IRC log for #asterisk on 20140217

00:14.52WIMPyAh, isn't it fun to talk with 16-24% packet loss?
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01:41.14eXcAliBuRanyone using a billing module?
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02:39.48eXcAliBuRanyone using a billing module?
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04:54.35Psil0Cybinhey guys I want to use SugarCE and Asterisks to make phonecalls from a voip line, what else do I need in order to make that happen? Am I missing a voIP server? The company that is giving me voip lines gave me all informations about servers,. etc?
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05:03.23Psil0Cybinwould I need to use asterisks and twinkle for example in order to make phone calls?
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06:01.37ChannelZPsil0Cybin: well if they gave you server credentials you should be able to configure your phone directly, you don't have to run asterisk unless you want to
06:06.33Psil0CybinOh because they kept asking what software we where using and I said asterisks, and they said that is just a server obviously so do they mean softphone? such as twinkle, etc?
06:07.19Psil0Cybinso far I found twinkle, ekiga, sflphone, zoiper
06:07.36Psil0CybinChannelZ, am I on the right track?
06:08.42[TK]D-FenderAsterisk is a PBX toolkit.  It is one possible part of the picture but it is not a "phone"
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06:13.33Psil0Cybinyea I know asterisks alone, is not a phone but in combination with software it is, I am trying to figure out what I would need or what guide I can read to understand how voIP really works
06:13.47Psil0Cybinlike I have questions, if I am running asterisks do I need a voip provider?
06:13.51Psil0Cybincan I be my own voip provider?
06:14.09fauxallianceffs
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06:14.45afinkhowdy [TK]D-Fender
06:15.12[TK]D-Fendery0
06:15.47[TK]D-FenderPsil0Cybin: What you're missing from the equation is SERVICE
06:16.01[TK]D-FenderPsil0Cybin: How do you reach the "real world phone network" or PSTN
06:16.26[TK]D-FenderPsil0Cybin: Can you be your own provider?  That's kinda circular.  Everyone gets service from someone else in a way.
06:16.44[TK]D-FenderPsil0Cybin: If you have a phone line you can get a hardware interface and use that
06:17.02[TK]D-FenderPsil0Cybin: If you want to use an ITSP you could use that as well...
06:17.05[TK]D-Fender~itsp
06:17.05infobot[~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.
06:17.07Psil0Cybin[TK]D-Fender, exactly, alright sorry I really do not want to bother you if these are silly questions maybe point me to a guide? I really want to understand, so at the moment If I understand correctly? I would get a Provider/Service -> than use asterisks in order to access the server -> then a softphone to access the asterisks server? I just want to understand.
06:17.25Psil0Cybinokay so lets say I go with a ITSP,
06:17.28[TK]D-Fender"than use asterisks in order to access the server" <- clarify this...
06:17.37Psil0CybinMy next step would be...just to use a softphone?
06:17.49Psil0Cybinor I am going to need asterisks to access the ITSP?
06:18.04Psil0Cybintrying to understand...
06:18.39[TK]D-FenderAsterisk is middleware.
06:18.46Psil0Cybinbtw [TK]D-Fender thank you for taking the time to explain this, I owe you a gift basket.
06:19.24[TK]D-Fenderas I said, Asterisk is a PBX Toolkit.  If you jsut want direct service you could connect a softphone, etc DIRECTLY to an ITSP account.
06:19.24Psil0Cybin[TK]D-Fender, Asterisks is middleware between the ITSP and Softphone?
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06:19.31[TK]D-FenderAsterisk lets you DO stuff on the otherhand
06:19.36[TK]D-FenderYes
06:20.01[TK]D-FenderDo you understand what a PBX is?
06:20.22Psil0CybinNo but googling it now to save you time and stress in thinking im a complete moron.
06:20.23Psil0Cybin1 second
06:21.59Psil0CybinAlright so maybe I am confused now, so would my ITSP have the IP PBX system?
06:22.09Psil0Cybinand I would juse use the softphone, hence why my boss is paying for the lines?
06:22.21ChannelZThe ITSP lets you make phone calls.
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06:22.54ChannelZWhether or not you want a PBX in between you and them is your choice.. to build a "telephone LAN"
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06:24.05ChannelZIf you want to create voice menus when people call you, or do your own voicemail, or conferencing, or anytrhing special like that, you could run asterisk.  If you just want to make calls, you can connect your softphone directly up to your ITSP.
06:24.38Psil0CybinChannelZ, Okay!! That is amazing, the second part of the statement just summed it up.
06:24.45[TK]D-Fender[01:22]Psil0Cybinand I would juse use the softphone, hence why my boss is paying for the lines? <- the ITSP PROVIDES the lines.  That is the service
06:24.50Psil0Cybinso I would just use a softphone opposed to the old phone that was being plugged into the ethernet
06:24.53Psil0Cybinis what you are saying.
06:24.55ChannelZSo you either setup and configure asterisk to talk to your ITSP, and your softphone talks to your asterisk to make calls through it, or you connect your softphone directly to the ITSP
06:24.58[TK]D-FenderPsil0Cybin: a PBX is a PHONE SYSTEM like companies normally have.
06:25.25[TK]D-FenderYou know... the thing that gives callers menus, lets companies have 100 phones on desks, but only 4 lines out, voicemail, etc
06:25.38Psil0CybinChannelZ, okay so I would probably want to connect my softphone directly to the ITSP if I just want 10 people calling out
06:25.44Psil0Cybinusing a computer instead of a phone unit
06:25.46Psil0Cybinas we had before.
06:26.03[TK]D-Fender[01:25]Psil0CybinChannelZ, okay so I would probably want to connect my softphone directly to the ITSP if I just want 10 people calling out <- no, that is where a PBX comes in
06:26.07Psil0Cybinand I could asterisks to create voice menus when people call the office? correct?
06:26.09ChannelZwell the softphone is just 1 thing.
06:26.23[TK]D-Fender[01:26]Psil0Cybinand I could asterisks to create voice menus when people call the office? correct? <- you create it with Asterisk.
06:26.29ChannelZIf you have 10 people in an office, you probably want to build a phone system.
06:26.35ChannelZ(IE asterisk)
06:27.01Psil0Cybin[TK]D-Fender, and ChannelZ thanks for coping with me, You guys are really helping me. I am taking everything you say and saving it so i can google terms in elaboration later when you guys are offline
06:27.29Psil0CybinChannelZ, but all the 10 people in the office are going to be doing is calling out, not really getting calls in.
06:27.40Psil0Cybinkind of just like collection calls type thing i dont know how to explain it better.
06:27.42ChannelZAnd keep in mind you would need to get multiple channels from your ITSP to make multiple simultaneous calls, most basic accounts usually frown (if not outright limit you) to 2
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06:28.32Psil0CybinWell ChannelZ the ITSP I had previously, set up phones that had ethernets plugged into them and you would just pick up and dial normally, is this what you mean? I was just planning on using those ethernet cords and plugging them into a computer instead of a phone.
06:29.11Psil0CybinWell the ITSP came by and set up the phones, but I am confused if perhaps the phones had some kind of software on them?
06:29.14ChannelZIf they were SIP phones you could keep using them probably.
06:29.37Psil0CybinWe could but we dont want to keep manually typing in the numbers.
06:29.46ChannelZPhone numbers?
06:30.06Psil0Cybincorrect, I was thinking there was a softphone I could use, to create a call list, and have it just dial automatically.
06:30.22Psil0Cybineven if they have to enter it, it is fine as long as they can do it via a  computer and headset to use notepad and email
06:30.25ChannelZMaybe, that part is your problem
06:30.54[TK]D-Fender[01:30]Psil0Cybincorrect, I was thinking there was a softphone I could use, to create a call list, and have it just dial automatically. <- Softhones are dumb.  If you want automated calling, that is what a central PBX does.
06:31.02ChannelZIn any event you can probably just use your softphones directly to an ITSP since your needs don't seem to be that picky.
06:31.58ChannelZand it's not like you have to decide instantly, if you start running into limitations in what you want to do, you can always setup asterisk later.
06:32.10Psil0CybinChannelZ, they are not picky at all, so do you think that is the easiest solution? I just want to use a program like perhaps even twinkle to call, so could I use twinkle and configure it to the ITSP settings?
06:32.14Psil0Cybinand I would be set? for basic features/
06:32.33Psil0Cybinplug in a headset, microphone type thing?
06:32.33ChannelZEasiest, yes, so long as you can find a suitable softphone you like that does what you want.
06:32.43Psil0CybinOkay :) That is what i was confused about
06:32.57Psil0Cybinso maybe I need to read up on asterisks, I had the wrong idea of what it was.
06:33.10Psil0Cybinbut what you guys explained is phenomenal.
06:33.44Psil0CybinWhen you call, call centers and they have robotic machines answer, is that an Asterisks PBX Toolkit?
06:33.49Psil0Cybinjust rying to understand
06:33.53ChannelZCould be
06:33.58Psil0Cybinor extensions to different offices?
06:34.05ChannelZOr something like it, but yes.
06:34.08Psil0Cybinwow
06:34.13Psil0CybinI am mind blown with what you can do.
06:34.14ChannelZa PBX
06:34.21Psil0CybinThank you ChannelZ.
06:34.33ChannelZasterisk is just a software PBX
06:34.42Psil0Cybinlast question ChannelZ [TK]D-Fender how did you guys learn about this stuff? In Detail? if I want to get your kind of knowledge :D
06:34.45Psil0Cybinare you guys in school for this?
06:34.55ChannelZNo just started screwing around with it
06:35.07ChannelZI started my own company and needed a phone system
06:35.24Psil0CybinReally...wow, I am 23 and want to sort of start my own company doing web design, CRM solutions, Linux based server stuff.
06:35.32[TK]D-FenderI work for a company and wanted to replace to Norstar we had.
06:35.43Psil0CybinWow you guys actually rock
06:35.54Psil0Cybinso how would I start learning? where do I start
06:35.59[TK]D-Fender~book
06:35.59infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
06:36.00Psil0Cybinmaybe this is very very....vauge.
06:36.06Psil0CybinAmazing!
06:36.08ChannelZ~primer
06:36.08infobotprimer is probably http://burner.com/asterisk-primer
06:36.10Psil0Cybinty!
06:36.20ChannelZThere's lots of resources out there
06:36.32Psil0CybinI know but as you see. I need to understand basics.
06:36.37[TK]D-Fender[01:32]Psil0CybinChannelZ, they are not picky at all, so do you think that is the easiest solution? I just want to use a program like perhaps even twinkle to call, so could I use twinkle and configure it to the ITSP settings? <- btw... this is the equivalent of using your computer as a phone direct to that service, ie NO added functionality, no sharing of a line, no automated dialing, nada
06:36.38Psil0Cybinor I just get confused and have conjumbled ideas.
06:36.49[TK]D-FenderRead the book.
06:36.55[TK]D-FenderRead up on basic PBX's
06:37.02Psil0Cybin[TK]D-Fender, Well they might be giving is 10+ lines, so that is what my silly boss is already paying for.
06:37.07Psil0Cybinhe just wants computers, to look fancy I guess...sigh
06:37.11Psil0Cybini dunno all i know is ill get paid so...
06:37.12ChannelZYeah as soon as you start picking up a little of the terminology it will click
06:37.18[TK]D-FenderSoftphones SUCK
06:37.41[TK]D-FenderBetter to use actual dedicated phones of some kind
06:37.49Psil0CybinYea maybe it is the terminology that completely throws me off :D aha, all I know is 2 years ago I had no idea what linux is and today all I use is linux and love learning so
06:37.52Psil0CybinI am going to keep digging
06:37.55Psil0Cybinand chilling here with you guys :D
06:40.39Psil0CybinGetting that ebook as we speak, thank you so much I am going to be reading a lot tonight, I was going to make this much more complicated than I needed off the bat, like I am going to start by using a simple softphone and as I learn more about Asterisks build some thing fancy!
06:42.10ChannelZHave fun
06:44.04Psil0CybinI always do :D
06:44.33Psil0Cybinthanks again team for pointing me in the right direction.
06:44.40Psil0Cybinsaved me hours of googling wrong terms, etc
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07:15.08Psil0CybinChannelZ, I could use asterisks in order to connect people calliing in for example to available support?
07:15.18Psil0Cybinvia a softphone?
07:15.20Psil0Cybinonce I learn more
07:15.37Psil0Cybinwatching the introductory video.
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08:42.29sephiroth1234Hello
08:42.39sephiroth1234i have a question about asterisk and NAT
08:42.44sephiroth1234can anyone help me?
08:46.48bulkorok~ask
08:46.48infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
08:48.30sephiroth1234Is there any module or patch for Asterisk to detect the following scenario:
08:49.06sephiroth1234I have an Asterisk box with a public IP address and two SIP  clients behind the same NAT device; I also have SIP clients behind  different NATs. I want to know is it possible for Asterisk to detect if  both clients are behind the same NAT and use direct media between them  and use other options for clients that are behind different NATs?
08:49.32sephiroth1234My asterisk box has a public IP address
08:50.15sephiroth1234And the signaling protocol is SIP
08:52.29sephiroth1234Any idea??
09:06.57sephiroth1234?????
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09:31.33sephiroth1234No answer?
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09:48.32sephiroth1234Why no one answered me?
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10:45.20illuderanyone tried this script to show when asterisk dies?
10:45.21illuderhttp://bbs.openvox.cn/viewthread.php?tid=754
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11:08.55WIMPyAsterisk usually doesn't crash. It rather locks up.
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13:16.31illuderany facility to detect when Asterisk locks up, by sending an email or other notices?
13:17.34WIMPythat's a lot harder to detect.
13:20.12ChainsawYou'd have to originate a test call on the CLI from a cron job.
13:20.32ChainsawAnd then check return codes.
13:20.53Chainsaw(Or you could try a TCP connection to port 5060)
13:22.25[TK]D-FenderAMI <-
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13:30.40BeachBallMorning everyone
13:31.18ChainsawGood morning. That reminds me of Macs somehow.
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13:39.04illuderi think maybe something like cacti would detect lack of traffic (within x minutes) during office hours and report the pbx down?
13:42.26Chainsawilluder: Bit of a delay though. My usual "office full of people" detection net is quicker then that.
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13:48.12illuderyeah, i know... but in this case, the server went bang at 2am, and the staff works 24 hours... they (manager on duty) didnt know who to call to fix it... we had to hear shouting from the owner at 8:00 am..asking if we dont have monitors in place to see if the asterisk locked up..
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13:50.14Chainsawilluder: I have a cron job that places an internal call, quicker then the handsets can answer it. If you get anything other then "NOANSWER" back, you have a connectivity problem. And if the whole asterisk command won't run, probably a daemon lockup.
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13:50.51Chainsawilluder: But I'm going to double up soon anyway. The Polycom phones have a concept of backup SIP servers that sounds rather useful.
13:53.04BeachBallanyone doing billing with asterisk?
13:53.47[TK]D-FenderNo.  Asterisk doesn't do billing.
13:54.03ChainsawIt does CDR, which you can plug into billing packages though.
13:54.03[TK]D-FenderAsterisk generates CDR .... but billing from that is up to you
13:54.20BeachBallthats not the question
13:54.40[TK]D-FenderIt's the answer to the one you asked.  Ask better questions...
13:55.08BeachBallis anyone using a billing module with their asterisk?
13:55.41[TK]D-Fendermany people.
13:55.50BeachBallit is too early to be pissing me off
13:56.25[TK]D-Fender"a2billing" is popular for those running calling card / flow-through ITSP type installs
13:56.41[TK]D-FenderThough it requires a lot of concessions as to how it's configured
13:57.48*** join/#asterisk davlefouAMD (~david@41.225.234.62)
13:57.50[TK]D-FenderBeachBall: https://www.google.ca/#q=billing+solutions+for+Asterisk <----- took fewer words than your original question and gives direct links.  Always Google.  Google is your friend.
13:57.50Chainsawilluder: But it sounds like the main problem you have is that there is no panic button/pager.
13:58.05BeachBalli googled plenty
13:58.11BeachBalland didn't like anything i saw
13:59.19[TK]D-Fender4 direct link, one to a wiki page with another 10.  All as of the first page of results.
13:59.27[TK]D-FenderSo you looked at all 14 or so of those?
14:03.32*** join/#asterisk yago (~kresp0@gateway/tor-sasl/kresp0)
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14:04.30BeachBallthis is gonna be a bad day... i'm gonna watch a movie and pretend to work
14:05.55[TK]D-Fenderdrmessano: Do you think they'll be hiring a replacement for his position immediately?
14:06.13*** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net)
14:06.40*** join/#asterisk Katty (~Katty@97-91-121-70.static.stls.mo.charter.com)
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14:16.47snadgejust write your own billing system using php.. or find an open source one and mod it :P
14:26.11[TK]D-Fendersnadge: he left.....
14:28.51snadgewhat a shame.. i had such hopes for them
14:30.43*** join/#asterisk BeachBall (~eXcAliBuR@206.162.174.6)
14:30.49BeachBallis sorry [TK]D-Fender
14:31.13BeachBalltooks his pills and feels better now
14:31.32WIMPyIs there any way to influence what dundi uses as ${IPADDR}?
14:32.50Faustovwe're hard-coding it WIMPy so I guess not
14:33.15WIMPyBad
14:33.29Faustovwhy?
14:33.44WIMPyAnd even more interestingly. For some reason I don't have a lo, only an eth0, but still it uses 127.0.0.1.
14:34.06Faustovhmm, that could be a bug
14:34.38FaustovI've got enough interfaces I had to hard code it on all systems in my dundi network
14:35.32WIMPyI guess using some dyndns service is the only answer then.
14:36.02*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
14:36.28BeachBallhugs [TK]D-Fender, do you still luv me?
14:46.29*** join/#asterisk sephiroth1234 (~Gholamrez@2.146.90.237)
14:47.45sephiroth1234Hello
14:48.04sephiroth1234can anyone help me in finding the answer to this question
14:48.06sephiroth1234I have an Asterisk box with a public IP address and two SIP  clients behind the same NAT device(I also have SIP clients behind  different NATs). I want to know is it possible for Asterisk to detect if  both clients are behind the same NAT and use direct media between them  and use other options for clients that are behind different NATs?By  detection I mean is it possible for Asterisk to take a look at the  public IP address of packets and if both
14:48.06sephiroth1234<PROTECTED>
14:50.58sephiroth1234??
14:52.36wonderworldYou could could allow and deny directmedia for your clients directly in sip.conf check the directmediapermit and directmediadeny settings...
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14:53.35sephiroth1234I want this to happen dynamically
14:53.40sephiroth1234not per client
14:53.53sephiroth1234I have too many clients that register automatically
14:54.13sephiroth1234using options for individual clients is not an option for me
14:54.57Kattymorning
14:55.46sephiroth1234is it possible?
14:56.02[TK]D-Fendersephiroth1234: No.
14:56.10*** join/#asterisk lnb (~lnb@CPE000347b24a71-CM602ad06bec2f.cpe.net.cable.rogers.com)
14:56.16lnbmorning everyone
14:56.43sephiroth1234Someone told me that he could write something for this (using an Asterisk module)
14:56.52lnbi cant believe it, just sent a 39 page fax but it goes via our freeswitch server
14:57.47[TK]D-Fendersephiroth1234: It's be a serious patch to the asterisk SIP channel driver.
14:57.48lnbstarted 9:29:02, finished 9:55.04
14:57.56[TK]D-Fendersephiroth1234: Not some miracle extra module.
14:58.44sephiroth1234So it is not something trivival to implement
14:59.35sephiroth1234why Asterisk does not have such a functionality? Does it break anything?
14:59.54[TK]D-FenderIt doesn't have it because it doesn't have it.
15:00.02Kattysticks face into mt dew
15:00.18[TK]D-FenderCould it?  Possible I suppose.  Complicated?  Dunno... issue a request on the tracker for it.
15:00.38bulkoroklnb: freeswitch ftw!
15:00.56sephiroth1234o.k
15:01.04sephiroth1234thx alot
15:01.07Katty[TK]D-Fender: how are you functional
15:01.17lnbbulkorok:  sorry thought it was fs #
15:01.26bulkorokno prblem :-D
15:01.31[TK]D-FenderKatty: Highly :)
15:01.45Katty[TK]D-Fender: what?
15:01.52Katty[TK]D-Fender: your grammar doesn't even parse.
15:01.55*** part/#asterisk bulkorok (~Benjamin@85.183.61.47)
15:02.04Kattyor maybe i need more caffeine
15:02.12[TK]D-FenderKatty: it does.. you are clearly insufficiently caffeinated :)
15:02.18Kattypets the mt dew.
15:03.02Kattyi wanted to stay home and play mario 3d world this morning :< pesky job and bills.
15:03.35wonderworldKatty: on 3DS?
15:03.44Kattyno on wii u, but i think it's the same game
15:04.21wonderworldit was great on 3DS. really fresh. nintendo surprises me over and over again.
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15:16.21jaflongI have a problem with chanspy. I am calling it from within a originate. When I start listening in I can hear the spy-sip.gsm audio play only however there is no audio thereafter?
15:20.21*** join/#asterisk saint_ (~saint@c-50-166-85-78.hsd1.nj.comcast.net)
15:23.44Kattywonderworld: yeah it's not what i'd consider my usual type of game.
15:24.04Kattywonderworld: but it's fun, and that's all that matters (=
15:25.44*** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl)
15:25.59[TK]D-Fenderjaflong: Is that... a question?
15:26.10[TK]D-Fenderjaflong: Are you not sure if you have audio or not?
15:27.20*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
15:27.24jaflongthats the problem. I have audio only when the plays the sound file but no audio when talking
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15:30.10[TK]D-Fenderjaflong: You'll have to tell and show a lot more for us to be able to assist you....
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15:33.37jaflonghttp://pastebin.com/M5Stfw8A please have a look
15:35.04pabelanger<PROTECTED>
15:35.16pabelangerYou are hanging up for channel
15:36.45jaflongthats When I put the handset down
15:38.25[TK]D-Fender<PROTECTED>
15:40.31jaflongI was looking for the way to include the options paramaters (E)
15:41.19jaflongit still luanched chanspy -  Launching chanspy
15:41.37jaflongwhats the the correct way?
15:42.30[TK]D-Fender"core show application Originate"\
15:43.36Kattydebates.
15:44.08KattyNugget: telnet
15:44.08Nuggettelnet is eeeeeeevil!
15:44.24Nuggeteyes Katty
15:45.11Katty:>
15:45.14Kattyhugs Nugget
15:49.38saint_in a SIP trace (wireshark), what is the difference whhen I send DTMF between a payload type: DynamicRTP-type-101 , and type 97 ?  The 101 shows a RFC 2833 RTP Event , but the payload 97 does not show it. Is this the diff. between InBand and OutOfBand ?
15:50.19BeachBalli'm trying to install a2billing but i get connection failed, that leads me to a db problem... can someone help me with this?
15:50.46*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
15:52.11[TK]D-FenderBeachBall: These systems are almost universally despised here (because of the businesses that use them, thus very little support here.They have their own support resources you should flagging down in the meantime...
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15:56.54BeachBalli fixed my db problem, and it's running now
15:56.55BeachBall;D
15:57.26BeachBallwants a hug too :{
15:57.46BeachBallhugs Nugget hoping for a hug back
16:10.00*** join/#asterisk Farkie (~Farkie@213.229.88.2)
16:10.19FarkieHey, can anyone help me diagnose what I think is a NAT problem?
16:11.03FarkieWe have 2 lines, 1 BT hardline, and 1 SIP line (from Gradwell). First call comes in on the BT, rest come in from SIP.
16:11.28FarkieFirst call comes in fine - both sides can hear. If a second call comes in at the same time, we get 1 way audio. They can hear us, but we can't hear them
16:12.00*** join/#asterisk navaismo (~navaismo@200-52-45-221.dynamic.axtel.net)
16:12.35FarkieVersion: Asterisk 1.6.1.18 - don't think it'd be that though
16:14.13[TK]D-Fenderthat version is very old, and the branch no longer supported
16:14.15*** join/#asterisk g_r_eek (~g_r_eek@78-89-187.adsl.cyta.gr)
16:14.43[TK]D-FenderAs for 1-way audio, NAT is often the prime issue.  We'd have to check your SIP DEBUG to validate if things look configured right.
16:14.45[TK]D-Fender~pb
16:14.45infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
16:14.47[TK]D-Fender^^^^
16:14.49[TK]D-FenderShow us the calls
16:15.25jaflongany feedback on my chanspy issue
16:15.59Farkie[TK]D-Fender, thanks for the reply - I'll have to wait till it happens again to get some feedback
16:16.26*** join/#asterisk devil_evoxxx (~root@webs3.as48500.net)
16:16.34devil_evoxxxhello guys
16:21.11[TK]D-Fender[10:42][TK]D-Fender"core show application Originate" <----------
16:21.15[TK]D-FenderFarkie: ^
16:21.23[TK]D-Fenderjaflong: rather
16:21.41devil_evoxxxis there a proper way to test a subscribecontext?
16:21.49Farkieah I was wondering why I was getting command failed
16:21.49Farkielol
16:22.04devil_evoxxxi'm going to be crazy with BLF, seems not working, but a custom hints on devstate work fine
16:24.22[TK]D-Fenderdevil_evoxxx: Proper way to test : use it.  Look at the subscription
16:25.22devil_evoxxx[TK]D-Fender: subscription are ok, notify are sent to phone (looked on a sip capture)
16:25.36devil_evoxxxbut, the "BLF" on phone does not change status
16:26.28[TK]D-FenderSounds like a phone issue then...
16:26.41devil_evoxxxanother thing i've noticed is that on core verbose i can't see "Extension changed 421[mycontext] new State
16:26.52*** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com)
16:27.30devil_evoxxxi'm using the last 1.8 available
16:27.42*** join/#asterisk [NC] (~nc@173.245.31.71)
16:28.51Farkie[TK]D-Fender, Not too sure if I can upgrade Asterisk tbh
16:29.10Farkiecat /etc/redhat-release
16:29.10FarkieFedora release 12 (Constantine)
16:29.18Farkieoops pasted an extra line
16:30.38[TK]D-FenderI don't see where that version is not compatible with supported releases...
16:31.09Farkieto be honest I wanna flatten the box
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16:47.43Blackslikzhey guys anyone in here have a shell script that can reload asterisk automatically after every 3 mins??
16:48.20[TK]D-FenderThat's like 3 lines of bash....
16:48.31[TK]D-FenderAnd why on earth would you be wanting to do that?
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16:51.16Blackslikzbecos of what i run on asterisk
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17:02.33[TK]D-FenderBlackslikz: Sounds pretty messy
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17:26.33illuderi wonder if freepbx has been converted to multi tenancy before?
17:26.51tm1000illuder: that would be a crap ton of work so I seriously doubt it
17:28.10mjordanif you find yourself reloading Asterisk every 3 minutes, you're probably doing something wrong.
17:29.31[TK]D-Fenderilluder: Indded crap-ton, and no.
17:30.01[TK]D-Fendermjordan: I'm guess he's trying to protext against people not applying changes, etc.
17:30.15[TK]D-Fenderprotect*
17:30.39*** join/#asterisk hardwire (~hardwire@222-83-237-24.gci.net)
17:30.59illuderAny suggestions on open source multi tenant foundations i can start working on if fpbx wont cut it?
17:31.31illuderi did google, didnt come up with much options on multi tenant open source systems.
17:32.21[TK]D-FenderI've never seen a free one
17:33.21*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
17:34.16pabelangerwe're trying to build a multi-tenant queue system on ARI
17:34.24pabelangerHave most of the ground work in place for it
17:34.36illudermaybe setup a VM server and load fpbx in each container? maybe have like 20 vm's/fpbx instances in one server, i think that will work...
17:34.54pabelangerWe're leveraging OpenStack keystone to handle all the tenant stuff.  Since it does it out of the box, we just get it all
17:35.53illuderis openstack a vm program?
17:36.44pabelangeropenstack is a collection of APIs, we are just using their identity service
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17:39.51illuderjust cehcked out the openstack site...sounds interesting, but i dont think it will allocate good resources for VoIP?
17:42.31pabelangerRight, that all has to be built out.  Right now, we have provisioning of sip subscribers (kamailio) and queue / queue member (asterisk), which is build on a multi-tenant core.  We still need to do trick things in the asterisk dialplan, but we are working to replace it all using ARI
17:43.47illuderwhy are yo uusing Kamailio?
17:44.07pabelangerredundancy / scalability
17:44.18pabelangerAsterisk is just a bridge / media gateway is our setup
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17:45.41illuderhmm, i though Asterisk v11+ would have resolved these scalibility fears...
17:46.16pabelangerwhat makes you think that?
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17:47.39illuderfrom what i read of the new updates etc....
17:48.03illuderi may have misread it...
17:48.10illuderhence why i asked...
17:48.16Penguinhence why
17:48.39pabelangerwell, some aspects of asterisk are distributed, however some things are not.
17:49.15pabelangerIMO, the best approach is still to move your SIP subscriber into a proxy vs having asterisk trying to replication that information out
17:49.47pabelangerBut there is a lot of ways to do it, so the setup you need depends on what you are trying to do
17:49.55illuderand of course you're running the Asterisk in Realtime?
17:50.14pabelangerno
17:50.31pabelangerfor the majority of the setup, we provision asterisk using config files with puppet
17:51.15illuderoh :O   how will you use multiple instances of IVR (multi tenant) if there's no DB?
17:52.15pabelangerwell, there is a database backend, but asterisk doesn't connect to it using realtime. We have a series of AGI / AMI applications, and more and more trying to get ARI going
17:55.08illudergot it..
17:55.30illuderand all from scratch, yeah? not utilizing any opensource code?
17:56.45pabelangerAll we use it opensource code
17:56.51pabelangers/it/is
17:57.07pabelangerand everything is up on github
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18:07.48illuderplease excuse my ignorance, but what is s/it/is  ? Google doesnt come up with much on it
18:09.34pabelangerilluder, #kickstand, so we don't go off topic here
18:12.59illuderok, meet you there
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18:34.39Penguins/why //
18:35.02PenguinI guess there's a time limit on the replacement.
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18:45.41*** join/#asterisk Roland- (~win@unaffiliated/roland-)
18:46.05Roland-hello, is there a way to export the logs somehow in a more human readable way like statistics?
18:46.45navaismoRoland-, which log?
18:47.14Roland-master.csv one
18:48.34navaismothats the CDR basically, hmm you can use a DB to export it
18:49.14Roland-ok, is there a simple way for that ?
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18:49.53navaismoif i recall in voip-info exist an script to export it take a look on that site
18:50.04tm1000pabelanger: oooo you have a # channel
18:50.08tm1000<PROTECTED>
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18:56.02Kattyin the polycom macfoo-directory.xml file, does the ringtype <rt>3</rt> refer to the ringtone played?
18:57.49Kattyand what is meant by a Watched Phone in reference to bw (buddywatching) and bb (buddyblock)
19:00.31Kattyis this for presence stuffs?
19:03.49carrarYou're going to need to join #polycom
19:03.51carrar:)
19:04.05carraryes & yes
19:05.11carrarbw 1, bb 0
19:05.18carrarto subscribe to hints
19:05.34carrarI use rt 2
19:06.34carrarI think 1 is silent?
19:06.37carrarbeen a w hile
19:07.01Kattyoooh, a new channel to pester!?
19:07.12Kattyaww :< apparently not
19:07.14carrarhaha
19:07.24KattyLIES AND DECEIT
19:07.28Kattyhugs carrar
19:07.38carrarhuhs KattyRoo back
19:07.48Kattyhow're you dear?
19:07.59carrarI am GREAT!
19:08.03carrarthinking I should go for a walk
19:08.24carrarand find a way to slack off for the rest of the day somehow
19:08.38carrarSince after all it's a holiday
19:08.50carrarAs there have been some good Presidents in the PAST
19:08.51Kattyi agree.
19:09.01Kattyi hear the weather is going to be very suitable for walks soon :>
19:10.32carrarNice weather in Maui today
19:10.57carrartoo bad I'm not there
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19:13.26Kattyyesh :<
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19:24.19Kattyblames canada
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19:36.59Roland-I found some tools but none work as espected I need some sort of statistics on each extension
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19:40.07jeffspeffI'm having an issue with a phone registering. Here's the SIP debug logs. The phone in question is coming from IP 172.13.221.190 at extension 1225. And the traffic is coming from over the internet. http://pastebin.com/L42JG36H
19:45.27Kattywhat extension do you folkses use for Page()?
19:45.34Kattyis there something common-ish?
19:46.42jeffspeffKatty, I make star extensions (*10, *11, etc.) for each of our page groups
19:46.57Kattynods
19:47.10Kattyanyone else?
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19:53.09jeffspeffanybody with ideas on my debug logs?
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19:59.13paulcI like **PAGE
19:59.44navaismoRoland-, what kind of statistics?
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19:59.53paulcor in our office we have 72430 (page all) and 72431 (for a group) but no one really knows this I don't think..
20:00.58*** part/#asterisk bulkorok (~Benjamin@gw1.pinguin.ag)
20:02.05Roland-navaismo basic call statistics, I have a telesales call center and user want to see how much an agent is calling
20:02.14Roland-some basic stuff, that is available in master.csv
20:02.40navaismouse asternic or queuemetrics
20:03.33navaismoa call center must use queues and the queue log provide good information about that. You can use asternic for free, i recommend queuemetrics for a very good reporting tool
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20:05.11Roland-it's some cheap callcenter
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20:16.13Roland-the problem is this callcenter only does outbound
20:17.26Roland-I only need to get reports from master.csv cdr
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20:58.22navaismohttp://www.cdr-stats.org/
21:01.06newtonrneat
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21:06.06Roland-indeed
21:06.19Roland-lots of requirements
21:06.25Roland-I am doing it now.
21:20.47jeffspeffI'm having an issue with a phone registering. Here's the SIP debug logs. The phone in question is coming from IP 172.13.221.190 at extension 1225. And the traffic is coming from over the internet. http://pastebin.com/b4Ujy5qz
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21:43.26MarkS-Hello, is it possible to see what phone/SIP client answered a call after the call has ended? I want to run a script that sends an email after the call has ended with that information
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21:54.13newtonrMarkS-, Nothing that I can think of within Asterisk.  You'd have to do something outside of Asterisk, from a SIP proxy in front of Asterisk, or by parsing Asterisk logs (with a SIP trace running) to grab the user-agent of the BYE
21:56.44newtonrMarkS-, or write an application for Asterisk that allows you to pull headers from SIP messages on a channel.  SIP_HEADER only lets you grab stuff from an inbound INVITE
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22:24.31MarkS-newtonr: thanks, in that case I probably let the script get it from the CDR
22:24.41MarkS-as it is there available (and we use MySQL for it)
22:24.44newtonrthere you go
22:24.46MarkS-I think that might work
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23:24.52Roland-ok
23:24.59Roland-I have unanswered = no in cdr.conf
23:25.07Roland-however no answer is still logged
23:26.52[TK]D-FenderShow us an actual full call and the CDR associated with it.
23:26.54[TK]D-Fender~pb
23:26.54infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
23:28.04Roland-http://bin.cakephp.org/view/1921616141
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23:29.40[TK]D-FenderAnd the full matching log...
23:29.56[TK]D-Fenderand config...
23:30.02Roland-I sorry how do I provide it?
23:30.06Roland-wathing log?
23:31.44Roland-http://bin.cakephp.org/view/1028388493
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23:39.10[TK]D-FenderASTERISK CLI <_
23:41.43Roland-http://bin.cakephp.org/view/2038378605
23:42.42[TK]D-FenderRoland-: Perhaps I didn't clarify that the best way for you.  Show an actual full call trace and it's matching CDR
23:43.21Roland-[TK]D-Fender I don't know how
23:43.30[TK]D-FenderAsterisak CLI <-------------
23:43.43[TK]D-FenderGo to CLI. PLACE A CALL
23:43.51Roland-aaa
23:43.51Roland-yes
23:44.40Roland-well I am calling however nothing in cli and I am logged in
23:45.30[TK]D-Fender"core set verbose 10"
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23:46.09Roland-great
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23:50.08Roland-http://bin.cakephp.org/view/1165801498
23:50.17Roland-what is the default verbose level to set back ?
23:51.09Roland-again, call is logged.
23:51.39[TK]D-FenderWhat version are you running?
23:52.11Roland-seems like 1.8
23:52.22wacomcitocan i allow ani cid fron sip trunk?
23:52.49wacomcitowithout alloguest
23:52.54[TK]D-FenderRoland-: aim for better than "seems like"
23:53.21Roland-Asterisk 1.8.20.0
23:53.28[TK]D-Fenderwacomcito: allowguest has nothing to do with that.
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23:53.59[TK]D-FenderRoland-: You are 5 releases behind.  update to current and attempt to replicate the problem.  If it continues then post it to the tracker
23:54.21Roland-oh well this is some centos binary stuff
23:54.26Roland-I will update soon
23:54.42Roland-I kind of need the mysql logging facility
23:55.16wacomcitoi use a sip trunk with user and pass.. and it runs ok but the CID is alwais the same... the authenticated user...
23:55.41wacomcitoi want receibe caller CID and process it.
23:57.47[TK]D-Fenderwacomcito: So your peer and a complete inbound call with SIP DEBUG enabled.the call
23:57.49[TK]D-Fender~pb
23:57.49infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
23:57.50[TK]D-Fender^^^^
23:57.58[TK]D-Fenderwacomcito: Mask ONLY the secret

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