IRC log for #asterisk on 20130609

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02:01.07volga629How I can check what preventing from asterisk no finish the load in term of modules ?
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10:46.13[sr]WIMPy:  is it raining there?
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11:59.45vlad_starkovQuestion: Choosing between voice recording apps, which one is better and why: Monitor or MixMonitor?
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12:57.46flinghow to allow 'L16/16000' codec for a peer?
13:02.53flinghmm hmm I need slin16
13:04.35flingit works but I don't see it in codecs list :p
13:04.43flingalso the sound is ugly
13:07.54vlad_starkovQuestion: Choosing between voice recording apps, which one is better and why: Monitor or MixMonitor?
13:13.14kaldemarvlad_starkov: neither of them is better. read what "core show application Monitor" and "core show application MixMonitor" say and judge for yourself.
13:13.57kaldemarvlad_starkov: main difference is that MixMonitor mixes inbound and outbound audio to a single file, where as with Monitor you get two files.
13:14.02vlad_starkovkaldemar: I read somewhere that MixMonitor works better on call parking, on hold, call transfer etc.
13:14.55vlad_starkovkaldemar: I prefer to mix call manually to make it sounds stereo
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13:58.32ildohi same one can help me with my raspberry and asterisk?
13:58.49kaldemar~ask
13:58.49infobotQuestions 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.
14:00.35ildoi´m new to all this, i have just configured a sip out account, and want to configure the disa access!
14:00.52ildoand have no ideia how to do it!
14:01.23WIMPyWhat disa are you talking about?
14:01.48ildoi have a huawei usb pen in raspberry
14:02.30WIMPyI guess, you better tell use the whole story of what you have first and then what you want.
14:02.52ildoi want to make a call from any regular phone to the card in huawei pen, and with dtmf dial a phone number to goes out by sip
14:04.15ildoright now i have this
14:04.18ildo<PROTECTED>
14:04.18ildo<PROTECTED>
14:04.19ildo<PROTECTED>
14:04.35WIMPyFirst thing unclear: Do you want to tall out on the USB stick or is that (rather unrelated) only to provide you with internet?
14:05.12vlad_starkovQuestion: Can anyone explain me the main difference between Monitor and MixMonitor other than latter mixes two channels automatically, when first saves both channels recordings separately. Any other difference? Maybe there is something special on call transfer?
14:05.15WIMPyLooks like the first part. So where does SIP some in to play there?
14:06.00WIMPyAnd how do you connect that regular phone? And does "regular" mean POTS for you?
14:06.12ildoi have configured my sip.conf
14:06.39ildojust with my acount details
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14:07.33WIMPyThis still doesn't make sense. Please asnwer the questions.
14:07.34ildothe idea is to enter by gsm and goes out by sip
14:07.55WIMPyAh
14:09.53ildocan you help me configuring this?
14:09.57WIMPyOk, so from what you pasted, you're missing the extension that's calld from chan_dongle.
14:10.11WIMPyAnd the messages clearly tell you what's the right one.
14:11.36ildook, but i can make the estension but for where?
14:12.03WIMPyextensions.conf
14:12.38WIMPyAnd usually it's not a good idea to use the default context yourself. Better define your own.
14:12.59WIMPyBut that needs to be done in chan_dongle as well.
14:13.35ildo<PROTECTED>
14:13.37ildoexten=>s,1,Dial(sip/inforsms)
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14:15.19WIMPyWhy do you Goto?
14:16.38ildoi´m not sure, i´m looking for same information on internet
14:16.47WIMPyDid you read the book?
14:16.50WIMPy~book
14:16.50infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
14:17.09flingWIMPy: hello
14:17.24ildono
14:17.29ildoi did not
14:18.21ildowith this config i have this:
14:18.24ildo== Parsing '/etc/asterisk/cli.conf': Found
14:18.24ildo<PROTECTED>
14:18.24ildo<PROTECTED>
14:18.24ildo<PROTECTED>
14:18.24ildo<PROTECTED>
14:18.24ildo<PROTECTED>
14:18.25ildo<PROTECTED>
14:18.26ildo[Jun  9 14:16:56] WARNING[1859][C-00000000]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000000] Don't know how to indicate condition 33
14:18.26ildo<PROTECTED>
14:18.26ildo<PROTECTED>
14:18.27ildo<PROTECTED>
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14:18.38WIMPy~pb
14:18.38infobotA "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.
14:19.46WIMPyThat Dial() is good to call a terminal. To place a call via an ITSP you need to add the number you want to call.
14:20.45ildobut i want to do that by dtmf!
14:21.02WIMPyAnd dias is one of the possibilities to read a number. WaitExten would e another. You could also use Read.
14:21.21WIMPyYou need to do that before you pass on the call via SIP.
14:21.47WIMPyDoing it afterwards only works if you have PSTN interface hardware.
14:22.30WIMPywonders if chan_gulp will be able to do it.
14:22.32ildoi just have te rapberry and the huawei pen
14:22.47WIMPyBut your ITSP probably wouldn't support it anyway.
14:23.57ildoitsp!?
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14:24.46WIMPy~itsp
14:24.47infobot[~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.
14:25.55ildoyes, i have from betamax
14:26.11ildowithout incoming number
14:29.48ildoi´m a newbe, can you send me same code lines?
14:29.57ildoto do this?
14:32.35WIMPyI gave you some examples of applications you can use and a book.
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14:37.53ildook, thanks
14:40.00ildobut for asterisk answer in mode of disa what i have to do?
14:44.12WIMPyAnd Disa() is one of the possibilities to read a number. WaitExten() would be another. You could also use Read().
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14:48.28ildolike this:  exten => 000,1,DISA(no-password,default)
14:49.03WIMPyNow you just need to find out where to put that in your dialplan.
14:49.22ildohihihi
14:49.48ildocan you help me?
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14:52.00ildoif i do like this:   exten=>+1234567890,1,Goto(s,1)
14:52.01ildoexten => s,1,Background(Sound-file)
14:52.01ildoexten => 000,1,DISA(no-password,default)
14:59.03WIMPyYou should start with the dialplan basics in the book.
15:01.25ildook, thanks, i will read
15:01.37WIMPyWithout that, things won't make much sense.
15:10.26ildook, thanks, after i read the book i say samething
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16:26.30LokiScarletAnyone had alsa output issues before? I'm getting clicks of pitch relative to the sound that should be coming out. I don't know what details are relevant so if you need more specific info just ask for it and you shall receive ^_^
16:27.28LokiScarlet(This is with asterisk outputting so alsa, of course. Trying to use the sound card for PA)
16:27.38LokiScarlet(*to alsa)
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17:43.20igcewieling1we use dedicated devices on FXO ports for PA, never built in sound
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17:48.43LokiScarletThat sounds like a personal problem.
17:49.12volga629Hello Everyone, how I can know what exactly the error when asterisk says [2013-06-09 07:07:12] DEBUG[5784] app_voicemail_imapstorage.c: mailstream not set.
17:50.27volga629what is connection timeout or usename ? because mailstream not set is to general
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18:55.57wdoekesvolga629: if you're running non-ssl, you can sniff the traffic and look at the imap conversation
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18:57.39[TK]D-FenderAnd if you're running SSL .. you con MITM it and sniff the traffic :)
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19:24.30vlad_starkovQuestion: What are the main problems with call recording in Asterisk? Disk IO? Call transfers? Audio file transcoding?
19:26.43ChannelZNone of them are real problems unless you have a system already at capacity, tons of calls you're recording at once, etc.
19:28.29vlad_starkovChannelZ: Ok
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19:38.17ChannelZIs there some site out there that I can punch in an IP block and have it tell me who owns every subnet inside it?
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20:01.54wdoekesChannelZ: whois?
20:04.03WIMPyLokiScarlet: That's normal if your hardware doesn't support the samplerate/ You can easily get around that by using dmix.
20:19.19ChannelZyea but I kind of need 'recursive' whois
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20:28.16[sr]WIMPy: work tomorow?
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20:55.20WIMPy[sr]: Why tomorrow?
20:55.48[sr]WIMPy: holiday here
20:57.48gustoha
20:57.59gustowhat holiday?
20:59.44[sr]republic day
21:00.23WIMPyNope, no holiday. But holidays don't mean no work anyway.
21:01.01[sr]bingo
21:01.09[sr]i havent stop since friday
21:01.27[sr]that's 3 days coding without anyone calling me!
21:01.53WIMPyNice
21:04.49[sr]WIMPy: dont tell anyone i'm also a .net coder, nobody needs to know!!
21:04.58WIMPyoops
21:05.24WIMPywould like to lock himself in in the basement to do some coding.
21:05.46WIMPyBut instead I do all kinds of other stuff.
21:06.07[sr]i code more on saturday and sunday than the 5 week days..
21:06.57WIMPyIn those times when I used to sti in an office, I usually went home when I had to do some serious coding.
21:07.05WIMPysit
21:08.04[sr]i know the feeling
21:14.24ani216Hello everyone
21:16.43volga629If I will try raise level of debug ?
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21:59.43volga629same IMAP server inside network working fine and through internet connection from outside don't working with IMAP voice mail storage
22:00.14volga629another email clients from outside didn't report any issue
22:06.23volga629I see SSL traffic on tcpdump
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