IRC log for #asterisk on 20191002

00:19.18*** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net)
00:19.18*** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.1 (2019/09/05) 16.5.1 (2019/09/05), Security Only: 15.7.4 (2019/09/05); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22
00:22.10*** join/#asterisk DannyA (~DannyA@2601:14d:4380:2360:68a4:7e1c:b8f2:b750)
00:22.42DannyAhey all.  is there a way to programmatically park a call that's in progress?
00:50.20*** join/#asterisk mindthelion (techquila@gateway/vpn/protonvpn/techquila)
01:14.58*** join/#asterisk tehgooch (tehgooch@unaffiliated/tehgooch)
01:33.38*** join/#asterisk ircarcs (~quassel@169.9.159.77.rev.sfr.net)
01:35.41*** join/#asterisk mmx870 (~mmx@ptr1.cyberia.es)
01:55.09*** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772)
02:00.00*** join/#asterisk saint__ (~saint_@unaffiliated/saint-/x-0540772)
02:41.39*** join/#asterisk Katty (uid62315@gateway/web/irccloud.com/x-ejkmwiyczlxijnsj)
02:56.55*** join/#asterisk cp (~cp@b157153.ppp.asahi-net.or.jp)
03:20.00*** join/#asterisk overyander (~overyande@209.141.208.197)
03:35.21*** join/#asterisk sa02irc (~sa02irc@155-079-043-212.ip-addr.inexio.net)
04:40.15*** join/#asterisk Someone_Else (~Someone_E@14-230-144-85.ftth.glasoperator.nl)
04:40.56*** join/#asterisk mmx870 (~mmx@ptr1.cyberia.es)
04:45.38*** join/#asterisk mbecroft (~user@ak2.becroft.co.nz)
04:46.31*** join/#asterisk bipolar (~bipolar@offsite.guru)
04:51.34*** join/#asterisk Typhon (~Typhon@ipservice-092-211-210-208.092.211.pools.vodafone-ip.de)
04:54.36*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
05:17.33*** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca)
05:22.26*** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca)
05:43.00*** join/#asterisk sysgrammer (~sysgramme@d205-234-50-141.yt.northwestel.net)
06:22.25*** join/#asterisk jjrh (~weechat12@2607:f0b0:7:8170:216:3eff:fefe:444f)
06:33.31*** join/#asterisk jkroon (~jkroon@165.16.203.58)
06:53.45*** join/#asterisk pabe (~pabe@81.24.66.208)
07:03.08*** join/#asterisk opti (~opti@gen-180-214-68-80.ptr4.otw.net.au)
07:10.19*** join/#asterisk scgm11_ (~scgm11@27.red-213-4-21.staticip.rima-tde.net)
07:21.57optiwhat are the config options to explicitly register every x seconds, or is asterisk forced to register 15 seconds before default expiry?
07:22.14*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
07:22.14optidefaultexpiry seems to increase both
07:23.27optiasterisk as the client to a voip provider for clarity
08:11.05*** join/#asterisk scgm11_ (~scgm11@147.204.224.35.bc.googleusercontent.com)
08:15.26*** join/#asterisk scgm11__ (~scgm11@27.red-213-4-21.staticip.rima-tde.net)
08:30.05*** join/#asterisk mattchis (~mattchis@c-107-2-189-89.hsd1.co.comcast.net)
08:41.27*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
08:48.17*** join/#asterisk hehol (~hehol@gatekeeper.loca.net)
08:59.04*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
09:03.04*** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@41.140.26.30)
09:31.14*** join/#asterisk zapata (~zapata@2a02:1748:f71:380:a4e2:e806:32a9:977d)
09:31.28*** join/#asterisk sekil (~sekil@nat-73.net011.net)
09:45.10*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
10:04.43*** join/#asterisk jetlag (~jetlag@c-71-226-222-56.hsd1.nj.comcast.net)
10:07.22*** join/#asterisk pchero_work (~pchero@87.213.247.82)
10:19.49*** join/#asterisk ganbold (~ganbold@66.85.186.234)
10:40.38*** join/#asterisk m4rcu5 (nobody@84.106.248.133)
10:57.01*** join/#asterisk ketas (~ketas@0011-0000-0000-0000-d7dc-830e-07d0-2001.dyn.estpak.ee)
11:09.11*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
11:14.25*** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net)
11:17.39*** join/#asterisk emsjessec (~emsjessec@pool-173-54-255-231.nwrknj.fios.verizon.net)
11:19.35*** join/#asterisk ketas (~ketas@0011-0000-0000-0000-d7dc-830e-07d0-2001.dyn.estpak.ee)
11:20.43*** join/#asterisk _0x5eb_ (~seb@seb-hpws2.elen.ucl.ac.be)
11:37.14*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
12:03.17*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
12:12.35*** join/#asterisk Guest0193 (~martin@185.32.9.250)
12:13.24*** join/#asterisk dacod (~dacod@201.47.74.146)
12:14.23Guest0193Hello, when I initiate a call via ARI I push the call through my custom context (This happens before the dial). In this context I set a CDR variable. However this never makes it to the cdr table. So I did some testing and when doing core show channel SIP/x and SIP/y I found that only one of the channels had this variable and that must be the cause. How can I make sure that both channels have this
12:14.29Guest0193variable? Do I need to set it when the call is answered?
12:18.34Guest0193Here is the CLI output https://pastebin.com/raw/7jxjp0J8
12:41.54*** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou)
12:44.29*** join/#asterisk sekil (~sekil@nat-73.net011.net)
12:53.44*** join/#asterisk DannyA (~DannyA@2601:14d:4380:2360:54b1:c801:114b:8650)
12:54.04DannyAhey all.  is it possible to use one of the various Asterisk API's to park an existing call?
12:54.27SamotYes.
12:55.28DannyA@Samot awesome.  my idea is to build an on-screen interface in my software system to control the phones, since my organization will be using analog phones
12:55.52DannyAwhich method is best suited for that? or can any/all of them do it? and can you point me in the right direction for what command that would be?
12:55.59SamotYou should probably hire someone for that.
12:56.09SamotOr look into something that prexisting.
12:56.43DannyAok.  im interested in learning the programming to do it myself (if feasible).  i do all the software development for our company.
12:56.59DannyAso was just hoping to get the starting point and i'd attempt to learn from there
12:57.24SamotDo you have any experience with Asterisk?
12:57.47SamotAn understanding of the features that you can use like AMI or ARI, etc?
12:58.22DannyAa little bit.  no, haven't used AMI or ARI yet, but definitely eager to learn.  spent the weekend reading Asterisk: The Definitive Guide
12:58.34DannyAon a mission to learn as much as i can and become a solid asterisk developer
12:58.37DannyAso, gotta start somewhere
12:58.47SamotYes and that would be with the basics.
12:59.03DannyAplanning to install asterisk on an AWS server and start with a couple IP phones and begin practicing from there
12:59.05SamotNot going all out with a PBX and GUI interface to do things that you don't fully even understand.
12:59.42DannyAdefinitely not planning that from the outset.  part of my plan for learning was to take real-world scenarios and attempt to accomplish them
13:00.08DannyAhence, wanting to learn AMI ARI etc, i picked a scenario that i thought of which was programmatic call parking
13:00.34SamotSo why are you sticking with analog phones?
13:01.04DannyAour users are very set on their way of doing things, and don't want to switch away from the phones they've always used
13:01.16SamotOK so they have a GUI they use now?
13:01.26DannyAand unfortunately since they are all volunteers, it's a delicate balance between ordering them to do things and them just walking away
13:01.27DannyAyes
13:01.44DannyAi built a whole application that they use for various dispatch-related functions
13:01.53SamotBut do they use it to manage their calls?
13:02.00SamotPark them? Hold them? Transfer them?
13:02.19SamotOr wait, they all have analog lines so they don't do that now.
13:02.38SamotSo this whole "set in their ways" thing is BS. They'll have to learn new stuff no matter what.
13:03.01DannyAno.  since we don't do any of that today.  but one of my ideas for the new system is that if they need to get a supervisor to talk to a caller, right now theres nothing they can do, but in my ideal scenario, (something i was asking about the other day), the user could click and park their call, and a supervisor could call in from the outside and pick up the parked call
13:03.27SamotSo you want an external caller to be able to pick up a parked call?
13:03.31DannyAyes
13:03.38DannyAthat's doable, right?
13:03.57SamotOh sure as long as you know how to to expose other external calls to your internal acess.
13:03.59SamotOh sure as long as you know how to to expose other external calls to your internal access.
13:04.10Samot+not
13:04.22SamotThere are considerations for that type of access.
13:05.03DannyAof course.  my plan was to lock it down with access codes (DTMF), etc
13:06.03SamotWell I hope you have a nice budget for this.
13:06.11DannyAwhy?
13:06.26SamotWell, you've spent almost 4 hours plus already in here asking these questions.
13:06.32SamotThat's a cost.
13:06.54DannyAi work for my organization for free
13:07.03SamotOK.
13:07.12DannyA(and also, are u saying im asking too many questions here? if so, i can tone it down)
13:07.24SamotIt has nothing to do with the amount of questions.
13:07.30SamotIt has to do with the fact you don't know the basics.
13:07.43SamotYet you're going to write a call management GUI.
13:08.48DannyAthats my eventual plan.  im not saying im going to write it one week after reading a book.  this is a process.  doesnt everyone need to start somewhere?
13:09.08SamotOK so this isn't a project that needs to be done anytime soon?
13:09.18DannyA@Samot i promise you im not delusional or arrogant.  i just sincerely want to learn, do it the right way, as long as it takes.
13:09.36SamotAgain, this is a project that doesn't need to be done anytime soon?
13:10.09SamotThere is a difference between wanting to learn and just messing around and wanting to learn but holding up or hurting an active project because of it.
13:10.10DannyA6 months to a year
13:14.17DannyAalso, if i can get approval to go with asterisk, my plan is to hire a company i already found in NYC (referred by Digium) to do the implementation and whatever programming is necessary
13:14.17SamotWell I'd get hoping on this because that 6 months to a year will go faster than you think.
13:14.26DannyAso they will do all the heavy lifting
13:14.35DannyAi just want to one day be able to contribute features to the system myself
13:14.39DannyAand not always rely on a 3rd party
13:16.03*** join/#asterisk scgm11_ (~scgm11@27.red-213-4-21.staticip.rima-tde.net)
13:16.52Samot8:56:01 AM <Samot> You should probably hire someone for that. <-- So we spent 20 minutes going on about how you wanted to learn this and do it yourself with you to end with "We're hiring someone".
13:17.19DannyAyes, i want to be able to implement any number of features on my own at some point
13:17.20SamotWhen y ou could have just said that last part when I asked the question. "Yes, We're going to hire a company"
13:17.38DannyAbut im not going to be stupid enough to think i can do the entire initial implementation myse
13:17.51DannyAi apologize that i didnt state that from the outset
13:17.55DannyAmy mistake
13:18.37DannyAi was simply eager to talk about some of the features that ive thought of and would love to implement at some point myself without having to always go back to this company and pay them to do it
13:19.33SamotYou're putting the cart in front of the horse.
13:19.55SamotYou have no clue what this company is going to offer up as a solution.
13:22.08DannyAur right.  i dont.  im just overly enthusiastic. that's all.
13:22.16DannyAnot trying to waste your time.  sorry if it feels that way.
13:23.02SamotYou're not wasting my time.
13:23.13SamotI'm trying to stop you from wasting yours.
13:24.18SamotYour first step is to have a plan. Know your call flow and logic.
13:24.31SamotThe company you hire will need those details.
13:24.38DannyAalready did that
13:26.51Guest0193I have a fun one, how can I get a variable from the dialplan if I do not know the variable name
13:27.05SamotHow would that variable have been set?
13:28.25Guest0193As a payload parameter in an ARI request. Here is the dumpchan => https://pastebin.com/raw/Wb8kYrmj
13:28.33Guest0193As one can see there are variables there, at the bottom.
13:28.40SamotYes.
13:28.40FinboySlickDannyA: I'm a bit late to the discussion (an by no means an asterisk expert) but for having started with a similar-scale project myself, I suggest you grab cheap (but compatible) analog interface(s) and start implementing a basic skeleton yourself.  Once you'll have implemented a basic dialplan and thrown some calls around, you'll have a much better idea of the scale of your project.
13:28.45SamotThey are var1 and var2
13:29.00Guest0193Could be named qwerty and asdf as well, I dont know.
13:29.05SamotThey are var1 and var2
13:29.24Samotvar1=data1 <-- var1 has the value of data1
13:29.53Guest0193What I am trying to say is that "var1" may or may not exists. I do not know the property name, this is just for show
13:30.05SamotIs that an actual channel dump?
13:30.09Guest0193Yes
13:30.15SamotThen that is the name of the variable.
13:30.15Samotvar1
13:30.26Guest0193Samot: are you messing with me?
13:30.29SamotIt exists with the value data1
13:30.30SamotNo.
13:30.53SamotThe channel dump is showing your all the channel settings and variables that exist _on the channel_
13:31.11Samots/your/you/
13:31.14Guest0193Sure, but how do I extract those variables so I can do operations on them
13:31.24SamotYou would do it like any other variable.
13:31.36Samot${var1}
13:31.52Guest0193The variable name can be anything, I do not know that beforehand.
13:32.01SamotNo.
13:32.15SamotYou are setting that either on the peer/endpoint or in the dialplan.
13:32.22SamotIt's being declared and set by something.
13:32.59Guest0193Yes, and it wont be declared by me. But an external service. And I will not know the names of the variables. Lets say it is random every time.
13:33.10SamotWhat external service?
13:33.30Guest0193Is that relevant, how?
13:33.35fileI do not believe there is any mechanism to query such things. Variable names are expected to be known, as that is how they are given meaning and known to be used.
13:33.50Guest0193Thanks file, what I wanted to hear really.
13:33.52SamotIt's relevant as to know how that is set and being used.
13:33.56SamotWhat?
13:34.37Guest0193Now the second question is, as I do not know the variables names. Can I send all variables? Like sending the output of ChanDump for instance
13:39.39SamotI don't think there is an AMI event for a channeldump
13:40.21SamotYou can probably use ARI to get the channel details but you'll need to know the channel.
13:40.46fileI don't remember if that gives the variables
13:40.58SamotI don't know either.
13:41.02Guest0193the ARI channel variable request requires you to know the name of the variable
13:41.23SamotHow do you not know what variables are being set on your channels?
13:41.40SamotHow do you even know what to do with them?
13:41.44SamotWhat to look for?
13:41.59*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
13:43.48Guest0193That is the thing, I do not know what to look for. I only know the variables are there, with unknown names. So what I can do is call an external script that excecutes chandump on the channel and send back the output to be script that parses it
13:44.03Guest0193That should do it... not pretty but it will work
13:44.13SamotI think you're missing the main thing here...
13:44.27Guest0193Enlighten me good sir
13:44.29*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
13:44.33SamotWhat is setting the variables?
13:44.42SamotWhy is it setting these variables?
13:45.30SamotWhy don't you know the variables being sent since this is your system and you would need to know what they are to use them?
13:47.58Guest01931) An external service 2) To assosciate a call with some meta-data 3) because the meta-data is created on-the-fly and are always dynamic
13:48.43SamotThe data can be dynamic but why would the variables that hold it be?
13:49.07SamotI'm just not sure what you're supposed to do with random variable names.
13:49.17Guest0193That requires a couple of beers, a small town bar to explain. And I am guessing you are nowhere close to me
13:49.39SamotWell you're going to have to do some messy stuff for this
13:49.59SamotAnd why you have no control of what data is being sent to you for you to process is a bit confusing.
13:55.03Guest0193If I had a penny for every time someone said an integration or system was confusing I wouldn't sit here today. I cannot control this, I am just the brute here to solve it.
13:56.12SamotHow it is being done is confusing
13:56.30*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
13:56.41SamotIve integrated plenty systems. In all cases the data schema was known.
13:56.58Guest0193This is my first where it is unknown
13:57.17Guest0193And one day we are all going to die, and it will be a first for all of us
13:58.36SamotThen you'll need a way to get the channel dump so you can parse the variable section and get the var=value pairs.
14:06.47Guest0193can I do dumpchan from the CLI?
14:06.52Guest0193Or is it dialplan exclusive?
14:08.12*** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-kdhfewlpziqizvlh)
14:08.12*** mode/#asterisk [+o bford] by ChanServ
14:08.25Guest0193I cannot find it under core show applications
14:08.57Guest0193ah sorry I can, but not sure how I can call it from the CLI, feeding it with a channel
14:09.13SamotYou can't.
14:09.28Guest0193ehm.. it does not support a channel argument
14:10.05Guest0193What a bummer. The output of core show channel is unparseable if the variable name is unknown
14:10.26SamotYou can do "core show channel <channel>"
14:11.33SamotBut it's a different format.
14:12.58SamotSo basically you'll need an AMI listener so you can get the new channel id when it's created and then issue an AMI command to show that channel to get all the variables that are on it.
14:13.06Guest0193Yeah I think I can do it though... since I know between which strings the variables will be.
14:13.21*** join/#asterisk dadrc (~quassel@unaffiliated/drc)
14:18.19*** join/#asterisk dadrc (~quassel@unaffiliated/drc)
14:19.47*** join/#asterisk ChkDigit (~u388mw@74.3.144.66)
14:51.41*** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-czqgwlvcrmwkfluv)
14:51.41*** mode/#asterisk [+o kharwell] by ChanServ
15:11.19*** join/#asterisk Maxxed (~Maxxed@nrgt.net)
15:14.07*** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772)
15:19.34*** join/#asterisk Maxxed (~Maxxed@nrgt.net)
15:47.35*** join/#asterisk [TK]D-Fender (~joe@216.191.106.165)
15:50.01*** join/#asterisk miralin (~Thunderbi@81.177.58.137)
16:03.32*** join/#asterisk MLC (~MLC@63.249.40.11)
16:04.29MLCI have this in my cli_aliases.conf: config copy dev=!/usr/local/bin/mmasterisk-copy-and-fix.sh
16:04.29MLCbut when I use it I get this: No such command '!/usr/local/bin/mmasterisk-copy-and-fix.sh'
16:05.43*** join/#asterisk pabe (~pabe@81.24.66.208)
16:07.05fileshell commands like that aren't supported, it's a feature of the library used to give the prompt - not the CLI functionality in Asterisk
16:07.33MLCbummer
16:20.23*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
17:38.16MLCPJSIP. I'm trying to set up a 2nd line with voip.ms.  It works fine when the new line is the only one, but when I also include the voip.ms that was already there and working, the new one returns 401 unauthorized to the invite. Both are connecting to the same voip.ms server.
17:39.11fileyou'll have to be specific on who is responding to who, ie inbound or outbound
17:39.45MLCon an inbound call, my asterisk responds to the voip.ms invite with 401
17:41.59filegenerally ITSPs don't authenticate and it's expected instead to do IP based matching and direct accordingly
17:42.48fileit's entirely possible that if that isn't happening the inbound call is matching your other trunk and challenging for those credentials
17:44.51MLCyes . I removed auth= from both endpoints, and now I can get through on both, but both trunks are matching the first one. The 2nd should be going to a different context in the dialplan.
17:45.06MLCbut is going to the context for the 1st
17:45.11filematching such things is complicated
17:45.22fileyou can try line support but I don't recall if they support it
17:45.37MLCtried that, but doesn't seem to be helping
17:45.58MLCI'm on 16.2.  Would there be any difference on the latest?
17:45.58filethen you may not be able to differentiate
17:46.04filenope
17:46.23MLCthey have several servers, maybe I'll move the new line to a different server
17:56.56*** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood)
17:57.15SamotThe calls are coming from the same source.
17:57.28SamotThe first one to match wins.
17:58.34MLCmakes sense
17:59.08SamotYou would have the same issue with chan_sip
17:59.12filepart of my presentation at Astricon covers endpoint matching, in fact
17:59.40MLCdo you have those presentation materials on line?
17:59.50fileconsidering I haven't given it yet, nope
17:59.55SamotPut it on youtube or something. I will want to see it
18:00.02fileis talking for an hour about PJSIP
18:00.07MLCtime travel!   88 MPH
18:00.10MLCnice
18:00.26fileevery aspect of it, why things are the way they are, yada yada
18:00.54seanbrighti'll be there to do the appropriate amount of heckling
18:01.47filemy slide deck is currently...
18:01.48file49 slides.
18:02.54filehere's a single sentence! Don’t reinvent the wheel (implementation and terminology)
18:03.15MLC:)
18:08.23fileseanbright: would you like to come up and spout off random stuff?
18:27.16igcewielingDoes anyone know of pros/cons of using the Asterisk res_hep_pjsip, res_hep_rtcp, and res_hep .vs. using a capture agent and/or mirroring port to capture the packets?
18:31.42sawgoodigcewieling: It might not be much help: but when I want to capture:  I put a 2-port TAP on the same cable as the phone or PBX and instantly watch packets with Wireshark for example ...
18:32.59sawgoodI found a really low-cost solid TAP for $79 bucks while at the wireshark (SharkFest show) a few years back ...
18:33.43igcewielingI have 6 servers to capture, using a mirroring port would by easier if I go that route.
18:36.14sawgoodright on, sir: I was in that same situation the other day: and I too used the mirror port to caputre and send to Homer (so to speak)
19:28.58seanbrightfile: i don't think that would be prudent
19:54.45*** join/#asterisk vandyk (~vandyk@179.232.89.68)
20:05.17*** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131)
20:07.22*** join/#asterisk masked (~masked@hpavc/masked)
20:07.48maskedhi
20:08.25maskedim using opensips and relaying subscribe rewuests to pjsip/asterisk but asterisk responds to the subscribes with a 404 not found
20:08.59maskedim trying to receive application/simple-message-summary
20:09.50maskedim using odbc realtime
20:09.56maskedSamot: hi
20:11.01maskeddo i somehow need to tell it to look in the right context?
20:11.19maskedim using a itsp endpoint to connect the opensips and asterisk
20:16.56*** join/#asterisk techquila (~techquila@116.90.74.243)
20:39.51*** join/#asterisk techquila (~techquila@116.90.74.250)
20:47.15maskedAmyMalik: hi
20:56.03wyoungmasked: hi
22:03.10maskedwyoung: why hello there
22:16.22*** join/#asterisk jasonwert (~w3rt@198.167.245.16)
22:54.06*** join/#asterisk i9zO5AP (~BQcdf9eiZ@41.140.181.34)
23:26.37*** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl)
23:53.39optiIs there a config option to explicitly register every x seconds, or is asterisk hard coded to register 15 seconds before default expiry?
23:55.24optidefaultexpiry=x seems to raise the expiry and the register locked with the same 15 second refresh
23:55.51fileit depends on what the expiration actually is in the response, it re-registers 15 seconds before that
23:56.07filethe remote side may change it.

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.