IRC log for #asterisk on 20121202

00:01.12valueapeQuick question: I'm trying to execute a script when a conference closes down (to compress and ship recorded conferences). I had been using MixMonitor's script execution to run before - but now I want to use CONFBRIDE record_file in my dialplan ("ext-meetme").I'm not sure where to start how to do this.
00:01.58valueapeBasically I'd be using System () call - but I'm not sure where I need to put it - for when the conference closes not when a user leaves
00:08.13SeRislav3_kitten: why it wouldnt work?
00:08.20SeRiis cisco por non standard?
00:08.28SeRis/por/poe/
00:08.37[TK]D-FenderIt is its own standard
00:08.44slav3_kittenyep
00:08.47[TK]D-FenderWhich virtually no other manufacturer cared about
00:08.55[TK]D-Fenderit is NOT 802.3af
00:09.03slav3_kittencisco poe is pre standard. out of it 802.3af was developed
00:09.35slav3_kittenwhich is why all modern cisco stuff uses 802.3af and can do cisco as well
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00:27.54SeRiThats sucks for you
00:28.42slav3_kittenmeh
00:36.51ghost75does anyone know something about a lockfile for voicemail ?
00:43.08Kobazso how does srtp work if you're trying to do call recording
00:43.23Kobazaccording to polycom info, the phone will drop any call that's not srtp end-to-end
00:44.04Kobazbut using asterisk. you have two dialogues open, one from a phone to asterisk, and another from asterisk to another phone
00:44.13Kobazso asterisk is sitting in the middle, is it the srtp endpoint?
00:45.23[TK]D-FenderKobaz: * is a B2BUA.  There is no such thing as end-to-end.  You should know better
00:45.35Kobazwith a reinvite you have end-to-end
00:45.45Kobazbut then you cant record anyway with a reinvite
00:45.50Kobaz...just making sure
00:45.55[TK]D-FenderIf you're recording, you CAN'T reinvite
00:46.00Kobazyeah
00:46.02[TK]D-FenderThe audio HAS to pass through *
00:46.05Kobazyeah
00:46.13Kobazbut i was saying for argument sake if you weren't recording
00:46.16Kobazand did a reinvite
00:46.23Kobazthen it would be end to end
00:46.27[TK]D-FenderThey might completely renegotiate
00:46.44[TK]D-FenderI could imagine anyway, not having attempted it personally
00:47.36Kobazso i could have a one-off srtp endpoint sitting out there somewhere and everything else is just rtp
00:47.40Kobazwas what i was getting at
00:51.13[TK]D-Fenderyes
00:51.22Kobazcool cool
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01:16.28qakhanplz let me know cheap DID providers
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01:36.03jpsharphttp://bit.ly/Vccbnw\
01:38.55g_r_eekanyone know a good guide to configure asterisk 11 with real time?
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01:39.11g_r_eekor is it the same like 1.8 ?
01:43.30Kobazthe same
01:44.42g_r_eekKobaz: thans
01:44.45g_r_eekthanks
01:46.06ghost75your /var/spool/asterisk/voicemail folders also have no write rights on group ?
01:52.37Kobazright writes
01:52.56Kobazright rite writes
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01:53.57ghost75why i dont have?
01:56.55Kobazkite flight writes right fight rite
01:57.11ghost75shite
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02:06.31g_r_eekon a real time setup i am getting: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available -- do i need to install and add ons ?
02:08.24dfgas-cr48`hmmmm
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02:44.06g_r_eekfixed it had to install add-on on the menu select
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03:53.18infinity_does anyone know of a phone number that maps to enum that can be used to test with ?
04:01.58ChannelZeh?
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04:03.42jpsharpTried: +43 720 0101011 This number is reachable only via ENUM+SIP. After connect you will hear the word "ENUM" followed by some music.
04:03.47jpsharp?
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04:08.08infinity_do any companies support calling via enum?
04:13.56drmessanoinfinity_, wouldn't that take a public registrar that's accessible worldwide?
04:14.36infinity_i thought there were companies suporting enum infrastructure. just no one is registered.
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05:52.49SeRip3nguin: pkg still in transit.
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06:42.20ChannelZis that a poop reference?
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07:02.09dubzillai love asterisk
07:02.18dubzillapaired with a database you can do amazing things
07:17.37slav3_kittenwhere on earth is country code 97
07:18.15slav3_kittenhmm 972
07:18.16slav3_kittenisreal
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09:43.32ectospasmslav3_kitten: http://countrycode.org/
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09:56.07J4nusI would like to configure Kamailio as a SIP registrar and have several trunks with different Asterisk (playing the role of a media gateway). Is somebody able to give me some documentation or some help ?
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11:34.44weinerkPlease help, on asterisk 1.4 - in a macro if I do a lengthy operation (a few seconds), then other concurrent calls clober ARG1.
11:34.46weinerkIs that possible? Any ideas on how to address this.
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13:46.58slav3_kittenectospasm, thanks :D
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14:42.51BlackBishopin ael .. is there any way I can make a variable _XXXXX expression to match both 2 to 5 char ?
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14:50.16Kasper__hello
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14:53.37mathihi
14:55.24mathistupid question but... to configure asterisk you just need to update a few files (extensions.conf, sip.conf, ...). So when do you really need a GUI interface (FreeBBX, Elastix, ...) to manage your server?
14:57.26slav3_kittennever
14:57.46slav3_kittenall those gui projects are kinda dead iirc
14:58.16slav3_kittenthe confs can get convoluted an such but that's what syntax hilighting is for
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17:25.02sybrenhi there!
17:25.42sybrenI'm trying to hook up Asterisk with Google Talk, so that I can do SIP <-> gtalk voice calls.
17:26.30sybrenIt works just fine, but *only* if I do a video-call from gtalk -> SIP. Then the audio works just fine. When I do an audio-only call gtalk->SIP or SIP->gtalk the call is connected, but both ends are silent.
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17:48.23sybrenbtw, it seems to work fine when I use GTalk within GMail on my desktop, but not with the GTalk client on my mobile phone (GTalk version 4.1.2 on Android 4.1)
17:51.02InvaderSo I have thought of a new project and woundered what you guys thought Ras Pi with arduino/GPRS sheild. Install asterisk with open bts
17:53.35WIMPyWhere does trhe arduino fit in there?
17:54.44InvaderBroadcast a gsm singal For your own gsm network
17:55.31jpsharpYou need something more than a GPRS shield.
17:55.59InvaderYour not thinking a USRP are you jpsharp ?
17:56.48jpsharpSomething along those lines.  The GPRS shield doesn't have the smarts to be a base station.
17:56.59WIMPyYou need something that will do the radio part.
17:57.47WIMPyThe calypso terminal based basestation was put on ice as far as I know.
17:57.48Invaderjpsharp WIMPy that is kind of what I was thinking but was not 100% sure. I do have a few "cheap" usrps I can try that just work on UBS
17:58.08WIMPyBut the topic might fit better to #osmocom.
17:58.23InvaderI am talking to the openbts guys atm
17:58.32WIMPyok
17:59.05jpsharpAsterisk + openbts has been done lots of times.  There's just questions of financing the base station part of it and how legal you want it to be.
17:59.06WIMPyOr you get an old bas station and only use OpenBSC.
17:59.11WIMPy+e
17:59.11InvaderI work at a gsm company and the thought of creating an open network its awesome to me
17:59.49WIMPyWant to take over your employer? ;-)
18:00.03jpsharpSpectrum licensing will be your proverbial Achilles heel.  Radiowaves don't come cheap.
18:00.08InvaderWe are a small telco :) cant sorry
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18:00.32Invaderjpsharp, yep dont remind me.
18:00.44WIMPyI guess that depends on where you are.
18:01.02WIMPyDoesn't seem to be an issue here.
18:01.07InvaderThere are unlicensined spectrum but none in the gsm 700-1900 range.
18:01.14jpsharpTrue.  Not much spectrum regulation in outer botswannaland.
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18:01.52greenwolfhas anyone used the g.722 liscense in a production asterisk system? if so how was the quality of the call and is it worth purchasing?
18:02.06Invaderthe FCC sent us a fine open for 70K for being on the wrong band
18:02.10jpsharpg722?  Doesn't need to be licensed.
18:02.11WIMPyG.722 is free
18:02.19jpsharpg729 is good quality, though.
18:02.20WIMPyDid you mean G.729?
18:02.49greenwolfyes sorry thats what i meant
18:02.50greenwolflol
18:02.54WIMPyThere are many versions of G.729 with different quality.
18:03.05greenwolfhow is it against actually 64Kbps quality
18:03.27greenwolfwell does it sound very close to actual analogue phone qulaity such as ulaw or alaw
18:03.52jpsharpI'm always scared of getting an FCC notice whenever I screw up and transmit wrong on my ham station.
18:04.30greenwolfyea its happened to me...i am a ham btw
18:04.44greenwolfthey keep your liscense in a pending decision for a long time it sucks
18:05.08greenwolfive been pending my liscense for over 2 years now...i dont think they are gunna pull it but i think they do it just to scare you
18:05.46jpsharpThey probably lost the case file.
18:06.33InvaderMe and the FCC swear at each other a lot.
18:07.32Invaderjust to get a basestation up and working with the paper work at the fcc is 50k
18:07.33greenwolfi just checked the ULS on FCC website and im still PA decisons
18:07.38greenwolfholy crap its been over 3 years now
18:07.57jpsharpMy dad goes round and round with them on a regular basis licensing 220Mhz commercial stuff.
18:08.16greenwolfthey wont even update my address or call sign to my new address...crazy it you ask me..gotta be careful on ham radio because theres always someone out there that will snitch on you and send a letter to the FCC about anything u did wrong
18:08.26greenwolfjpsharp: yea pretty kool
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18:15.38greenwolfalways be careful of what your broadcasting on ham radio cuz theres always someone listening to rat you out to the FCC
18:15.52greenwolfits pretty sad now adays
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18:17.04greenwolfhas anyone seen a difference running asterisk on freebsd rather than linux?
18:21.06wltjris there a way to match 1 or 2 digit extension, with the same expression?
18:22.13[TK]D-Fenderwltjr: No
18:22.31[TK]D-Fenderwltjr: You can do 1 or LONGER, but not 1 or 2.
18:22.37wltjr[TK]D-Fender: thanks didn't think so, was hoping to not duplicate stuff, but likley have to oh well :)
18:23.22WIMPyYou can use Goto().
18:24.42wltjrWIMPy: I was thinking I might be able to go that route, probably can
18:31.27wltjrgoto worked fine less is more :)
18:31.29greenwolfwimpy: can u answer the question i asked
18:32.23greenwolfare their better results when running asterisk on FreeBSD as oppose to linux?
18:32.41WIMPyThe bsd one? No. But apart drom drivers for hardware I wouldn't expect any differences.
18:33.45greenwolfso if i have digium cards will the hardware work just as fine with BSD? or are the drivers only for linuxs
18:33.49greenwolflinux*
18:36.27WIMPyI thought there was a BSD version, but I don't see it. So I guess that makes a no.
18:38.33greenwolfso i would have to run a whole VoIP system with asterisk to be able to get asterisk to work on BSD
18:38.46greenwolfwith no digium or telephony cards
18:39.54WIMPyYou should at least be able to use CAPI, I guess, but I really have no idea if there are any other options available.
18:42.22p3nguinjpsharp, greenwolf: What kinds of radio(s) do you have?
18:43.35*** part/#asterisk sybren (~sybren@213.93.82.119)
18:45.49greenwolfi got 2 icom's and a yaesu
18:46.15greenwolfi used to have a bunch of mobile stations but got rid of them and moved everything to portable sattions
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18:46.31p3nguinI'm looking for model numbers.
18:46.51p3nguinLike a 2100 or a 3000.
18:47.05greenwolfso ur looking to buy?
18:47.23greenwolfmy friend has a bunch of HT radios for sale and a really low price
18:47.33p3nguinNo.  Just wanted to know what your're running.
18:47.36greenwolfi bet you will get a really good deal if you go thru him
18:47.42p3nguinWhat does he have?
18:47.50greenwolfyeasu Ft-60r
18:47.56greenwolfthats mine
18:48.08p3nguindual bander
18:48.16p3nguinI have an Icom 2100 mobile, but I don't use it currently.
18:48.42greenwolfyea the yeasu is a dual band and also has 6meter on it
18:49.00greenwolfi was going to have the crystal set modified so i can have 220mhz
18:49.04greenwolfbut never got around to doing it
18:49.11SeRi<PROTECTED>
18:50.06p3nguinThe FT-60R is a 2m/70cm radio... how do you have 6m on it?
18:51.02greenwolfsorry the 6m is a different one
18:51.06greenwolfits on the icom i have
18:51.18greenwolfyea the yeasu is a dual band 2m/440
18:51.30greenwolfhey have u ever played around with packet radio or echolink?
18:51.50greenwolfi was going to create my own echolink node station and have it interlink to the internet
18:52.00p3nguinI use echolink as a client node, but not as a link node.
18:52.03greenwolfmaybe even patch asterisk in it for calls
18:52.19greenwolfhave u ever got asterisk to patch phone calls for ham radio"?
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18:52.26p3nguinI never tried.
18:52.42greenwolfyour pretty smart i think you could get that going no problem
18:52.52greenwolfi was thinkin of doing something like that
18:53.13greenwolfi'll let you know how it works out...i might play with that tonight now that were talking about ham radios lol :)
18:53.22p3nguinIf it involves buying things, I probably won't get around to it.  I have a wife, kid, and rent to spend my money on.  I don't get to buy too many toys.
18:53.52greenwolfdamn too bad...but i hear ya you got alot of responsibilites to take care of before you can Play :)
18:53.53p3nguinI'm really interested in IRLP, but I'd have to buy the interface card.
18:54.13greenwolfwhat is the difference between IRLP and echolink btw
18:54.34p3nguinIt's very similar in how you use it, but it's a different technology underneath.
18:56.12p3nguinEcholink seems to be much easier for the user.  You just select the node and use whatever frequency they are set up on.
18:57.15p3nguinI guess I need to see if there are any IRLP apps for Android.  Then I can mess with the user end of that.
18:58.42p3nguinI was wanting to see if it was possible to run IRLP on HF.  I thought it might be interesting to talk on something like 80m over the internet.
18:59.06p3nguinEvery node I have seen so far is always 2m or 440.
18:59.25p3nguinor maybe 220, but I can't recall one specifically.
19:00.09p3nguinI think echolink only works with carrier squelch, so HF/SSB is out of the question, but I think IRLP is a little more flexible.
19:00.33p3nguingah
19:00.39*** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se)
19:02.48SeRicomcast seems to keep their node map very well hidden.
19:03.15p3nguinWhat are you trying to find on their system?
19:03.35SeRiI finally talk to the eng. He told me the issue was at the node in spring.
19:03.52SeRiso I am trying to see which one in this area.
19:04.27p3nguinWhat was your impression of the engineer vs. the other people you dealt with?
19:05.15SeRicurtious. He knew what he was talking about. he didnt try to talk me down as I dont know what the internet is. we where at level.
19:06.00p3nguinThat's often how they are.  Sometimes they are a little irritated when you ask technical questions, but they can give you the answers.
19:07.02SeRiWell he was very well educated with the history of my account so he knew I was the frustrated one.
19:07.10SeRihe started the converstaion the right way
19:07.20SeRino bull shit. staright to the point
19:08.23p3nguinGood.  I'm glad you had a pleasant experience with someone for a change.
19:08.38SeRiyeap.
19:08.47SeRiby the way the pkg still in route
19:08.50SeRiin MO today.
19:09.26SeRiI tested my own headset to see how far I can go with it. I went across the street with no issues.
19:10.00SeRiabout 3 houses down and still clear. But my office is in the front of the house which sort of helps
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19:22.21josefighello guys, If I want to setup my asterisk as a bridge, It's something like this: SIP-SoftPhone--->Asterisk--->H323-Provider I'd like to terminate my calls via my H.323 provider someone can help me ?
19:33.23leifmadsenjosefig: just use chan_sip on the one side for your sip softphone, then configure chan_ooh323 on the other
19:33.29leifmadsenAsterisk does the bridging automatically
19:34.54josefigleifmadsen: but I'd like to record the call details (CDR) on the asterisk and create the SIP users. I'm reading this guide http://www.callcentric.com/support/device/asterisk/1_8 just please let me know if I'm ok
19:39.22*** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4)
19:39.38[TK]D-Fenderjosefig: Multiple small bugs with that "guide"
19:40.22[TK]D-Fenderjosefig: And it is for setting up CC's SIP service.  This has nothing to do with using an H.323 ITSP
19:40.33josefigoh hello [TK]D-Fender :) nice to see you again
19:40.40[TK]D-Fenderjosefig: And doesn't show setting up a SIP phone type account either.
19:40.59[TK]D-Fenderjosefig: So far as I can see this doesn't actually demonstrate anything related to your stated goals
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20:08.42p3nguinzombie rabbit, or rabbit zombie?
20:14.25ghost75is there any way to change rights on voicemail folders within asterisk?
20:16.33[TK]D-Fenderghost75: When?
20:17.01[TK]D-Fenderghost75: and no, not "directly".  voicemail.conf dopes specify the perms it will use.
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20:20.31ghost75my voicemail folders have no write rights on group
20:21.19ghost75but all files within have 777, very strange
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22:38.52lukejtmight anyone know why I can't access most CDR() variables from the AGI when using the Dial command?
22:39.23lukejtnot only that but CDR(start) returns what should be CDR(end)
22:41.02*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
22:44.13[TK]D-Fenderlukejt: your timing is unclear.  When are you checking this exactly?  Where does the Dial come in?
22:53.11lukejt[TK]D-Fender: Simple script really: ANSWER, EXEC DIAL SIP/trunk/1234, user initiated hangup which sends a SIGHUP to my AGI. From my signal handler, I'm trying to access CDR(answer) but it's not available
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22:56.44lukejtAll CDR vars are populated when I just do an ANSWER, STREAM FILE file "", HANGUP
22:58.21ghost75the cdr vars are going to be modified during the call, is that the use of them?
23:02.37*** join/#asterisk josefig (~josef@unaffiliated/josefig)
23:03.21lukejtI'm just trying to access the answered time of my Dial command for billing purposes
23:04.27ghost75ok
23:04.39ghost75or just parse Master.csv :)
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23:26.16p3nguinSo...
23:26.17p3nguinI still need to know.
23:26.22p3nguinzombie rabbit, or rabbit zombie?
23:28.23Alex25What is the right way to play music in early media while desstination is 'ringing'? which command/option should be used?
23:30.10WIMPyPlayback(...,noanswer)
23:31.01Alex25yes, but i need Dial to execute while playing the file. how to get this done?
23:31.03WIMPyOr Dial(...,,m)
23:31.33p3nguinI would Dial with the m option and make sure to not Answer() before that.
23:33.21Alex25it seems the 'm' option only take moh classes, but not arbitrary files
23:33.27Alex25is this correct?
23:33.33p3nguinYes.
23:33.34WIMPyyes
23:33.50p3nguinIf you want to play files, you'll have to use Playback with the noanswer option.
23:34.25Alex25so is there a way to playback while Dial is executed parallel
23:35.17p3nguinI think a Dial macro should be able to do that.
23:35.38p3nguinDial(...,,M(your-macro-here))
23:37.21Alex25for the M option I'm reading it 'Runs the macro x when the call is answered...'
23:37.41p3nguinOh, crap.
23:38.23Alex25'answered' is not good :)
23:39.16Alex25ok np
23:39.46Alex25I'll set some MOH classes then
23:40.19p3nguinYou realize that playing it in early media may mean that no one ever hears it, right?
23:40.39WIMPyWhat makes you think so?
23:40.55Alex25it's only for a DID number
23:41.04Alex25so caller should hear it
23:41.20WIMPyDepends on your provider.
23:41.22ghost75moh reminds me every time of medal of honour
23:41.25Alex25i know
23:41.38p3nguinI occasionally try doing things in early media, and it sometimes is never heard.  Like when I test with my cell phone, I don't hear early media.
23:42.00Alex25you should hear it
23:42.31p3nguinAnything before the channel goes "Up" isn't heard.
23:42.53Alex25it should work in all developed countries
23:42.54p3nguinIf I do a literal answer or a playback, then I hear the audio.
23:44.00Alex25I actually have some apps which work in early media, so you can test
23:44.06p3nguinI'm not trying to tell you that it will not work.  I just wanted you to know that YMMV.
23:45.06p3nguinIf your results are satisfying, that's what matters.
23:45.10Alex25I know that
23:46.00Alex25usually bi directional early media is a problem in some countries, or using some providers
23:46.27Alex25but one-way audio allways worked for me
23:46.57p3nguinI'll have to check it out again soon.
23:46.59Alex25although I heard in some poor country it doesn't work properly
23:47.32Alex25call +390142276081
23:47.44Alex25it will not charge u
23:48.04Alex25and you'll be able to record a message
23:52.08p3nguinI can't right now; I'm in the middle of supper.
23:52.24Alex25bon apetit

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