IRC log for #asterisk on 20140517

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01:47.50fidividihi all, what is the best way to troubleshoot one-way audio?
01:48.51fidividiwhen i call out or receive a call, sometimes, randomly, the other side doesn't hear me. I checked my recorded call on PBX, the pbx also can not hear me.. what could it be?
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03:01.46ChannelZfidvid: first make an echo test extension and make sure you have bidirectional audio between your device and asterisk.
03:02.20ChannelZIf that works, then either your packets aren't making it off your network, aren't being sent to the right place, or are being blocked by
03:02.32ChannelZoops.. blocked by the remote end
03:03.52ChannelZBut it all depends on your setup.  Are you local to the asterisk box?  (IE same LAN)  Is it behind a firewall?  Are you?
03:08.00fidividiChannelZ: Asterisk is on public IP on internet
03:08.24fidividiChannelZ: all devices/peers/trunks connect from different remote locations
03:13.27fidividiChannelZ: echo test will be fine, i have tried calling within the extensions, all good... the problem occurs randomly
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08:20.53lok-jpra2billing cdrs are not logging into database..... using a2billing v2 with asterisk 1.4 /etc/a2biiling.conf file is with all the database connection.
08:21.02lok-jprplz suggest something
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08:25.33ChannelZHave some bacon
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12:39.09neeby_gooseyIs it possible to run applications from cli?
12:41.07WIMPyNot directly, no.
12:49.38neeby_gooseyWhat about call file? I can run app by moving "Application: <app>\n Data: <arg1>,<arg2"
12:49.45neeby_goosey, right?
12:50.00neeby_goosey(moving to asterisk spool dir)
12:52.37WIMPyYes
12:52.49WIMPyOr ther other ways to originate calls.
12:53.15WIMPyBut you always need a channel.
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13:33.23neeby_gooseyI couldn't make asterisk run an app, using call-file, because it tries to call a device under "Channel:"  before running the app.
13:33.49WIMPy>>But you always need a channel.
13:34.05WIMPyThat channel could be a local channel however.
13:34.38neeby_goosey"Channel: SIP/dongle\n Application: DongleSendSMS\n Data: dongle,89001001234,helloworld\n Archive: yes"
13:35.22neeby_gooseyLocal, you say? So I could just autoanswer... that would do, thanks.
13:41.55neeby_gooseyYes, "Channel: Local/s@autoanswer" part did the trick.
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14:43.43inebcan i use variables for extension names? f.eks. exten => {$ack_key},1,Wait(1) ?
14:44.58WIMPyYes, but only if you use the correct syntax.
14:45.17WIMPyAnd off course the variables are only read when the dialplan is loaded.
14:45.34inebcool, what is the correct syntax?
14:45.58WIMPy${name}
14:46.13ineboh, of course. ty
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15:05.50neeby_gooseyWhat does the 'group' option in dongle.conf mean?
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15:36.10inebcan i switch the contex in an GotoIf Statement somehow?
15:37.15inebi have something like this http://pastebin.com/8G8QRLxW
15:37.46inebbut the it does not work, asterisk complains, that there is no ack_finished_good label in 'a'-context.
15:37.57ineb*the gotoif does not work
15:38.02WIMPyThe destination can be anything, just as in a normal Goto.
15:39.18inebwhat should i write to make it work?
15:40.34WIMPy[[context,]exten,]priority|lable
15:43.25inebi see
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15:49.35inebWIMPy: yes, now it is working
15:49.41inebthank you. you are very helpful
15:50.06inebone step closer to the ultimative goal
15:50.41[TK]D-Fenderineb: pastebin it.....
15:50.44WIMPyWorld domination?
15:50.46[TK]D-Fenderwith the output....
15:52.10inebi would say ascension
15:52.18[TK]D-FenderI doubt it's actually working as expected....
15:53.40inebit looks like this now http://pastebin.com/44NuM8Hy
15:54.16[TK]D-Fenderineb: show the call...
15:54.23WIMPyalso doubts that works.
15:54.25[TK]D-Fenderineb: because that is not right...
15:54.40WIMPyI miss a $[]
15:54.42[TK]D-Fender^^^^^
15:55.26[TK]D-Fender$[] <- what you put in here is an expression.  Nothing else is.
15:57.51inebi see. strange that it worked anyway
15:58.00inebi tested both cases, true and false
15:58.47inebno. wait. it doesnt
15:58.57inebit jumps into it, no matter if true or false
16:00.09[TK]D-Fenderineb: with ${} that is just dumb text before the ?, and basically it's either 0 (false), or anythingelse (true)
16:00.14[TK]D-Fenderso fred = tru.
16:00.17[TK]D-Fendertrue
16:00.30[TK]D-Fender"A" ="B" is TRUE
16:00.40[TK]D-Fender$[] rather
16:02.02ineblogic behaviour
16:02.31inebanything but false should be true
16:02.40[TK]D-Fendercorrect
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16:46.16filefalls into existence
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20:00.12Kobazanyone run asterisk on mips64?
20:04.41wdoekes~ask ;)
20:04.41infobotNope, wdoekes! I won't ask ";)"
20:05.04wdoekes~poll
20:05.04infobotScript for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html
20:05.18wdoekesnever mind that
20:06.57Juggiefidonet ?
20:06.59Juggiewhat is this 1999
20:07.05Juggiedid i step into a time machine?
20:17.43Kobazfidonet haha
20:21.19WIMPyI guess you meant
20:21.24WIMPy~polls
20:21.24infobot"Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask>
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20:35.02ChannelZasterisk on irix?
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20:35.17KobazChannelZ: no mips64
20:35.28Kobazlittle 500mhz dual core router
20:43.40wdoekesbut your question probably isn't "is anyone using it"
20:44.59wdoekesbut, "does someone on mips64 have issues with x?" or "should I attempt to compile asterisk on mips64 or will I burn a lot of time without results?"
20:46.13wdoekesI can tell you that the sparc has unresolved issues. ASTERISK-21665
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21:08.35Kobazif anyone here was using it, it would lead to the followup of 'are there any issues you're run into'
21:08.48Kobazyou've..
21:08.49Kobazarg
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22:25.48dan_jHi, I'm looking at load balancing and redundancy and at the moment I'm looking at using Opensips. The question is, if all traffic to the asterisk servers is coming from the opensips IP, how do you debug on the asterisk server?
22:26.00dan_jIs there a better way to achieve redundancy?
22:26.55WIMPyYou already tried it with Asterisk itself, didn't you?
22:28.11dan_jHow would you failover with Asterisk itself? Using Heartbeat?
22:28.39WIMPyWhat's that good for?
22:29.20WIMPyIn the simples way by using DNS RR, otherwise the usual loadbalancing techniques.
22:29.26dan_jFrom what I understand, giving a virtual ip which switches over to the 2nd server is the first goes down. Although I dont think its the best solution
22:29.54WIMPyAs long as you only need one server.
22:30.07dan_jThe problem is DNS RR is that youre relying on the sip peer to realise there is an issue and switch.
22:30.20WIMPyTrue
22:30.44dan_jif you have opensips in front of the asterisk servers, it can deal with the failover and the endpoint doesnt know the difference.
22:31.02WIMPyBut who deals with opensips failing?
22:31.27dan_jBut from what i understand, if I do that, then asterisk will show all sip peers registered from the IP address of the opensips server.
22:31.49WIMPyI guess so, yes.
22:32.05dan_jYou can set up two opensips servers and a mysql db to join them. they can use a virtual IP or something to ensure one is always online.
22:33.20WIMPyYou can do that without opensips.
22:34.34dan_jYou can't load balance without opensips. Well, you can using DNS, but if you want intellegent load balancing, you need something else.
22:34.57dan_jEg. If some peers need to participate in the same queues, they all need to register on the same asterisk server.
22:35.03WIMPyOff couse you can.
22:35.26WIMPyBut I have no idea, which way would work best.
22:35.54dan_jOk, Looks like i'm going to have to play around a bit.
22:36.15dan_jI don't really want to lose the ability to seeing the peer IP addresses in the asterisk cli
22:38.51dan_jOk. Thanks for the discussion. Got things to think about.
22:39.23WIMPyYou should try to ask again during US office hours.
22:44.29dan_jOk, thanks.
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