IRC log for #asterisk on 20140323

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08:09.06twanny796chan_sip is not building, dependencies chan_local, res_crypto and res_http_websocket not met, where should I start?
08:12.21ChannelZyou probably are missing libuuid-dev or uuid-dev.  What distro, and what version of asterisk?
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13:02.31boratynskikamilHello.
13:03.02boratynskikamilI have one small question. Since 1st March i got 10 hangups with cause 41. Isn't it too many?
13:03.53WIMPyHow much is that in hex?
13:04.24WIMPyCould be anything, as usual.
13:04.32boratynskikamilWIMPy: What do you mean?
13:04.55boratynskikamilhttp://wklej.org/id/1309105/
13:05.14boratynskikamilhttp://wklej.org/id/1309106/
13:05.14WIMPycause 29
13:05.43boratynskikamilWIMPy: I didn't get hangups with cause 29.
13:06.00WIMPyYes you did. You just wrote it wrong :-)
13:06.17WIMPyAnyway: Is that a real line or a geteway (IAD)?
13:07.28boratynskikamilWIMPy: https://www.wolframalpha.com/input/?i=41+dec+to+hex+
13:07.29boratynskikamil<3
13:07.30boratynskikamilLovely.
13:07.31boratynskikamilSee it.
13:08.24WIMPyYou know, there's a reaso why cause codes used to be displayed in hex. They are organized in groups of 16.
13:08.48WIMPySo in hex you can see what group in the 1st digit and the cause in the 2nd.
13:08.53boratynskikamilEkhem. WIMPy I've got network terminator connected to ISDN card and then, SIP phone connected to Astrerisk-server.
13:09.08boratynskikamilWIMPy: Achso.
13:10.07WIMPyIf it's a real line, your telco seem to have trouble getting the call through.
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13:10.52boratynskikamilWIMPy: Yep. But, the problem is on previous server I had 41 hangups with this cause during 3 years. (:
13:12.24WIMPyWell, as te PSTN is being dismantled, it fails more often.
13:14.23boratynskikamilSo it isn't a server problem, right?
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13:23.24WIMPyNot yours, obviousely.
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13:26.06boratynskikamilWIMPy: Ok, could I ask one more question? I monitor OUTGOING calls: http://wklej.org/id/1309137/, and it monitors MoH. Is it possible to disable it by default?
13:27.00WIMPycore show application MixMonitor :-)
13:27.46WIMPyBut it depends on why you have MOH.
13:28.24WIMPyIf a bridged call is on hold, it would still be bridged.
13:28.34WIMPyAs fara as I know/remember.
13:29.03boratynskikamilHeh.
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19:57.14billum[Sorry, not sure if my last message got through - any help appreciated!]  *CLI> module load chan_dahdi.so
19:57.14billumUnable to load module chan_dahdi.so
19:57.14billumCommand 'module load chan_dahdi.so' failed.
19:57.14billum*CLI> [Mar 24 01:21:52] WARNING[28561]: bridge.c:218 __ast_bridge_technology_register: A bridge technology of native_dahdi already claims to exist in our world.
19:58.28billumBackground:  I had installed the wrong version of libpri.  I upgraded it, recompiled dahdi, and now asterisk (even from clean install) can't find dahdi
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20:02.04billumany dahdi experts here?  Would be happy to buy some of your time :-)
20:05.25[TK]D-Fender[15:57]billum*CLI> [Mar 24 01:21:52] WARNING[28561]: bridge.c:218 __ast_bridge_technology_register: A bridge technology of native_dahdi already claims to exist in our world. <- this seems to imply that it is already loaded, and we didn't see you check for that first
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20:05.39[TK]D-Fenderand PASTEEBIN instead of flooding to the channel
20:05.42sekilWIMPy: there?
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20:06.08[TK]D-Fender~pb
20:06.08infobotA "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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23:31.12volga629Hello Everyone, I am trying set asterisk act as standalone voicemail server
23:31.39volga629but always get reject '120@voicemailbox' rejected because extension not found in context 'voicemailbox'.
23:32.32[TK]D-Fendera very clear dialplan error
23:32.49[TK]D-Fendermeans exactly what it says
23:33.11[TK]D-Fender[voicemailbox] has no extension to match "120@voicemailbox"
23:36.08volga629I have something like this http://fpaste.org/87929/95617731/
23:39.59WIMPywould klick that link if he had enough space on his desk to move the mouse over it.
23:40.36[TK]D-Fendervolga629: that error means exactly what it says
23:40.48[TK]D-Fenderyou have no match for is being sent in
23:41.04volga629yes, I am correcting dial plan right now
23:41.20[TK]D-FenderVoicemail(su${EXTEN}) <- and this is ANCIENT Asterisk 1.0 syntax
23:41.38[TK]D-FenderIt was still valid in 1.2 and dead 1.4+
23:41.55[TK]D-Fenderexten => *97,1,VoicemailMain(${CALLERIDNUM}) <- also very wrong
23:42.32[TK]D-FenderDid you just blindly copy this off the voip0info wiki or something?
23:42.52WIMPyvolga629: what version are you using?
23:43.20volga629I am using very old box
23:44.32volga629I got kamailio box running only voicemail not checking properly, I am putting normal way dial plan and test it in sec
23:54.43volga629yes, it working
23:54.48volga629I putted exten => _1XX,1,Answer
23:54.50volga629exten => _1XX,n,Wait(1)
23:55.10[TK]D-Fendervolga629: what version are you even running on that box?
23:55.39[TK]D-Fendervolga629: and that STILL won't match

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