IRC log for #asterisk on 20151111

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09:49.27utrackHowdy! Is there a way to make CDR count sanely when making a call with Originate?
09:50.16utrackAt the moment it counts everything as BillableSeconds (i.e. when the 1st side picks up the call BillableSeconds counter starts)
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13:16.00talntidI don't know how to propose changes to the asterisk.org wiki, but....
13:16.02talntidhttps://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints
13:16.15talntidthat page shows the syntax being exten = XXXX
13:16.20talntidinstead of exten => XXXX
13:18.19newtonrtalntid, simply comment on the page
13:18.28newtonrtalntid, or for big changes file an issue on issues.asterisk.org/jira
13:18.47newtonrtalntid, both syntax are valid
13:19.20newtonrtalntid, newer examples on the wiki will use = instead of => as the > is unnecessary.
13:22.58[TK]D-FenderI like my arrows.  Feels classier
13:25.06newtonr:D
13:43.34talntidahh, i see. yes, i like the arrows too :P
13:44.17talntidgot my * box all integrated tonight... it emits events over XMPP, line presence over XMPP, etc...
13:44.33talntidain't i speshul...
13:44.58talntidAMI events, that is
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14:03.51stephanjnewtonr, [TK]D-Fender: i am automatically generating asterisk sip.conf, voicemail.conf and extensions.conf for a project... i spend 2 hours looking on how to *cleanly* hack the ini writer to use '=>'. a day later i read the proper docs and they told me that the arrow was not mandatory.. i was a bit annoyed ;)
14:04.21newtonr:D
14:05.08newtonrI'm sure you'll use those skills you enhanced in the future!
14:05.48stephanji still have neither read the complete docs or the book
14:06.03newtonrThey are helpful!
14:06.21stephanjprobably!
14:07.30newtonrstephanj, So why are you using chan_sip instead of chan_pjsip?
14:08.25stephanjeasy because i still run debian jessie with asterisk 11
14:08.42stephanjand chan_pjsip is a invention from asterisk 13?
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14:10.36newtonrYes the pjsip resources and channel driver were added in 12.
14:11.17stephanji tested 13 briefly when messing with TLS in asterisk
14:11.26newtonrWhen you think about moving to 13 you'll want to use chan_pjsip as that is where the development focus is for the future of SIP connectivity in Asterisk.
14:11.32stephanjcool
14:11.51stephanjuntil i found out that me not being able to provide a correct cert file does not varrant keeping a self compiled asterisk in production :)
14:12.39stephanjwe use a config writer which pulls stuff out of ldap and into flatfiles.. so supporting a new backend is quite easy
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14:14.59newtonrstephanj, :) cool.
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15:05.20Phil-WorkI'm trying to automatically originate 2 outbound calls and conference them together on an inbound call... to create a 3 way call
15:05.36Phil-Workpresume I need to use ConfBridge but I'm not entirely sure how to implement it
15:05.59Phil-Workif I Answer(), ConfBridge() then Dial()... the Dial() isn't executed
15:09.53[TK]D-FenderYou don't dial
15:10.23[TK]D-FenderWhat is the starting trigger for this process?
15:11.40Phil-Work[TK]D-Fender, call coming into exten 11111 on the "alarm" context
15:12.15Phil-Workthat should simultaniously dial 2 numbers on a SIP trunk and conference them all together
15:12.59Phil-Work*simultaniously
15:14.01[TK]D-Fenderthat's a 2-way call
15:14.41[TK]D-FenderSo in this case I'd just directly Dial() and redirect the 1st to the second after answer and confirmation
15:14.56Phil-Workthere's 3 parties - the original caller and then the two that are dialled
15:15.22[TK]D-Fenderok, original + 2
15:15.27Phil-Workyes
15:15.51[TK]D-FenderSo in that case trigger ther 2 originates and continue on to dump the caller into the conf immediately awaiting their arrival
15:17.00Phil-Workhow do I originate those calls if not using Dial()
15:17.13Phil-Workoh, Originate()?
15:19.26[TK]D-Fenderuse AMI or call files.  The dialplan app version is blocking
15:19.46Phil-Workhm, ok
15:19.59[TK]D-Fenderor CLI call
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16:17.32craigifythat's a lot of simultaneous ping timeouts
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16:37.41Phil-Workis it possible to get the number of users in a given confbridge as a variable?
16:38.57WIMPyCONFBRIDGE_INFO
16:41.06igcewielingPhil-Work: you should read the output of "core show functions"
16:42.03Phil-Workok, thanks
17:04.29Phil-WorkI have something like this to create a 3 way conference call on an inbound call - https://gist.github.com/phil-lavin/b52ed32395238d3854b7
17:04.41Phil-Workthe NoOp doesn't happen, presumably because the ConfBridge() blocks
17:04.45Phil-Workwhat's the way around this?
17:06.59Phil-WorkI guess I could originate a 4th call to another local channel which handles the NoOp
17:07.35igcewielingPhil-Work: I assume the problem is the truely horrible way you are using Originate
17:07.53igcewielingPhil-Work: "core show application Originate"
17:08.32Phil-Workigcewieling, I've read the manual but not doing it via System() also blocks until the call is answered
17:09.03igcewielingPhil-Work: correct.
17:09.22Phil-Workso what's the non-blocking alternative?
17:09.36igcewielingI normally use .call files or AMI to Originate.
17:10.07Phil-Workthose were amoung [TK]D-Fender's suggestions - I'll look into those as less dirty alternatives
17:10.12Phil-Workthanks
17:10.56igcewielingPhil-Work: using System() like that will cause all sorts of issues.
17:13.58lvlinuxPhil-Work: I have N-way calling setup---is that what you are needing?
17:15.13Phil-Worklvlinux, yes
17:15.14lvlinuxI set it up for an office where the recptionist answers the phone, the caller needs to talk to both the receptionist and the owner at the same time.
17:15.40Phil-Worksimilar concept - this is for an intruder alarm
17:15.57Phil-Workthe alarm dials asterisk over PSTN and Asterisk automatically calls n people and conferences them together
17:16.15lvlinuxI'll put up the config a bit later---have something I have to do now but if you'll be here a while I'll get back with you.
17:16.29lvlinuxThat's not exactly how my thing works but you should be able to adapt it pretty easily.
17:16.44Phil-WorkI'll be online but possibly not here - just highlight me and I'll pick it up when I get home
17:16.52Phil-Workthanks :)
17:17.20lvlinuxk
17:17.38WIMPywonders if the classic conference could somehow be emulated with Bridge().
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17:19.09lvlinuxBridge() doesn't do mixing does it?
17:20.14WIMPyNo, but it allows you to connect to other channels that aren't part of the current call (yet).
17:21.06lvlinuxhmm, interesting
17:46.18Phil-Workam I right in assuming that Asterisk cannot distinguish a call that was legitimately answered and one that went to voicemail when calling, for example, a cell phone over a SIP trunk?
17:47.54[TK]D-FenderIt is answered
17:47.56[TK]D-Fenderone way or the other
17:47.59lvlinuxehh, I _think_ there are methods for determining if it's a real person on the other end, but I don't know how effective they are.
17:48.05[TK]D-FenderWHAT answered ... is a "point of view"
17:48.11[TK]D-Fender"core show application amd" <-----
17:48.14igcewielingPhil-Work: correct.
17:48.56igcewielingPhil-Work: you should read the output of "core show applications" 8-)
17:49.39Phil-Workhm, amd looks interesting
17:49.42Phil-WorkI'll give it a try
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17:51.04igcewielingPhil-Work: Tuning AMD is not easy.  I used to work for a company which considered their AMD settings to be a trade secret. 8-|
17:52.19igcewielingPhil-Work: you could use answer confirmation, do a search for: asterisk answer confirm
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17:58.23Phil-WorkI'll have a play with AMD
17:58.43Phil-Workit'll be the same 2 voicemails that are hit so once it's tuned for those, it should be alright
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19:16.48jmordicaDoes anyone know if homer by sipcapture has an irc channel?
19:16.59jmordicaor does anyone use homer here?
19:36.28rrittgarni'm starting to implement homer in my environment - seems like it will be useful. I do not know about the IRC channel though
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19:37.03rrittgarnQuestion about app_followme in 13 - does the syntax in the file work like queues or sip.conf where you can have templates? Or has anyone played with it in realtime?
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19:55.47dan_jHi. Is phpagi compatible with asterisk 13?
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20:01.41copystringmy asterisk seems to shut down for no apparent reason but the process doesn't get killed. on top of that another instance of asterisk is startet after that and that instance of asterisk is then no longer able to connect to the trunks. the two instances then result in this:  Failed to bind to 0.0.0.0:59361: Address already in use. any ideas why this happens?
20:02.49WIMPyWell, you can't hace two instances running.
20:03.11WIMPyYou need to find out what starts the new instance.
20:03.34copystringyes. i've been trying that. the logs seem to show nothing :\
20:07.45mjordando you have safe_asterisk running?
20:07.58mjordan(granted, it should only fire a new Asterisk instance up when the old one dies...)
20:08.09copystringno. what's that?
20:08.25mjordan"man safe_asterisk"
20:08.54copystringnot sure if that helps since the old process never actually dies ... it just sits there
20:09.01copystringdoing nothing
20:09.12copystring(apart from blocking the port)
20:09.35cmendes0101ps uxa | grep asterisk
20:09.42cmendes0101who owns the other process?
20:09.52copystringboth are owned by root
20:10.30copystringah no. both are owned by asterisk
20:11.11cmendes0101hmm, just kill both and try to run it again
20:11.26cmendes0101or does that always happen?
20:11.45copystringit happens all the time
20:11.55cmendes0101how are you running it?
20:11.59copystringit runs for like 20-30 minutes then boom. tits up
20:12.33cmendes0101oh so you have one process and everything is happy for 20-30?
20:12.46copystringyes, roughly
20:12.59copystringit's running on pfsense via the official package from pfsense. also tried the freebsd package (version 1.8 and version 13)
20:13.21cmendes0101what level are you logging?
20:13.57copystringi've connected to the console of asterisk with log level 80 and waited for it to happen. want a pastebin of it?
20:14.00cmendes0101you can run asterisk with -g and that should dump core if it crashes
20:15.16cmendes0101does the logs in /var/log/asterisk show anything else from what your seeing remote?
20:15.42copystringyes. the log level was only in the console
20:18.32copystringi'll make a new log with log level 80 in the asterisk.conf
20:18.38cmendes0101try running with -c -g. When I have modules having issues the -c helped show that since it wasn't logging for me
20:19.03copystringok. i'll do that
20:20.04cmendes0101just curious what do you mean log level in asterisk.conf?
20:20.14cmendes0101I haven't really seen that before
20:20.29copystringoh. i meant verbosity
20:21.46cmendes0101I think the max on verbose is like 10 but that setting is just for CLI i believe. For changing logging its in logger.conf
20:23.25copystringodd. when i changed verbosity in the asterisk.conf it said in the console the log level was whatever i've set it to
20:24.01mjordanJust as an FYI: max debug is 5; max verbose is 4.
20:24.16cmendes0101oh there we go lol
20:24.20mjordanOnce upon a time some really spammy junk was tossed out at 10 for some reason, but that's been removed for ages.
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20:24.42mjordanYou can always specify higher; it just won't do anything
20:25.08copystringyeah. i thought so. i've just set to 80 because i didn't bother finding out what the max was :)
20:26.12copystringgot it running with -c -g now. just waiting for it to happen again
20:34.33rrittgarnwhen using extconf to load things into real time - how does that work programmatically? I know that some modules have native support for it like sip.conf. I'm trying to get app_followme to work. I am not getting SQL errors on it, so i know its connecting, but when i query for an ID it isn't found
20:35.16rrittgarnI'm not entirely sure that app_followme supports realtime, but i don't know where to start fixing if there is fixing required at the dev level... i'd love to contribute if necessary though
20:36.15rrittgarndisregard - googled better than i did before
20:36.25rrittgarnfail on my part for not finding https://wiki.asterisk.org/wiki/display/AST/Followme+-+Realtime
20:49.07igcewielingjmordica: I use pcapsipdump but it is not designed to handle the scale Homer is designed for.
20:50.38igcewielingcopystring: there is only one verbosity level, it applies to all log files and the CLI.     Make sure Asterisk is not blocking on a DNS request.
20:51.29copystringblocking on a dns request?
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21:10.20igcewielingcopystring: Asterisk can stop processing calls if it is having trouble looking up hostnames for ip addresses configured on the system.
21:10.45igcewielingUsually this happens when the server's internet connection is down.  Even extenstion to extension calls fail.
21:11.44igcewielingcopystring: I recommend running a local caching only name server, point /etc/resolv.conf to it   Update /etc/hosts to have a name for every ip on the system.   Enable dnsmgr in Asterisk.
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21:16.52copystringsounds a lot like that might be it. the dnsmgr is enabled and the resolv.conf is configured to the caching dns already. just the hosts file isn't. i'll give that a try next time it goes weird
21:17.28igcewielingcopystring: the key is to do it before it freaks out.   Many things could cause the problem, DNS is only one.
21:19.43copystringyup. i've just tried something else just now and i'm waiting to see if that already did it.
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21:30.23lvlinuxcopystring: it's a major pain. I think using pjsip fixes it though since I don't believe pjsip DNS lookup is blocking.
21:32.08copystringi thought about using pjsip. does it work with a normal sip trunk and is it any good?
21:33.50copystringhm, asterisk 1.8 doesn't seem to support pjsip
21:37.56filechan_pjsip only exists in 13+
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21:46.29copystringright. i have another log now with the issue and the -c -g parameters. it does a few dnsmgr_lookup and then:-- Remote UNIX connection
21:46.30copystringBeginning asterisk shutdown....
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23:03.42EncryptHum, so boritos isn't registered :(
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