IRC log for #asterisk on 20130705

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00:46.49hackishEvery time my internet goes down or my asterisk server is down my iax2 won't auto register with the provider.
00:47.29hackishiax2 show registry  Give me: (null)                N       981999999  <Unregistered>             60  Unregistered
00:47.49hackishobviously I changed the real phone number... Google isn't giving any hints. Ideas?
00:48.52hackishif I try to call me I get the dreaded call can't be completed as dialled.
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01:14.43cneb3000Hi everyone.. hate to be the guy who comes in the room and barks out for some advice but could do with some help!
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01:15.14cneb3000I have an issue involving media not being correctly established on a call. here's a flow diagram of the call - http://imgur.com/0Nh6JyO
01:15.30cneb3000as you can see, when we are asked to make the call, we immediately respond with a 100 trying, as normal
01:15.47cneb3000but then we immediately, send a 200OK, at which point the person calling acknowledges with an ack
01:15.50cneb3000and media is sent
01:16.24cneb3000in the mean time, our carrier has informed us that the call is now rining. and when it is eventually answered (183 in progress) the media is all over the place. and eventually the call just gets cancelled
01:16.52cneb3000we have another caller using the same trunk, same details etc. with no issues. and both clients have the same config
01:17.14cneb3000I didn't know whether there was something in this clients invite, that was provoking us effectively telling them to send media early
01:17.40cneb3000..any advice would be appreciated, cause im stumped if theres some config in asterisk to help get around this!
02:02.48cneb3000bah.. thanks anyway everyone :(
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02:37.24wrouesnelhas anyone encountered an issue where a Sipura SPA-3000 or similar will drop call audio if it tries to reregister with Asterisk when the call is picked up?
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04:05.01ChannelZcneb3000: would rather see a SIP debug of the call with verbose console as well
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04:39.17wrouesnelok is there a way to filter manager logons out of the cli and verbose logging? I don't need to see "manager logon/logoff" every few seconds
04:39.29wrouesnelsetting displayconnects=no doesn't seem to work
04:39.36sawgoodwrouesnel: good question and point ...
04:39.54sawgoodwrouesnel: you'll see that is more out of scope (could be) FPBX related ...
04:40.22wrouesnelit's coming from a small web app I use to monitor phone connections
04:40.53sawgoodOh for me, I have a different experience (sorry)
04:41.12wrouesnelif i drop the verbosity way down (to 1) it's fine, but defeats the point of being able to log my dialplans progress
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05:01.51kaldemarwrouesnel: the setting won't take effect on a "manager reload". have you restarted asterisk?
05:05.49enyawixis ignorecryptolifetime=yes supported in Asterisk 11? was a patch to get option in 1.8
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05:22.26wrouesnelkaldemar: not yet. it's a work day so I'll try it this evening i guess.
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06:32.17bulkorokhi :-)
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07:00.15AndyN_Hi guys, is it ok if I ask questions here?
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07:05.32bulkorok~ask
07:05.32infobotQuestions 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.
07:06.53AndyN_I'm having issue with Snom phone cannot pick up another extension call after upgrading from 1.2 to 1.8.
07:07.09AndyN_My set up: Asterisk 1.8.22.0, Snom 360 firmware v5
07:08.13AndyN_I can see the BLF light blinking, but pushing it show 'connected' but the other phone will just keep ringing and basically nothing happens
07:09.02AndyN_extra info please let me know
07:10.25kaldemaryou should tell how you have implemented pickup, what the button on the phone is supposed to do and show an asterisk CLI output of all that with verbosity and sip debug enabled.
07:12.50AndyN_roger
07:13.04AndyN_In asterisk sip.conf I used this
07:13.06AndyN_notifyringing=yes notifyhold=yes allowsubscribe=yes callcounter=yes counteronpeer=yes call-limit=99
07:13.26AndyN_then use Hint() in extensions.conf
07:14.06AndyN_on Snom 360 I set up Function keys to monitor the other exts. This works perfect in asterik 1.2
07:15.21AndyN_By 'monitor' I meant if a call comes direct onto ext 100, a person sitting at his desk with ext 101 could push the Function key and 'steal' that call from 100
07:16.38kaldemarcounteronpeer is not a valid option for anything in sip.conf (unless your asterisk has been patched), what's that about?
07:17.14kaldemarby "monitor" you really mean a subscription from the phone to asterisk, if we speak in SIP terms. :)
07:19.40AndyN_yes kaldemar, that's subscription, ple pardon my ignorance
07:20.51AndyN_I picked up counteronpeer in an online guide, should I remove it?
07:21.21kaldemarfeel free to remove it.
07:21.44AndyN_yes will do
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07:24.12AndyN_kaldemar anything else I could double-check?
07:24.36kaldemarthe CLI output with verbosity and sip debug.
07:24.40kaldemar~pb
07:24.40infobotA "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.
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07:31.14AndyN_Thank you. Got it - http://pastebin.com/b7sdPEX7
07:31.41AndyN_notice how it shows SIP/2.0 401 Unauthorized. What do you think?
07:35.22kaldemar401 is a normal part of authentication
07:36.34kaldemari'd be more concerned about the 404 on the subscription.
07:37.03kaldemar"Looking for 789 in internal-extensions (domain )" => "404 Not Found"
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07:39.05AndyN_ok. 789 is no longer used
07:41.37kaldemarthat's what something is still subscribing for. is it the phone that's at 192.168.3.30? there's no "User-Agent" header in the request.
07:42.42kaldemarif the phone is at 192.168.3.65, your pastebin contains no useful information. just a single qualify (keep-alive) with OPTIONS.
07:43.50AndyN_kaldemar, Thank you. I gotta leave now but will jump back onto this later. Really appreciate your help
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10:24.11cneb3000Hey everyone. this sounds stupid - but never had to post on an asterisk forum for help. Either get it from here on the book, but both have let me down ;)
10:24.18cneb3000what is the go to forum for asterisk to ask tech questions?
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10:36.30wdoekes~ask
10:36.31infobotQuestions 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.
10:38.13cneb3000sure I get that, I asked last night and I don't think anyone could help :) but thanks, I'm just goign to use the digium forums
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11:16.37hrolfHey, I'm getting a tick (sort of) sound in the start of each file, whenever I Playback.
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11:17.12hrolfWhen I listen to the same files in a player, they are fine. But in Asterisk I hear this sound before a file starts playing back.
11:17.18hrolfHow do I debug this?
11:20.55cneb3000hrolf: when you play them back in a media player
11:21.02cneb3000are they EXACTLY the same files that are in asterisk?
11:21.18cneb3000for example, most people will get a file as an mp3. and when they put it in asterisk,. covert it gsm.. g711.. wav etc.
11:21.51cneb3000because when you convert to gsm etc. it for the most past is converting it to lower quality audio. and this might be bringing out problems in the audio that you may have otherwise not seen
11:22.33cneb3000for example, if you use an audio editor to cut a song in half. It actually will leave a little 'spike' on the end/beginning. but you wont be able to hear this, because the spike is represented with all of its frequencies, and sort of blends in
11:22.45cneb3000but if you compress the same file, it removes a bunch of the frequencies, and the spike is more audible
11:22.53cneb3000it might be something else though :)
11:22.57hrolfcneb3000: Yes, they are the same files. .wav files to be specific.
11:23.14hrolfcneb3000: Bit-rate: 128kbps
11:23.35hrolfcneb3000: Sort of a like click sound
11:24.05cneb3000would you mind sharing the file?
11:24.11hrolfYes
11:24.16hrolfcneb3000: Let me do it.
11:24.17WIMPyMetadata?
11:24.44hrolfcneb3000: Where do I upload it? (If it matters.)
11:25.29cneb3000don't matter lol try http://vocaroo.com/?upload ?
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11:26.53hrolfcneb3000: http://vocaroo.com/i/s1t95LV6A0lW
11:27.27cneb3000and u hear the click at the beginning yes?
11:27.31hrolfcneb3000: Yes.
11:27.40cneb3000ok let me have a look. i cant promise anything :)
11:27.50hrolfcneb3000: Each time a file is played. It is same with all my files.
11:28.04hrolfcneb3000: All are .wav, 128kbps.
11:28.10hrolfcneb3000: Yeah okay, no problem.
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11:30.00hrolfcneb3000: http://stackoverflow.com/questions/5326135/artificat-on-playing-a-audio-file-on-a-phone-from-a-asterisk-server
11:30.18cneb3000yea, thats the same thing i'm talking about
11:30.24hrolfcneb3000: I think it is the same problem as above, but I don't understand what it is saying, anyone clarify?
11:30.35cneb3000sure
11:31.03cneb3000basically, like I said, you may have a small bit of audio at the very beginning of the file. and whatever is playing the audio, is aplifying it. and you get a click
11:31.17cneb3000im just going to upload the file again for you. try it and see if its any better
11:31.25hrolfcneb3000: Okay. Thanks.
11:31.46cneb3000hrolf: http://vocaroo.com/i/s1i5zF43rGpM try that :)
11:31.56cneb3000i've added a very quick 'fade in' at the beginning
11:32.04cneb3000if it doesnt work then I'm not sure what the problem is sorry :(
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11:34.35hrolfcneb3000: Dropping incompatible voice frame on SIP/999-0000001e of format ulaw since our native format has changed to 0x2 (gsm)
11:34.50hrolfcneb3000: WARNING[2396]: format_wav.c:103 check_header_fmt: Unexpected frequency 22050
11:34.53cneb3000oh sh** i didnt convert it to a proper asterisk friendly format
11:35.17cneb3000ttry http://my.digium.com/en/products/ivr/audio-converter/
11:35.24hrolfhttp://my.digium.com/en/products/ivr/audio-converter/
11:35.58hrolfcneb3000: Which format do I convert it to in order to be "Asterisk-friendly" ?
11:36.27cneb3000go to that site, convert it to 8khz
11:37.18hrolfcneb3000: Yes, it is gone now.
11:37.26cneb3000ahh that fixed it? :)
11:37.30hrolfcneb3000: What did you do? And how do I make the rest of my files?
11:37.33hrolfcneb3000: Yes.
11:37.38cneb3000awesome, to secs ill find a link
11:37.54cneb3000have a look at this http://www.youtube.com/watch?v=Xa9S-bel9P4
11:38.02cneb3000i basically used a piece of audio software called audacity
11:38.10cneb3000and added a quick fade in to the beginning
11:38.28cneb3000the fade in, got rid of that little click at the beginning, cause it made it quiet
11:38.46hrolfcneb3000: Okay, let me give it a look.
11:38.52cneb3000good luck!
11:39.11kaldemarsox can be used to batch-process files using the "fade" filter.
11:39.42kaldemarmaybe a little easier and faster than editing a bunch of files by hand using audacity.
11:40.09hrolfkaldemar: Thanks, I'll also give it a try.
11:44.18hrolfcneb3000, kaldemar: What length of fade-in do I give?
11:47.32kaldemarhrolf: try what works for you.
12:04.48cneb3000hrolf: a very small one
12:04.59cneb3000like, i faded in to where it actually starts if that makes sense
12:05.33hrolfcneb3000: Tried 170 ms and it worked fine.
12:05.42hrolfcneb3000: Thanks.
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14:37.24Kattypesters Qwell
14:37.47*** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net)
14:37.54Kattyhi tzafrir
14:38.21tzafrirhi
14:38.27Kattyhow're you dear
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14:39.24tzafriryou're not confusing me with tzanger, right?
14:39.24tzafrirand quite OK
14:39.38Kattyno i'm not (=
14:39.44Kattyand i'm glad you're doing just peachy!
14:39.56Kattytzanger: but hello to you too :P
14:40.17tzafrirenjoying the weekend
14:40.24Kattyyou get a 4 day weekend?
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14:40.31Kattyhugs slav3_kitten
14:41.01slav3_kittenyo Katty
14:41.04slav3_kittensnug purrs
14:41.25Kattyslav3_kitten: what's shakin
14:41.44slav3_kittensitting in pittsburgh at anthrocon
14:42.46Kattywhat's anthrocon?
14:42.53jmetrofurry convention
14:43.16Kattyah right.
14:43.19sruffellslav3_kitten: really?
14:43.28Kattyslav3_kitten: hope you're having a grand ol time!
14:43.33Kattyslav3_kitten: each some junk food for me
14:43.47Kattyhugs jmetro
14:43.49jmetroi have a friend that goes to MFF
14:43.50jmetroHi katty :3
14:44.10Kattyjmetro: i see you survived the fireworks
14:44.16Kattyfistbump jmetro
14:44.38Kattygmorning sruffell (=
14:44.48slav3_kittenyea sruffell
14:44.49sruffellwaves
14:45.01slav3_kitteni'll eat all the junkfood for you katty
14:45.05slav3_kittenALL OF IT!
14:45.06Kattywoohoo!
14:45.08sruffellhah…that's cool.  For work or just personal interest?
14:45.59sruffellProbably one of the only convention with a page on Behavior Around Fursuiters. http://www.anthrocon.org/behavior-around-fursuiters
14:47.14jmetroKatty: I did! *fistbumps* My doggies almost didnt.
14:47.23jmetroThey were shaking so hard.
14:48.32slav3_kittenpersonal interest sruffell
14:48.58Kattyjmetro: poor things )=
14:49.04slav3_kittenbeen a furry for a long time
14:49.08Kattyslav3_kitten: did you get all dressed up for it?
14:49.24jmetroKatty: We wrapped them in blankets and made a fort for them =)
14:49.32slav3_kitteni got my skirt, hosiery, tail, and ears on
14:49.41slav3_kittenso yea totally dressed up for it
14:49.47Kattyslav3_kitten: woohoo!!
14:49.53sruffellnice!
14:49.58Kattyslav3_kitten: how long did the tail and ears take you to make?
14:50.03Kattyslav3_kitten: or were they premade?
14:50.19slav3_kittencouple hours
14:50.24Kattyjmetro: that was nice of you :> i wear that they make a sort of snuggly fit doggy wrap for such things.
14:50.29Kattyerr i hear.
14:50.35slav3_kittenlong as hell pruple feathered fur with a green tip
14:50.46slav3_kitteni loves them so much
14:50.51Kattyjmetro: i've seen them at petco for plane trips and what not
14:50.59slav3_kittenmistress is also here from england
14:51.05Kattyslav3_kitten: that sounds pretty elaborate!
14:51.09slav3_kittenso it's been pretty fun so far
14:51.49jmetroKatty: They do, its called a Thundershirt.
14:52.37Kattyjmetro: ah right. it's nice to know that a Thundershirt (or blanket wrap) will help
14:52.46Kattyjmetro: should i ever get another doggy
14:53.15slav3_kittenKatty, yea it is. i may get a picture taken for you if you want
14:53.25Kattyslav3_kitten: sure if you want!
14:53.30Kattyslav3_kitten: you can send it to me in /query
14:53.36slav3_kittentit for tat though :P
14:53.44Kattylol
14:54.10slav3_kittenand i'd not have it for a while. need to get all the pictures an shit sorted after con is over
14:54.41Kattywhenever is fine, i just like crafty things!
14:54.57slav3_kittenso, want me in clothes or out?
14:55.11Kattyyou can just take a picture of the costume if you want.
14:55.34Kattyor whatever crafty bits you're particularly proud of
14:58.39slav3_kittenwell i'm quite fond of my crafty bits :P
14:59.00Katty^_^
14:59.11slav3_kittenif you know what i man
14:59.13slav3_kittenmean
14:59.25slav3_kittenanyhow. off to con!
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15:13.57aberriosslaps Rumbles
15:14.10jmetro^ with a large trout?
15:14.15aberriosand that
15:16.44Kattyahh, the classic Dead Fish Greeting of IRC
15:17.39WIMPyWho said it has to be dead?
15:18.31aberriosusually wet, being a fish..
15:18.44jmetrothe fish flop adds extra force!
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15:19.17KattyWIMPy: well considering we started trout slapping in say, the early 90s...
15:19.37KattyWIMPy: either the fish is either dead, or older than some of the asterisk folks
15:20.50WIMPyOhhhh, now it even has to be the same fish all the time.
15:20.58Kattyhehehehe
15:21.22[TK]D-FenderFEELING LIKE A DEAD DUCK!
15:21.28[TK]D-Fender</tull>
15:21.42Kattygood morning fenderbender.
15:23.10jmetrogenderfender?
15:24.12[TK]D-FenderKatty: Mew.
15:24.20Katty[TK]D-Fender: how're you dear?
15:25.07[TK]D-FenderKatty: TGIF ... doing OK.  Work running smooth, had meeting with band last night and ironed out or new material process, trimmed our existing list, and planned out the marketing responsibilities
15:26.17Katty[TK]D-Fender: wow, sounds like a full week
15:26.47[TK]D-Fenderthat was last night :)
15:26.47Katty[TK]D-Fender: we don't have a lot of band marketing out here, just fliers and what not posted about
15:27.41Kattygood lord, do you never sleep!?
15:28.01[TK]D-FenderKatty: We ironed out the process for getting "staged" pics" for posters, getting multiple proposed official logo samples, etc.  Then we'll be able to finalize the image on our FB page, build up a website, make posters & business cards, etc
15:28.10WIMPyYou get enough sleep when you're dead.
15:28.21Katty[TK]D-Fender: ahh.
15:28.37Katty[TK]D-Fender: i hope your stage photos are humorious
15:28.39[TK]D-FenderKatty: We didn't actually DO the work, jsut laid out the path
15:28.59Kattyalways like seeing folks with a sense of humor
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15:43.57tzangerKatty: hello
15:48.10igcewielingscreams obscenities at PHP
15:50.33jmetroigcewieling: Dont you talk that way to him! He has done nothing wrong!
15:53.12cneb3000whats wrong with PHP?
15:53.40WIMPyprepares for a big spike in used bandwidth.
15:54.12jmetroa lot of people call it a "double claw hammer" because it "does everything wrong"
15:54.26jmetrobut i call them "c++ addicts that like unneccessary outdated code languages"
15:54.42Kattystuffs a burger in igcewieling's mouth
15:55.25igcewielingcneb3000: I'm sure we'll make up soon the makeup coding will be awesome
15:55.51jmetro^ seee
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15:57.24igcewielingPHP is like using a C Hammer, a POSIX screwdriver, a Perl saw all in one tool.   It may not do anything perfect, but it does lots of stuff well.
15:58.52[TK]D-Fendermore like "well .... it does lots of stuff" :)
15:59.07jmetroI'd rather code in PHP than look at C++ code.
15:59.39jmetroAs "bad" as php is, keep in mind you could be working in c++ and feel good about yourself
15:59.41WIMPyThe chinese multitool from the DIY maket.
16:00.37igcewieling"strings" (or their non-existence) in C is why I hate C.
16:00.39mattpWhen I join the conference (provided by MeetMe application), I get a short ~100ms jingle (it sounds a bit like a trumpet).  Where does this come from? I've not found anything likely in the sound directory (but also don't know what I'm looking for)
16:00.59jmetromattp: does it show in the console when you join?
16:01.00igcewielingmattp: either meetme.conf or the options passed to MeetMe
16:01.13WIMPyigcewieling: That was an easy one. But a massive one as well.
16:01.40WIMPyProbably the reason for at least 50% OF ALL CRASHES AND SECURITY ISSUES WE SEE.
16:01.43WIMPyoops
16:01.47igcewielingHmm?
16:02.08WIMPyThe other half is the pointer in general.
16:02.18igcewieling*nod*
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16:04.28mattpjmetro: no.
16:05.51mattpigcewieling: neither of those. But interestingly, I just got a spam call from an autodialler. When the human came on, I heard the same mini jingle.  Coincidence that they use asterisk, perhaps, but it's an indication that the sound is one which comes bundled with it.
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16:14.21srulikhi everyone, i am after a PBX with a particular feature, can anyone tell me if asterisk has this feature?
16:14.21srulikI want to call into my sip account, get a prompt for a pin and then get a dial tone to dial an external number (like a calling card) does asterisk have this feature?
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16:16.30WIMPysrulik: No, Asterisk is not a PBX, but such a thing is easy to configure.
16:16.32cneb3000hey srulik, you mean DISA?
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16:16.47sruliki dont know what is DISA ;-)
16:17.16cneb3000http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA <--- 'direct inward system access'
16:17.41srulikcneb3000: thanks i will check that
16:17.41cneb3000or https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_DISA
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16:17.51cneb3000^^^ that will probably be more up to date
16:18.09ChannelZcore show application disa
16:18.17ChannelZeven more
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16:19.34cneb3000hm. if I type that into the asterisk11 box right in front of me, it's not there. probably wasn't compiled in?
16:20.40srulikcneb3000: can this be setup from the web GUI?
16:20.56cneb3000srulik: which web GUI?
16:21.05cneb3000srulik: freepbx or something?
16:21.06srulikasterisk
16:21.14srulikyes freepbx
16:21.27WIMPyAsk the ones who did the GUI. Asterisk doesn't have one.
16:21.36cneb3000srulik: asterisk is kind of like the underlying software as it were.. it does all the dialling and stuff. FreePBX is just a website to make it easier to use
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16:22.13WIMPy"claims to make it easier"
16:22.18cneb3000sure lol
16:22.24cneb3000havn't used one in a while myself
16:22.26sruliki understand that, i have only started using linux 6 months ago (what a divorce from MS) so i am not yet good enough with command line
16:22.36cneb3000haha srulik we have to start somehwere
16:22.43cneb3000in Linux, get used to reading manuals
16:22.54cneb3000because, there's a culture to learning to doing things yourself :)
16:23.06cneb3000for example - http://wiki.freepbx.org/dashboard.action
16:23.07srulikam reading, still have a day job and family, cant read all in 1 day
16:23.11cneb3000haha
16:23.20cneb3000read that website. your DISA answers are in there somewhere :)
16:26.10cneb3000srulik: I've never used freeswitch until recently. that also has a learning curve.. trust me, learning to read manuals is a good skill.
16:26.33cneb3000espiecally since.. alot of the time.. some of this software is free, and the developers dont always have the time to delve into how to actually use the software :)
16:26.53cneb3000(not referring to asterisk.. dont hit me..)
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16:40.26cneb3000generally speaking - what would be the most efficient way of storing asterisk CDRs on a remote server. Assuming the server was on the same network (i.e. ping of < 1ms).
16:40.37cneb3000am i right in saying storing them in a database as opposed to a csv?
16:41.01*** join/#asterisk teff (~teff@client-82-26-202-36.pete.adsl.virginmedia.com)
16:41.03cneb3000this is generally speaking.. i'm sure one type of database would be better than another. just looking for a push in the right direction :)
16:43.00GreenlightDepends on your requirements
16:43.24GreenlightIf you need to access them and search them regularly then, sure a database is best
16:43.45GreenlightIf you just need to know you have them somewhere, then archiving the csv's might be better
16:44.41*** join/#asterisk Mattski (~MattN@cpc11-whit4-2-0-cust235.5-2.cable.virginmedia.com)
16:44.49GreenlightOn a billing gateway, we store to local csv, local mysql, and remote mssql
16:45.30MattskiGood day, all. I hope you're well, and that your tankards are suitably filled.
16:46.17ChannelZMy whatnows?
16:46.28cneb3000cheers greenlight
16:46.30MattskiTankards, steiners, pint glasses...?
16:46.32Mattski;)
16:46.37GreenlightAhh pint glasses
16:46.38ChannelZOh.
16:48.08MattskiNuts, forgot I left the ironing on - brb
16:48.18ChannelZYe Olde Water Cup next to me is, anyway.
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16:51.35KattyMattski: my tankard is empty :<
16:51.47KattyMattski: but i did run across a hobnob recipe, so that makes it a bit brighter at least :>
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17:16.37MattskiHobnobs! Chocolate ones would get me into a beartrap tbh
17:17.54jmetroThin mints.
17:19.13MattskiThey'd get me into a nuclear reactor. Thank God I've got pizza otherwise I'd be drooling right now..
17:19.20[TK]D-FenderThin Mints : Staple food of fat people....
17:21.22jmetrohttp://weknowmemes.com/wp-content/uploads/2012/04/ate-4-boxes-of-thin-mints.jpg
17:22.11*** part/#asterisk ipiera (~Paul@ipiera.plus.com)
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17:29.59MattskiWanna be that cat.
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17:42.14MattskiDo any of you fine people have a few minutes to hear me ask a question you've probably heard a million times...? You can beat me with sticks afterwards if it helps!
17:44.37WIMPyDo we have to wait til then?
17:44.39WIMPy~ask
17:44.39infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
17:48.13MattskiApologies. Here goes: I'm using Asterisk 1.8.21.0. I've got the server talking happily to Sipgate.co.uk and incoming calls work, but I can't place any outbound calls. It looks like I've fudged authentication somewhere along the line. Setup is: home PC -> Asterisk box -> Sipgate.co.uk. Debug shows failure to authenticate on invite.
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17:48.57MattskiI've been Googling it for a couple of days, but I'm missing something. (Probably something small and obvious.)
17:50.43jagster`mattski, download wireshark
17:51.12jagster`you can get a packet capture going, filter to sip and see what the response is to your invite
17:51.28MattskiCheers, I'll give that a go now. :]
17:51.50jagster`once you know the problem, you can start working on the fix ;)
17:51.51[TK]D-Fender"sip set debug on" <-- from * CLI
17:51.56[TK]D-Fenderdo not waste time on external tools
17:52.06WIMPyOr upgrade to a version that tells you the result.
17:53.37Mattski[TK]D-Fender: I've set debug mode to on, and I see the following in the headers: SIP/2.0 407 Proxy Authentication Required
17:54.05MattskiI'm thinking I've got to tweak auth details somewhere, but I'm not entirely sure where. :<
17:56.33[TK]D-FenderMattski: pastebin the ENTIRE failed attempt from the very beginning to the end.
17:56.35[TK]D-Fender~pb
17:56.35infobotA "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.
17:56.37[TK]D-Fender^^^
17:56.51MattskiGotcha. 2 sec..
17:56.59cneb3000Mattski: have you seen this buddy - http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk
17:57.12jmetrocneb3000: have you seen this buddy -
17:57.13jmetro~book
17:57.13infobotAsterisk: 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
17:57.34cneb3000woah what did i do?
17:57.53MattskiI have seen that, and the config I'm using is based on that; I just can't make outbound calls for some odd reason. Back to PB..
17:58.07jmetroi dunno i just like using the book link
17:58.14cneb3000lol
17:58.49cneb3000Mattski: you see the top line. "register => SIP-ID:SIP-Password@sipgate/SIP-ID". where did you put that in the sip.conf file?
17:59.59cneb3000NORMALLY, most people put it in under the [general] context. so..  something like - http://pastebin.com/dDuVqLB5
18:00.11cneb3000but on that help link, i guess it doesn't really tell you that
18:00.17[TK]D-FenderRegistration is not the issue
18:00.20[TK]D-FenderThe call is ARRIVING.
18:00.25[TK]D-FenderLets just get the debug
18:00.48cneb3000you owe me a snickers if I'm right.
18:02.00[TK]D-Fendercneb3000: He already confirmed the call was coming in and getting a 407.
18:02.07[TK]D-Fendercneb3000: You lost that bet the moment you made it
18:02.51cneb3000\0/
18:03.15MattskiSorry for the delay, here's what I could get: http://pastebin.com/4gFcbmgB
18:03.47[TK]D-Fendernot the whole call.
18:03.55[TK]D-Fender"core set verbose 10"
18:04.02[TK]D-FenderWe need to see it from the very start
18:04.08MattskiOk, 2 ticks
18:04.44MattskiOh arse, that's the Putty buffer misbehaving..
18:05.32Kattytakes the snickers from [TK]D-Fender and eats it
18:05.44Katty[TK]D-Fender: so sweet, you shouldn't have!
18:06.20[TK]D-Fenderwaits for the tranquilizers to work their way through Katty's system....
18:06.29[TK]D-Fenderidsposes of the needle
18:06.31MattskiH'okay, try this: http://pastebin.com/1qh4u1Zj
18:06.59Katty[TK]D-Fender: i could go for a nap.
18:07.18[TK]D-FenderMattski: Still not the entire call.  We don't see the start of the INVITE
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18:08.23[TK]D-FenderMattski: the ENTIRE call.
18:08.40[TK]D-FenderMattski: if this is you using another device to place a call... we should see THAT debug along with it.
18:08.51cneb3000im off - good luck mattski
18:08.51[TK]D-FenderMattski: When I say "entire call" I mean ENTIRE CALL.
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18:11.39Mattski[TK]D-Fender: I'm using Linphone from a W7 box, there's nothing in the debug window except for keepalive messages. Hopefully this has got everything this time: http://pastebin.com/y5iaDx4s
18:11.49MattskiIf not, make with the stick hitting.
18:12.17[TK]D-FenderWhat's with the ANSI junk in there?
18:12.29[TK]D-FenderIt's polluted with crap...
18:12.43MattskiSorry about that. Not having the best day ever, as you might've guessed
18:14.41[TK]D-FenderMattski: idea : it seems to circle on that auth and you change the callerid prior.  add "fromuser=[yourusername]", "sendrpid=yes", "trustrpid=yes" to you peer and retest
18:16.07WIMPyOr better set them to pai than yes.
18:16.31[TK]D-FenderSure, why not...
18:18.46MattskiStill no joy. I'll just change the buffer settings for Putty while I'm at it
18:20.25MattskiBetter. Latest paste -> http://pastebin.com/75kh10ys
18:20.48WIMPyDo you still (try to) change caller ID?
18:21.55WIMPyyes.
18:22.07WIMPyThat doesn't work.
18:22.17*** join/#asterisk italoros_ (~italoross@187.60.66.11)
18:22.38WIMPyEither you have to use SIPID/name or use P-preferred-identitiy.
18:24.32MattskiOk, so what kind of syntax would I be looking at there? Set(CALLERID=SIPID) or similar?
18:24.51WIMPyLeave that part out until you've got your calls working.
18:27.51MattskiOk, so how about: exten => _90[1-9].,1,Dial(SIP/SIPID-here@sipgate,30,trg) ?
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18:32.18[TK]D-Fenderno
18:32.35[TK]D-Fenderyou need to spccify the target # there
18:32.39[TK]D-Fenderand you took that out\
18:32.57[TK]D-Fenderand r= evil
18:35.45The_MiserGreetings: I am using dahdi 2.3.0.1.  Do recent versions have the ability to do per FXS port different ring frequencies on a TDM400P card?  I have an old party line (54hz) phone I'd like to hook up, but before going through what is required (upgrading my install) want to be sure it will work.
18:42.45[TK]D-FenderThe_Miser: Seriously doubt it...
18:43.17[TK]D-Fenderthe you might try to divvy up your system.conf and change the loadzone & tonezone between the ports you're specifying....
18:51.08jmetroi love forging new territory with equipment that was designed to do what im trying to do but has 0 documentatin
18:55.41The_MiserFender: Interesting.  I did not realize that was possible.  Time to read a bit.  I could have *swore* I found a link that said it was possible with recent versions of dahdi and the wctdm module, but of course cannot now find it.  (which is aggravating in itself).
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19:07.03eject_ckHi all, any suggestions for sip softphone for linux with g729 codec ?
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19:35.09ChannelZeject_ck: the non-free version of zoiper
19:35.28ChannelZg729 being a licensed codec you're unlikely to find anything free
19:36.15jmetroconvert it to g722 on your end, success.
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19:47.52eject_ckjmetro: what about bandwidth requirements? Only g729 works for me on 3G
19:48.03eject_ckI even ready to pay
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19:48.29jmetroprobably the paid zoiper then. i only run g711 or g722
19:49.54ChannelZthere's always gsm
19:50.31[TK]D-Fender[15:47]eject_ckjmetro: what about bandwidth requirements? Only g729 works for me on 3G <- iLBC does better on PLC for quality
19:55.10eject_ckChannelZ: I'm using voip client over 3g data connection, why you are talking about gsm ?
19:55.27eject_ck[TK]D-Fender: what's PLC?
19:56.03[TK]D-Fendereject_ck: Googleable
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19:57.33jmetroprogrammable logic controller?
19:58.22eject_ck<PROTECTED>
19:59.02jmetropigmy life circus?
19:59.04Kattypacket loss concealment.
19:59.09Kattythat sounds like..
19:59.18Kattyyou know what, nevermind. i'll just keep that one to myself ^_^
19:59.21eject_ckwell, what linux softphones support  ilbc ?
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20:09.52eject_ckwhat this warning means when I'm calling with iLBC codec  via via external trunk which is using  g729 codec
20:10.08eject_ck[2013-07-05 23:07:26] WARNING[2982][C-00000b43]: translate.c:343 framein: no samples for ilbctolin
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20:22.30jmetroIf i want my queue to ring an agent following rules i have to log them in using a local context right
20:22.36jmetrorather than logging in as a peername
20:24.05Kattyi bet you'd love to ring an agent in the queue.
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20:33.16[TK]D-FenderAnd .... it's checkout time... BBIAB
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20:38.23qakhanhi all
20:39.12qakhani am recording calls on asterisk. i want to create folder each date recordings
20:39.26qakhanhow can i make folder in dial plan?
20:40.14jmetrosame => n,NoOp(SHELL(php dosomescript.php))
20:42.54qakhanis there any fuction of veriable which can make date folder?
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20:49.05jmetrowell
20:49.09jmetroyou can mkdir
20:49.15jmetroand timestamp
20:49.19jmetroand then mkdir with a timestamp.
20:49.28jmetro\o/
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20:50.55qmr[Jul  5 16:50:39] NOTICE[12049]: chan_sip.c:20325 handle_request_invite: Call from 'qmr' to extension '1002' rejected because extension not found in context 'users'
20:51.10jmetroi think extension 1002 doesnt exist in the users context.
20:51.33qakhanjmetro what the the command>
20:51.34qakhan?
20:51.47jmetroqakkhan lmgtfy.com
20:51.58qmrhttps://wiki.archlinux.org/index.php/Asterisk  my setup looks pretty much like that
20:52.06qmrI have 2 softphones connected but they can't call each other
20:52.36jmetrowhat is their context
20:52.47qmrcontext=users
20:53.03jmetrodoes your extensions.conf have a [users] context
20:53.25qmrprobably not - what kind of bits should I put in there?
20:54.08jmetroi think you need to set your [users] context in sip.conf to have them dial out in [INSERT-CONTEXT-NAME-TO-DIAL-OUT-ON-SOMETHING-UNIQUE-PROBABLY]
20:54.50jmetrothen make that context in extensions.conf and have it be filled with stuff like exten => [INSERT-EXTENSION],1,Dial(SIP/[PEERNAME-OF-EXTENSION])
20:54.54jmetro~book
20:54.55infobotAsterisk: 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
20:55.05jmetro^ base your server off of stuff like that and it will work gooder.
20:55.58jmetro[the free online book is going to be your best resource on the internet btw]
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20:56.25qmrlast time I set this up I just added users and everything worked :\
20:56.46jmetrocontexts are like subroutines [if you program much]
20:56.50jmetroeverything has to point the right way.
20:57.05jmetrophones are set up with a default dialing context and it sounds like they have the wrong one.
20:59.47qmrderp
20:59.49qmrthanks jmetro
21:00.13jmetronp
21:00.28jmetrowas it in sip.conf?
21:00.34qmrany recommended connection to PSTN?  something I can spend $5 on and not worry about.  I won't be making many calls
21:00.44qmryea - I had the group in sip.conf instead of extensions.conf
21:00.50jmetro:)
21:01.40jmetroim not too sure about connection to the pstn, but you can look at a SIP Trunking provider [i use that and it works pretty great]
21:06.53qmrany that you recommend?
21:07.02jmetrodepends where you live.
21:07.03*** join/#asterisk transfinite (~aleph_0@unaffiliated/transfinite)
21:07.05jmetroi think theres a list?
21:07.10jmetro~SIPTrunking
21:07.22jmetroi know theres a list but i have no idea how to bring it up
21:09.39qmrI'm in .us
21:10.12jmetrowell yeah
21:10.20jmetroim in the US as well but midwest varies from east coast, etc.
21:10.38jmetrogoogle "[your region or state] SIP TRUNKING PROVIDER"
21:10.53jmetrothey are usually per minute
21:12.16eject_ckwell, I've paid for zoiper 29 EUR
21:12.27eject_cktheir site sucks
21:13.17eject_ckI've downloaded some binary zoiperbizg729 which don't start :(
21:13.31eject_ckubuntu x64 12.10
21:13.45jmetroyoure running ubntx64?
21:13.48jmetroand its for 64bit?
21:16.11qmrjmetro: Do you have any tips on recording all calls?
21:17.19jmetromixmon is the best from what i know
21:17.35jmetrodump them into a harddrive that is NOT your main one
21:17.53jmetroor you can do like, packet captures.
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21:54.13qmrjmetro: I almost have this working.  I have a connection to localphone now, I just need to get outgoing calls to work
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21:55.13jmetroqmr: that will all depend on connecting to your sip trunk provider
21:55.16jmetroor etc.
21:56.18qmrI am connected, just not sure how to dial out to the world
21:56.25qmrhttp://www.localphone.com/help/voip/device_guides/softphone/asterisk
21:58.32qmrlooking at the consoel it looks like it tries to call an extension when I enter   a phone number
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23:43.35igcewielingqmr: you need to read the book
23:43.38igcewieling!book
23:43.42igcewieling~book
23:43.43infobotAsterisk: 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
23:48.14Kattyevening
23:49.57WIMPymorning
23:50.29Kattywaves to WIMPy
23:50.58WIMPywaits for the video stream
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