corydon76-homeI said loving, not f***ing
17:16.53gslima1980yes, but i don't know if callerid works in Brazil the same way as other places
17:17.48*** join/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n)
17:17.49*** mode/#asterisk [+o Cresl1n] by ChanServ
17:18.14Corydon76-homeGet your collective minds out of the gutter
17:19.24Corydon76-homegslima1980: it's doable, yes, even if callerid works differently
17:20.33*** join/#asterisk oneseventeen (~onesevent@web64.webfaction.com)
17:21.03gslima1980i'm a begginer programmer, what do I need to start this small project?
17:21.19Corydon76-home~thebook
17:21.19infobotit has been said that thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com
17:21.47Corydon76-homegslima1980: read the documentation ^^^^^
17:22.02oneseventeenI'm trying to set up AsteriskNOW as a VM to play with, what version of linux does it use?
17:22.35Corydon76-homeoneseventeen: centos, I believe.  Or Fedora
17:23.39gslima1980thanks for the help
17:23.40*** join/#asterisk Joelio (~joel@lcw-212-23-5-1.zen.co.uk)
17:23.47JoelioHi, is anyone aware of a simple ruby config parser for asterisk? (not a full blown product like adhearsion).. inherited a cludgy mess of a voip platform where configs get added manually
17:24.36dan__tleifmadsen, thanks again for that link, that's a great guide.
17:24.40dan__tAnd good luck in March :)
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17:26.03dan__tSo I was reading that Vitelity doesn't recommend using IAX2 with them.... I don't remember that being the case with them in times past.  IAX2 isn't going away or anything, is it?
17:26.50dan__tNot trying to pick on Vitelity at all, just saying, that's the example I'm using.
17:32.14JerJerVitelity does not know how to scale with IAX
17:32.26p3nguinThey probably don't keep their stuff updated, and IAX2 seems to perform better with like versions of Asterisk on each end of the interconnect.
17:32.40dan__tah,
17:32.44dan__tfair enough.
17:32.58dan__tAgain, not bagging on Vitelity, just curious.
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17:39.11Linuturkfxs is a phone. fxo is an incoming pot?
17:39.45p3nguinPhones are FXO.  Phone wall jacks are FXS.
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17:40.20Linuturkfxo are powered then, and fxs don't require power
17:40.34Linuturkbah
17:40.37Linuturkreverse that
17:40.45p3nguinFXS provides power, FXO receives power from the FXS jack.
17:41.05LinuturkfxSendPower
17:41.09Linuturkgot it
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17:56.34Linuturkhttp://store.digium.com/productview.php?product_code=1TDM2460EF << does anyone use this in production? how does asterisk recognize this device, and how do I route calls to and from this device?
18:04.44leifmadsenLinuturk: do you already own that device?
18:05.13leifmadsenthe device is recognized via installing the DAHDI drivers, and is accessed through the chan_dahdi.conf configuration file in Asterisk
18:05.17Linuturkno leifmadsen
18:05.45leifmadsenI have a few in production at a clients call centre
18:08.41Linuturkdo they work well? how would you compare them to a sip gateway?
18:08.48Linuturkor, analog channel banks?
18:08.58*** join/#asterisk mikeones (~mikeones@pool-71-252-169-67.dllstx.fios.verizon.net)
18:09.56LinuturkI'm looking to get rid of 2 analog channel banks and either a) replace them with a fax server and old sip phones for the lobby OR b) get a more compact analog solution in place.
18:10.39mikeoneshello, if I get a linksys spa2101 and plug it into a phone jack on my loop inside my house, will all te jacks be active then?  Does the linksys have to be fxo?
18:12.39Linuturkleifmadsen: ^
18:15.10cmnkymikeones, SPA2101 ... i believe ... does not plug into the telco line ... just lets you connect phones -> net
18:15.21cmnkyi could be completely wrong tho
18:16.07cmnkyi have a SPA2102 sitting here ... and as far as I know ... its 2 phones -> net only ... no line port
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18:16.45p3nguinIf you connect your device's FXS port to your wall jack, you will probably damage the device.
18:18.07p3nguinIf your device has an FXO port, and you connect the FXO to your wall jack, now the device will act like a phone and be able to receive calls from the PSTN.
18:18.53*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
18:20.28p3nguinIf you disconnect your telco's drop in the NID to isolate your house wiring from the powered line, you can connect your device's FXS port to your wall jack to "activate" all the jacks in the house with the Linksys device.  You'll then have to get your phone service on the device via IP.
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18:34.19Kattyhai
18:35.43citywokohaithar
18:36.34cmnkyno ... how *you* doin
18:37.40Godfather_ChannelZ, with callprogress=yes i dont hear anything
18:38.19Godfather_the detection of the answer is correct but theres no audio, neither the dial tones
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18:41.24drclueFrom within the AMI interface , does anyone know how to get a list of the recorded calls in /var/spool/asterisk/monitor?
18:42.39citywokls /var/spool/asterisk/monitor -- afaik asterisk doesn't keep any information on the files it stores in there
18:42.42leifmadsenCommand with a value of !ls /var/spool/asterisk/monitor ?
18:43.04leifmadsenjust a guess
18:46.27drclueI guess I could go at it from !ls  sorta seems a bit messy , but thats all that has occurred to me too.
18:47.47p3nguinWhat information are you looking for?
18:48.17drclueManaging the recordings is about all I have left to do on this Google/HTML5 based HUD I'm doing.
18:49.32citywoki use a daemon to monitor the CSV CDR log and use all that information, transcode the recordings to MP3, and save all the information in SQL
18:49.37drclueWell, it would be nice if there were a clean way to list and associate recordings, but thus far I'm figuring to settle for hiding data in the filenames and using !ls to get the list
18:50.17citywokdrclue: use a script in the hangup extension which has access to all of the information, and store the information in a database somewhere
18:53.14drclueWell basically my project "AsterClick" is a generic tool that uses webSockets to allow robust realtime web applications. The project even affords a tool that deploys those web applications as stand alone desktop executables. One of the things I have managed to do along the way is keep it largely out of the dialplan
18:53.45citywokthen use a daemon and watch the CDRs
18:56.48*** join/#asterisk d-fens (~jscalia@su-nat.int.smq.datapipe.net)
18:56.50drclueWatching the CDRs is probably a bit problematic in that the web clients come and go, so unless I want to add a lot of storage overhead I guess the !ls and embedding basic data in the filenames is my best bet for now.
18:58.04p3nguinMy recordings have file names containing the date, time, and phone number(s).  What more info do I need to know about the recordings?
18:58.27citywokwho made/answered the call?
18:59.06citywokwhat queue they were logged in to at the time, information to link the recording back to the Dial button in the agents interface, who the call gets billed too, etc.
19:01.06drclueI'm embedding information in the filenames already, much akin to that mentioned by citywok, but just wanted to make sure that I was not missing some logic  built into Asterisk for this type of thing
19:04.51neurosyshow can you see your kernel timin on linux?
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19:08.40cmnkyneurosys, the kernel config values are exported in /proc somewhere, config or config.tar.gz i think .. if its enabled
19:09.19cmnkyneurosys, although it might be elsewhere, and that might not be the best method of getting it .. its 1 way to skin the cat
19:09.29neurosysim checking
19:10.17cmnkyi think the config var has 'TIMER' in it
19:10.18cmnkyfor grepping
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19:11.49leifmadsenwtf... I have a member in a queue, and because I can't Dial() then from the Local channel, the caller is kicked out of the queue, even though I have leavewhenempty=no..... hmmm
19:12.18citywokreally? that sounds like a bug
19:12.23leifmadsenya
19:12.29neurosyswell, the TIMER var doesnt say much :/
19:12.30citywokdoes it happen if the member doesn't answer?
19:12.31leifmadsenI don't remember it doing this to me yesterday though
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19:12.44citywoki'm assuming thats what you mean by can't dial them
19:12.51leifmadsencitywok: the member is an Agent channel called via Dial() from the Local channel. The agent isn't logged in right now.
19:13.13citywokhmmm.  retrys maybe?
19:13.16cmnkyneurosys, try 'cat /proc/timer_list'
19:13.29neurosysnope
19:13.41cmnkyneurosys, i see some values about halfway down on my box that look like 300MHz
19:14.04neurosysThis is CPU timings with the kernel . not CPU Speeds
19:14.10cmnkyneurosys, cpu: 0
19:14.11leifmadsencitywok: I didn't define that value
19:14.13cmnky.resolution: 3333116 nsecs
19:14.20leifmadsencitywok: I want it to try indefinitely
19:14.52cmnkyneurosys, i know which value you are talking about
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19:15.14citywokah, for our queues we have just a couple retries and maybe 2-3minutes of hold and if no agents answer we shoot it to voicemail
19:15.19cmnkyneurosys, i just cant find it for you, as all mine are 300MHz w/ /proc/config info disabled
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19:15.47citywokHmmm. thinking of that is there any way to get whether or not a voicemail was recorded in CDR's, rather than just app=Voicemail ?
19:15.47leifmadsencitywok: ya but if the status of the Local channel is Not In Use it'll keep trying it anyways
19:16.19Linuturkthanks to [TK]D-Fender
19:17.52leifmadsencitywok: is there a ${VOICEMAIL_STATUS} or something channel var you could write to a field?
19:18.11citywokhmm. not sure. let me look and see what's set after
19:18.37*** join/#asterisk infinity_ (~brendon@web2.artsopolis.com)
19:19.16infinity_how do i get the recommended sip gateway list from the bot?
19:19.39citywokVMSTATUS=SUCCESS
19:19.41citywokVM_MESSAGEFILE=/var/spool/asterisk/voicemail/default/1593/INBOX/msg0000
19:20.31citywokHmm. I could do it with simple dialplan logic as well, if a voicemail isn't left that's generally because the caller simply hung up. so after voicemail() executes i'm pretty sure they left a message
19:20.43infinity_who is the bot
19:20.48citywokinfobot?
19:20.49leifmadseninfobot is the bot
19:21.11leifmadseninfobot: tell infinity_ about itsp-us
19:21.23leifmadseninfobot: tell infinity_ about itsp-list-us
19:21.32leifmadsendamn, I don't remember
19:21.36citywok~itsp
19:21.36infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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.
19:21.42leifmadsenah no hyphen
19:22.12citywokif you do that will infobot PM the person or say it to the channel?
19:22.15infinity_citywok: thanks
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19:26.17leifmadsencitywok: heh, I had an Answer() in the Local channel :)
19:26.39citywokoh no
19:26.41citywoklol
19:26.45leifmadsenit'll do it :)
19:26.52citywokyea that'd definitely screw it up
19:27.38citywoki just modified cdr_custom, will logger reload reload that for me?
19:27.49leifmadsencitywok: cdr reload ?
19:27.53leifmadsenmodule reload cdr_custom.so ?
19:27.59leifmadsencant' remember
19:28.05leifmadsenI doubt logger reload is what you want
19:28.12leifmadsen(that would be logger.conf)
19:28.28citywokhmm no cdr reload, and i'll avoid reloading the entire module during production just in case
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20:32.33Yalowahello
20:32.40Yalowai have a problem
20:32.45Yalowaplease help me
20:33.26Yalowaim using asterisk 1.6 and astercc-crm
20:33.46Yalowabad im not call to another extension from astercrm
20:33.50Yalowai have logs
20:34.03Yalowaeverbody here ?
20:34.47Yalowa:\
20:36.13Corydon76-homeYalowa: please do not message channel operators.  We are not here at your beck and call.
20:36.54Corydon76-homeYou need to better describe the issue you're having before anybody can assist
20:38.03Yalowaokey
20:38.28Yalowa[Dec  3 22:38:09] ERROR[26184]: utils.c:1175 ast_careful_fwrite: fwrite() returned error: Broken pipe
20:38.47Yalowai call from astercrm 6000 extension to 6001 extension
20:38.51Yalowabut i so that log
20:38.54Yalowaon asterisk
20:39.27Yalowai guess conf problem but i check, no poblem here
20:39.27Corydon76-homeAnd what's at that extension?
20:39.47YalowaSIP
20:40.10Yalowaim call my softphone to another softphone. this is runing
20:40.14Corydon76-homeWell, that error is not from the SIP driver.  It's likely to AMI
20:40.33YalowaAMI ?
20:40.42Corydon76-homeAsterisk Manager Interface
20:41.01Yalowaokey i check asterisk gui
20:41.06Yalowabut no see to problem
20:41.18Yalowayou see that problem once ?
20:41.51Corydon76-homeAt this point, you'd probably be better off seeking assistance from astercrm, whatever that is, as this likely relates to their system
20:42.47Yalowaim understand
20:42.49Yalowathank you Corydon76-home
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20:57.57CherebrumAnyone know anything about German E1 circuits?
20:59.06CherebrumDeutsche Telekom installed a PMX (smart jack?) and I've connected my Cisco E1 interface to the S2M(2 pair E1 interface) on the PMX and I get no link. :(
20:59.37WIMPyYou installed Asterisk on a Cisco? :-)
21:00.13CherebrumWIMPy: no. that's stupid
21:00.57WIMPyThen you're obviousely wrong here. But make sure you have net5 signalling configured.
21:01.12CherebrumWIMPy: I don't even get layer2
21:01.20Cherebrumnet5 is layer3 or 4
21:01.42CherebrumIs it general practice to have to use a crossover cable?
21:01.50Qwellit depends
21:01.54WIMPyno
21:02.54WIMPyA normal straight patchcord should be the right one.
21:03.16Cherebrumand the S2M is the right interface on the PMX?
21:03.25CherebrumThere is another RJ45 jack on it labled T3ab
21:03.38Cherebrumbut from what I read online the S2M interface is right
21:03.45WIMPyCherebrum: PMX is the german abbreviation for PRI.
21:03.52Cherebrumaha
21:04.21WIMPyPrimärMultipleX
21:04.56CherebrumThey installed a very odd smartjack. It's not the card type the mount on the wall. It's a little box white box that we have to put on a shelf
21:05.37WIMPyA G.SHDSL modem.
21:06.02WIMPyBut you can get them in 19" 6RU versions as well.
21:06.15Cherebrumthere isa green RJ45 from DT connected to an interface labeled UKS
21:07.09WIMPyThe U interface is to the CO.
21:08.22Cherebrumright. that's the long distance single pair interface
21:10.02WIMPyAnd where do you have a crossover cable? And what is the status? Red alarm on both ends?
21:10.26Cherebrumwe don't have a crossover cable
21:10.33CherebrumI wish I had one to try
21:10.53WIMPyYou shouldn't need one.
21:10.53Cherebrumthe guy on site in germany doesn't have the tools to make one either
21:11.01Cherebrumright
21:11.10Cherebrumso my guess is DT didn't fully turn up the circuit
21:11.33WIMPyThe modem should tell.
21:11.33Cherebrumthe only green light on the box is labeled "EIN"
21:12.06CherebrumS2M is red, E1 is off, UKS is red, FSP is off
21:12.23WIMPyIf it's the box I think it is, that would be good. The others are fault indicators.
21:12.45WIMPyred should be a red alarm.
21:12.57Cherebrumit's a white box. it has the orange T mobile logo on it
21:13.12CherebrumI only have a picture of it
21:13.31Cherebrumpart of the model is covered by a sticker but the last part is MSKU-T
21:13.46*** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net)
21:13.50WIMPyFour feet going up the whole height of the side?
21:14.00Cherebrumit's a very small box
21:14.16Cherebrumsize of a 13" laptop
21:15.01WIMPyAnd about 2" high? You call that small?
21:15.24Qwellsmall for legacy telephony equipment
21:15.28WIMPySo what's the status on the cisco?
21:16.04WIMPyOther DSL modems are the size of two packs of cigarettes.
21:16.19CherebrumE1 0/0/0 is down.
21:16.40WIMPyLA? RA?
21:16.50CherebrumTransmitter is sending remote alarm.   Receiver is getting AIS
21:17.38WIMPyAIS? Is that cisco for a yellow alarm?
21:18.35QwellWIMPy: blue
21:18.49Qwellyellow is RAI
21:19.15CherebrumAccording to Cisco docs: A receive (rx) Alarm Indication Signal (AIS) alarm indicates an error occurring on the T3 line upstream from the equipment that is connected to the port.
21:19.22Cherebrumso I'm getting a remote alarm
21:19.26WIMPyhmm. Am I mixing that up?
21:20.01Qwellthere's an error on the upstream line
21:20.07Qwellit's passing that to you as blue
21:20.23WIMPyAnyway: It means it's receiving something. So that's not too bad.
21:22.03WIMPyMight be you just have to start communicating.
21:23.30Cherebrumyea.. it's a blue alarm
21:25.35Cherebrum<PROTECTED>
21:26.29CherebrumWe just moved this Cisco from down the street where it was connected to a different DT E1 PRI circuit. The config hasn't changed and it worked down the street
21:27.36*** join/#asterisk BenC[UK]_ (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com)
21:32.25cjkhi i have double dtmfs on pri2pri calls when dtmfs come in fast (redial, or fax machine).  any hint is welcome
21:32.41*** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net)
21:33.02CherebrumWIMPy: thanks for your help. I've got the guy on the phone with DT now
21:33.07oneseventeenany preferred software SIP phones?
21:33.24i_heart_asteriskis their an option for a caller to stop recording a voicemail and discard it ?
21:34.40citywokhang up?
21:34.47citywok~softphones
21:34.54citywokahh, no ~softphone tag infobot? sad
21:34.56Qwellhanging up during recording would save it
21:35.10*** join/#asterisk NickNick (~nicholas@93-97-188-195.zone5.bethere.co.uk)
21:35.11citywokoh right, i was thinking of record() nto voicemail
21:35.13*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
21:35.21WIMPyi_heart_asterisk: # and you need to enable review.
21:35.33i_heart_asteriskQwell: yes, once you start recording a voicemail, there is no way to discard it, only re-record
21:35.36WIMPyIt will actually tell you.
21:36.00WIMPyThere is no discard?
21:36.09i_heart_asteriskWIMPY, no there is not
21:36.25Qwelleven with review enabled?
21:36.46_Corey_you could re-record a short blank message perhaps
21:36.55i_heart_asteriskyes, you can push # during recording and it gives you several options, send vm, listen to vm, re-record
21:36.55_Corey_if you have a minsecs set ...  might work
21:37.02WIMPyYou just press # and hang up.
21:37.25i_heart_asteriskWIMPy, i believe it will still send the VM, i'll test right now
21:37.36WIMPyNope
21:37.40*** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net)
21:37.41WIMPyJust tried it :-)
21:38.26i_heart_asteriskWIMPy, i just tested and it still sent the VM
21:39.12i_heart_asteriski pushed '#' during recording and i hung up  as it was giving options
21:39.21i_heart_asteriskbut the VM was still sent
21:40.21oneseventeenI just installed AsteriskNOW with FreePBX on a VM and want to test out connectivity... wasn't sure if anyone knew of a good guide
21:40.22WIMPyHmm. You're right. Maybe my first message was too short.
21:41.03_Corey_You're probably going to have to choose to re-record and then hangup
21:41.45*** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net)
21:41.51i_heart_asteriskwould be nice for a discard button
21:41.53WIMPyOr add a new feature.
21:42.01i_heart_asteriskmaybe i'll do it
21:42.17i_heart_asteriski've gotten to know the voicemail code pretty weel
21:45.18*** join/#asterisk neurosys (~neurosys@173.200.203.58)
21:52.57CherebrumWIMPy: What are the chances of DT being able to resolve this for us at 11PM on a Friday in Germany?
21:53.24WIMPyDepends on your contract.
21:53.53WIMPyBut I'm still unsure as to how to interpret the output from the cisco.
21:53.55citywokand how busy they are
21:54.03citywokhas the line ever worked before?
21:54.15WIMPyThat still depends on your contract :-)
21:54.17CherebrumWIMPy: This is a new circuit
21:54.35CherebrumWIMPy: we just moved the Cisco from the old DT cicuit down the street to this new building
21:54.57Cherebrumand DT already moved the DIDs to the new Circuit
21:55.38Cherebrumthey moved the DIDs while we were still moving the Cisco
21:55.59WIMPyYou could call one of the numbers ans see what cause you get.
21:56.43citywokyea if it's a new circuit it's probably broken. lol.
21:56.58*** join/#asterisk neurosys (~neurosys@173.200.203.58)
22:01.34citywokthe supermicro 1u dual atom is going to make a great asterisk box.  now just to pick the distro for it
22:01.51Qwelltrixbox, obviously
22:01.53Qwellduh
22:02.02WIMPyalso plans to use one of those.
22:02.18*** join/#asterisk mayfield (~mayfaild@76-250-152-203.lightspeed.snantx.sbcglobal.net)
22:02.21citywoki just ordered it off of newegg w/ 4gigs of memory (it was included lol), and an X25-V SSD
22:03.07citywokheh, i'm going to play with *now and see how well it works.  I've never used it.  And I've only touched 1 trixbox i didn't even set up.
22:04.08citywokWIMPy: i'll let you know how it goes.  I'm gonig to bench it and see how many concurrent calls it can handle.
22:04.25citywokI got the most expensive one, the D525 for $400, lol.
22:05.46WIMPyI think the one I found was only a 330. But let's see what's available when they have to money to order one...
22:06.19*** join/#asterisk mlsmith99999 (~store@adsl-76-240-11-130.dsl.ltrkar.sbcglobal.net)
22:06.25citywokheh, yea. i just put all of it on my creditcard and send them a bill.  ripping out a frac t1 saves them $600/mo so they're excited it pays for itself in less than 12 months
22:07.56mlsmith99999afternoon, quick question, what VoIP providers do some of you recommend? Business, 2 Lines, US only, need unlimited.
22:08.03citywok~itsp
22:08.04infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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.
22:08.43mlsmith99999~itsplist-us
22:08.43infobotHere 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 , http://voip.ms and http://flowroute.com
22:08.58*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
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22:22.07*** part/#asterisk oneseventeen (~onesevent@web64.webfaction.com)
22:30.44*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
22:40.59*** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com)
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22:58.33citywokI love ripping out $700/mo voice T1's and installing asterisk. lol
22:58.46Qwelldo they ever let you rip the cable from the wall?
22:59.00citywokIn my own company I ripped the entire phone system off the wall
22:59.05citywokand the 2 racks of equipment
22:59.16citywokand ripped the sheets of plywood off the walls :P -- THAT was fun
22:59.37citywokbut I couldn't rip out the T1's, sadly that's all we can get for internet. lol
23:07.39ecraneAnyone have experience with the asterisk app 'Festival'? I configured it per http://www.voip-info.org/wiki/view/Asterisk+festival+installation but I am having a problem with no audio. I'm looking carefully at the asterisk and festival consoles and it looks like it should be working...
23:08.15ecraneLike the guy on this page who was forced to use an AGI script, I am having the 'no audio' problem while using a SIP channel. 'This seemed to work well however, it only works when called via the zaptel card. When I call via sip, I would see the Festival cmd execute on the cli, but I heard nothing, and the sip call hung up. '
23:31.09*** join/#asterisk bneff (~bneff@12.104.144.2)
23:31.30bneffanyone here familiar with configuring SIP + checkpoint firewalls?
23:40.48*** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net)
23:44.54lanningbneff: :
23:45.08lanningbneff: "don't cross the streams"
23:45.57lanningthe last time I dealt with checkpoint and sip, I ended up bypassing around checkpoint.
23:46.12lanningsupposedly it works, if you use Cisco Call Manager.
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23:49.38bnefflanning: what do you mean by dont cross the streams?
23:50.20drmessanoCrossing the streams would be very bad
23:50.53bneffI bet...it would probably make a mess
23:51.23drmessanoTry to imagine all life as you know it stopping instantaneously and every molecule in your body exploding at the speed of light.
23:51.35drmessanoTotal protonic reversal
23:51.38bneffsounds like nirvana
23:52.03DeHackEd... I think it's a reference to a movie
23:52.09drmessanoGhostbusters
23:52.12DeHackEd(I say 'think' sarcastically)
23:52.30bnefftakin' it way back with Ghostbusters
23:52.38bneffmust have been on Comedy Central or something recently ahah
23:52.49drmessanoNo, but it is the greatest movie ever
23:52.59drmessanoWe came, we say, we kicked it's ass
23:53.02drmessanosaw*
23:53.04drmessanoGAH
23:53.11frigidzephyri saw an awesome proton pack replica in person recently
23:53.13frigidzephyrwas beautiful
23:53.26bneffhave you seen zombieland?
23:53.39bneffgreat bill murray part in there
23:53.42drmessanozombieland sounds very new school
23:54.08bneffahhh woody harlleson and bill murray are in it..
23:54.14bneff(sp)
23:55.04citywokHow many people in here build/resell asterisk PBX's?
23:56.10*** join/#asterisk Mez (~mez@ubuntu/member/mez)
23:56.15Mezhates sipgate
23:56.20citywokor i suppose just install asterisk PBX's for businesses
23:56.38riscphreedoes, citywok
23:57.16Mezspent 3 hours trying to get it to work with the damned incoming calls... to no avail.
23:57.40citywoklol
23:58.13Mezcitywok: I hope that lol was at me
23:58.22Mez(or sipgate)
23:58.28citywokmez yes it was
23:58.38Mezcitywok: good...
23:58.49MezAny suggestions for good UK VoiP providers?
23:58.54Mez(that actually work!)
23:59.48citywok~itsp
23:59.49infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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.
23:59.58citywoksorry, no UK list apparently

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