IRC log for #asterisk on 20160701

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02:13.36raspberrypifananyone around im getting an issue [Jun 30 22:09:51] NOTICE[26238][C-00000106]: chan_sip.c:25859 handle_request_invite: Call from 'tech1' (xxx.xxx.xxx.xxx:5060) to extension '09xx463592' rejected because extension not found in context 'stations'.
02:13.43raspberrypifanim trying to dial out of my gateway
02:13.47raspberrypifannot directly to an extension
02:13.51raspberrypifaninside asterisk
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03:21.01wyoung<PROTECTED>
03:23.46raspberrypifanwut
03:24.18wyoungI am here
03:24.41raspberrypifando i know u
03:24.57wyoungprobably not, but you did ask if any one was around
03:25.16wyoungI am assuming you want assistance to an issue?
03:25.17raspberrypifanoh
03:25.20raspberrypifanyup
03:25.23wyoungok I have read it
03:25.33wyounghow are you placing the call?
03:25.40wyoungfrom a SIP phone?
03:25.58raspberrypifanyea
03:28.40wyoungok so what does your stations context look like?  can you paste it in bpaste or hastebin?
03:29.06wyoung(extensions.conf, unless you are using .lua or .ael)
03:30.13raspberrypifansure
03:30.20raspberrypifanone moment stvp
03:35.02wyoungok
03:35.14raspberrypifanhttp://pastebin.com/FErmxVSr
03:37.08wyoungI want to look at your extensions.conf only, not advertisements too
03:38.52raspberrypifandont see no adverts
03:38.54raspberrypifanbut sorry if i did
03:41.48wyoungwhat extension is the top extensions in/
03:41.54wyoungoops, what context even
03:42.14wyoungand in your sip.conf, what context are your SIP phones in?
03:43.08raspberrypifan[outbound]
03:43.15raspberrypifansip.conf they are in stations
03:54.00raspberrypifanapparently the solution was to add a .
04:14.20wyoung?
04:16.04snadgethis is going to be a super dumb question.. but if you have [SomeTrunk] host=dynamic username=test secret=password
04:16.18snadgewhy do i have to register with username SomeTrunk instead of test
04:17.46snadgeim scratching my head wondering why im getting auth failed.. going.. no.. thats the right username.. thats the right password.. wtf.. in this case the username and the trunk name were the same, but the trunk had a capital at the beginning
04:18.01wyoungsnadge: you register with the username, the [SnomTrunk] is what you use to reference it in extensions.conf
04:18.14snadgethats what i thought too
04:18.33snadgethis is sip.conf rather
04:18.39wyoungsnadge: you can also have situtations where you register with one username and password and have a peer entry ([SomeTrunk]) that uses different credentials
04:19.04drmessanousername is used in outbound authentication to another peer
04:19.07wyoungI mean I don't know why you would want to do it but it is possible
04:19.23drmessanoIf you're registering TO Asterisk you use the [peername]
04:20.07snadgeright.. of course that makes sense now.. i clearly need more sleep or something
04:22.45snadgenormally i create separate inbound and outbound trunks.. i knew there was a reason why :P
04:22.53snadgethis one just happens to be used for both.. because lazy
04:23.26snadgeand apparently stupid too
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05:37.32wyoungsnadge: I just setup a friend and stick it in a pretty limited context
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08:46.54UncleKiwithanks for asterisks
08:46.57UncleKiwi:)
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11:12.12cervajs2hi, i have asterisk pjsip trunk - asterisk chan_sip trunk . trunk works but sip options  from chan_sip to pjsip are not authorized. any hint?
11:12.32cervajs2res_pjsip/pjsip_distributor.c:368 log_unidentified_request: Request from
11:18.07cervajs2solved: mistake in identify section
11:19.11cervajs2partially solved. log_unidentified_request are gone. but in sngrep i see that response is still 401 unautorized
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11:23.38[[thufir]]does this article describe just inbound calling, receiving calls from the trunk?  http://support.siptrunk.com/hc/en-us/articles/204848079-SIPTRUNK-com-CONFIGURATION-GUIDE-FOR-ASTERISK
11:36.59[[thufir]]I can't find the syntax, but I recall reading that it's possible to break down extensions.conf into seperate files.  What is that called?
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11:37.52[[thufir]]#include I think
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11:56.30[TK]D-Fender[[thufir]], that just should the SIP pper & register strings.  that covers both directions
11:56.34[TK]D-Fenderpeer*
11:59.11[[thufir]][TK]D-Fender: but the context is from-trunk.  Shouldn't there be a corresponding to-trunk?
11:59.36[TK]D-FenderPeers only have one context
11:59.47[TK]D-FenderWhen you use a peer to call out SIP knows NOTHING about "context"
11:59.52[TK]D-Fenderthat is an * dialplan concept
12:00.10[TK]D-Fendercontext= is where calls FROM the trunk will land
12:00.16[TK]D-Fenderand that name is whatever you want it to be
12:00.45[TK]D-FenderThey are very clearly showing this set up for a FreePBX system
12:01.19[TK]D-FenderAnd that is the context FreePBX uses to handle calls from what they'd consider a "trunk" so that it gets matched against Inbound Routes
12:01.38[[thufir]]ok
12:01.43[TK]D-Fender~book
12:01.43infobotAsterisk: 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
12:01.45[TK]D-Fender^^^
12:02.28[[thufir]]why do you say freepbx, though?  the title has "asterisk".
12:05.29[TK]D-FenderA. because of the specific context they chose.  B. Becuase read the LINKS on the right hand side. FREEPBX MODULE.  ELASTIX.  FREEPBX.
12:06.17[TK]D-FenderAnd they know anyone who knows anything about Doing it in Asterisk for their own manual dialplan will know to set the context to wherever they wanted anyway
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12:09.34[[thufir]]ok
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14:11.10dopplergood morning
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14:16.01dopplerif anyone here has experience with polycom equipment, give me a holler :)
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15:04.43||cwdoppler: just ask the question
15:05.28dopplerwell…
15:05.42dopplerhow well do soundstations typically play with asterisk?
15:06.05doppleri'm thinking of picking up a few 550's or 650's, and from what i can tell, they will *work*
15:06.11dopplerbut the question is, how well? :)
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15:06.49dopplerprimarily interested in polycom since their speakerphones are supposedly excellent
15:07.02doppler(and they don't look like toys)
15:08.16dopplerthey're cheap enough on the used market that i could just buy a few and find out firsthand, but i thought i'd see if some other users had an opinion
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15:12.53dwigton_Does it matter where your asterisk box is in your network topology as long as it has access to the web? Or does it need to be outside of any firewalls etc? Or at least have a 1-1 NAT?
15:14.10dwigton_I am just a bit confused how it can sit on your LAN without opening up a hole.
15:14.56dwigton_Unless asterisk intiates the connection to your SIP provider.
15:15.47shido6eliminate nat if possible - but it can be behind a firewall - test at scale before certifying the service as production ready.
15:16.12shido6you may find your firewall has hidden functionality
15:16.36dwigton_hidden functionality?
15:16.48shido6SIP ALG, SIP DDoS detection, etc.
15:17.49dwigton_we are using pfsense
15:18.09shido6in that case, hidden sip proxy
15:18.43shido6If someone went out of their way to configure the proxy - you may need to disable it -
15:19.42dwigton_I have no idea any of that means. Any SIP traffic on the lan gets handled by pfsense instead of making it out to the isp?
15:20.11shido6its possible but we can discern those things when make test calls and post the debug in pastebin.com and send us the link to review the data
15:21.01dwigton_I am not at that point yet. I am in the reseach phase to make a proposal for replacing our ancient pbx.
15:22.53dwigton_I will try to scrounge up some hardware or a vm for a freepbx install. Can asterisk make connections to the SIP provider that by passes the current appliance the isp provided for our legacy system?
15:23.12shido6yes
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15:23.32shido6do you have a link to the public internet that doesn't use that appliance?
15:23.55dwigton_No, I don't think so.
15:25.04dwigton_Does the test need to be done on their fiber or can we connect from another location?
15:26.03dwigton_Also I need to verify that they actually are providing sip. Do we just ask them for credentials for our 5 lines?
15:26.54dwigton_^probably an idiot question but no idea how this really works.
15:27.00shido6you can ask them , sure
15:27.11shido6find out what you are paying
15:27.30shido6look at your bills for the last 6 months -
15:27.59dwigton_Ah looking for SIP charge?
15:28.26dwigton_I will have to hunt the real admin down and see if he will let me look at those.
15:28.45shido6phone bill - telecom charges, lines, phone numbers, you can get on the phone your provider if you have questions -
15:29.02shido6with with with with - whats wrong with my fingers today with with with
15:29.24shido6let you?
15:30.12shido6Speak with an intelligent tone and inquire
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15:54.22albatrossshi all
15:56.15albatrosssis it possible to call a global variable by adding channel variable
16:01.14albatrosssi have a global variable ----> system_day_on=0900 and system_day_off=1500, i can call this in the dialplan no problem, i however want to call it changing the day why won't this work $[${system_(${day})_open}]
16:02.21albatrosssforgot to mention i have multible global variable with a different day
16:03.51WIMPySo you have a system_monday_on etc?
16:04.22albatrosssyes
16:04.34WIMPyBut you don't have variables called system_(monday)_on?
16:04.54albatrosssi do
16:05.05albatrosssfor every day of the week
16:05.25WIMPyYou have both names?
16:05.51albatrosss?
16:06.27WIMPyHint: With or without brackets.
16:07.33albatrosssglobal var looks like this system_Friday_open=0900
16:07.37albatrosssno brackets
16:08.17WIMPySo you won't find a value if you're looking for a varaibale wit brackets in it's name.
16:08.45albatrosssi get the day using STRFTIME and want to pass that too the global variable
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16:12.02albatrosssset(foo="Friday) then call global  NoOp(${system_${foo}_open})
16:12.15albatrosssi think i lost the plot this also does not work
16:12.33TandyUK2anyone aware of an ATA with 4 analog ports (for 4 seperate accounts)
16:12.49WIMPy"Friday" or "friday"?
16:12.51albatrosssgrandstream
16:13.13albatrosssFriday
16:14.11TandyUK2HT704? you actually used one?
16:14.18WIMPyAnd die you set foo=Friday or foo="Friday"?
16:14.35Penguinset(foo="Friday)
16:14.38Penguin"Friday
16:14.42WIMPyBuilding a similar name doesn't do anything. It has to be exact.
16:15.10albatrossslol sorry gents been staring at this screen the whole day missed that one
16:15.27albatrosssmy code does have both ""
16:15.56WIMPyRemove them.
16:15.59PenguinI suggest that you don't use quotation marks in your set.
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16:16.38albatrosssthx will give it a go
16:20.38albatrosssyou guys rock, working thank you so much
16:25.49TandyUK2anyoen know of a good voip handset for use in very dirty (dusty) environments
16:26.12TandyUK2it would be wall mounted, where they cut stone blocks
16:26.17TandyUK2so insanely dusty in there :P
16:44.16albatrosssTandyUK2:why don't you slap down a ata, analog handsets are more robust the voip handsets, also much cheaper to replace
16:45.57Samot^ Probably the best option.
16:46.35SamotDon't want to spend $100 on a phone to have it destroyed by brick dust getting inside it or shrapnel hitting it.
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16:48.03PenguinI would suggest a $10-$15 basic bell telephone installed on the wall and an ATA placed in a cleaner location with a regular phone cord connecting the two.
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17:00.49doppleri second this
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18:53.46albatrosssis there a way that you can change a quember possition on the fly?
18:54.01albatrosssthat is queuemember sorry
18:57.53fileno.
19:10.49albatrosssthx
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20:05.55albatrosssif you force a queue member to pause whiles agent is on a call will you disconnect the call the user is busy with?
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21:06.07WIMPyHmm. Does Asterisk accept both = and == in comparisons?
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21:30.54TandyUKWIMPy: if its like other lanbuages, = sets something, == compares
21:35.14WIMPyIt's not a programming language :-)
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21:47.53brettnemhey all, I'm calling an AGI script which drops a user into a conference room. But what I really want to do is have the user basically on hold while my AGI script loops, making REST calls until a condition is met, then redirect the call. Because the callout to ConfBridge is blocking, I can't really check my REST call. What's a good way to do this?
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21:49.55brettnemI tried using AMI from my agi to redirect the call into a conference room and continue in my AGI, but when I redirect, the AGI is terminated. I've tried various combinations of signal trapping, and AGIEXITONHANGUP and AGISIGHUP but I can't seem to figure it out
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21:51.50WIMPyBecause thee redirect action redirects the call away from the AGI application.
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21:56.18robmalHey, anyone familiar with cisco 79xx? I've almost got them to do their work but DnD still works not the way i'd want. When enabled all it does it mutes the ringing, but i'd like it to i.e. go to voicemail. Cisco says it can be done when call forwart no answer is set but i can't find any way to set it via xml configs. Any clues?
22:04.42brettnemWIMPy: yeah gathered that. I'm looking for a better way to do this. Any ideas?
22:05.38WIMPyDunno what you're up to.
22:07.01brettnemWIMPy: well just like I said above. I want to be able to do blocking things, like have the call wait in a conference room until an event happens then redirect it away. So I want to be able to park the call for some time, then later come back and redirect it.. Problem is with straight agi, once I drop into a conference I can't do anything else in the channel because it's blocking
22:08.38WIMPyYes, that's the way the dialplan works.
22:09.25brettnemWIMPy: I'm aware that's how it works. I'm asking if anyone has any ideas on how to do what I'm trying to do here
22:09.26WIMPyBut I still don't konw *when* you want to do something. But maybe you want to use more AMI? Or even ARI?
22:10.09brettnemWIMPy: I want to drop the call into a conference room. Then every 10 seconds I'm going to call to a REST endpoint. it will tell me "wait" or "go" when it says "go" I want to redirect the channel
22:11.02WIMPyThen don't do it inside your AGI. Fork another process.
22:11.40WIMPyOr write an application that stays connected to AMI.
22:12.30brettnemWIMPy: yeah my next attempt was a fork. Just wondering if there was a cleaner asterisk way to do it while maintaining the local channel connection.
22:12.57WIMPyAMI/ARI
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22:13.28brettnemI see some reference to async AGI whichj looks appealing but I'm failing to understand how using it is even possible
22:13.46brettnemonce you call async:agi you are blocked and can't run a background app that performs the AMI actions on it.
22:14.31WIMPyEither fork or don't use AGI.
22:15.45brettnemok thanks
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