IRC log for #asterisk on 20190524

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10:54.17bolyonHi guys.
10:54.37bolyonPlease , I have an issue with my patton registration on my asterisk Pbx
10:54.56bolyonchan_sip.c:17309 check_auth: Correct auth, but based on stale nonce received from '50'
10:55.38bolyonI have previously add a GOIP gateway with username '50' and the patton later with username '51'
10:55.47*** join/#asterisk lankanmon (~LKNnet@CPE64777d632383-CM64777d632380.cpe.net.cable.rogers.com)
10:56.10bolyonasterisk seems to reassigne the same nonce it give to the goip gateway.
10:56.24bolyoncan someone help please?
10:56.34bolyonSorry for my broken english, thanks
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11:52.34p1ke_Is it possible to set SIP Session Timers settings per trunk or only globally?
11:53.49*** join/#asterisk yokel (~yokel@unaffiliated/contempt)
11:56.51SamotYes
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12:28.56*** mode/#asterisk [+o bford] by ChanServ
12:37.45Kobazis there an option to have yelink hilight the current line you're on?
12:38.01Kobazif i have say, three registrations
12:38.18Kobazi answer a call that's for reg 2 (attached to button 2)
12:38.28Kobazthere's no visible indication at all that my call is on button 2
12:40.06Kobazhttps://imgur.com/a/xD5Z54L
12:40.09*** join/#asterisk defsdoor (~Andrew@cpc120600-sutt6-2-0-cust232.19-1.cable.virginm.net)
12:40.15Kobazall the buttons look the same
12:42.48Kobazalso... this is for receiving a call.... when placing a call it does show what line key you're using
12:42.58SamotWhat is the line that is red?
12:43.21KobazBLF
12:43.31Kobazmonitoring the station that called me
12:44.23SamotAnd what happens when you hold the call?
12:44.38Kobaznothing interesting
12:44.43Kobazit goes on hold
12:44.50Kobazdoesn't light up any buttons
12:45.54Kobazusing line type 15 for the top left three buttons
12:46.08Kobazoh crap
12:46.10Kobazi think i know what this is
12:46.15Kobazlinekey.2.line = 1
12:46.15*** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com)
12:46.19Kobazthaaat's why
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12:48.30Kobaz(I hope)
12:49.05Kobazmakes sense though
12:49.33Kobazalso, yealinks can't dial numbers with dashes in them
12:49.35Kobazthose punks
12:50.06Kobazyay cool, line buttons work now like they are supposed to
12:55.07*** join/#asterisk defsdoor (~Andrew@cpc120600-sutt6-2-0-cust232.19-1.cable.virginm.net)
12:55.48SamotWhy would there be dashes in the numbers?
12:55.57Kobazbecause, why not?
12:56.04Kobazpolycom supports it
12:56.14Kobazwhat if i want to dial 1001-1  and 1001-2
12:56.38Kobazit converts it to 10011 and 10012
12:56.49Kobazso that's why i'm using 1001a and 1001b, etc
12:57.09Kobazneed some sort of distinction from an actual 'normal' number
12:57.14SamotSo those are actually three different accounts?
12:57.36Kobazyeah
12:57.45SamotOK.
12:58.07*** join/#asterisk grandfso (~quassel@185.180.205.67)
12:58.08Kobazand i want to map an exten to those accounts, 1:1
12:58.15*** join/#asterisk [TK]D-Fender (~joe@216.191.106.165)
12:58.20Kobazfun times
12:59.12SamotJust out of curiosity, why is there 1001a, 1001b
12:59.24Kobazdoing custom stuff for shared lines
12:59.43Kobazgo direct to line 2, sort of thing
12:59.49grandfsoI'm new to asterisk, I'm trying to work with the database to see the actual success rates based on the data in the cdr table, but I have much more connections that I would expect to see. Some are internatl transfers i believe. Anyone has some guide or some suggestion as to how a beginner should proceed to understand the data ?
13:00.01Kobazand... asteirsk 11 chan_sip... so no AOR
13:01.01Kobazgrandfso: transfers... you'll need CEL... and good luck!
13:01.07Kobazfollowing transfers is so much fun
13:01.12[TK]D-Fendergrandfso, Sounds like you haven't secured your server
13:01.58Kobaz[TK]D-Fender: what makes you say that?  he hasn't described what the data looks like at all
13:02.06[TK]D-Fendergrandfso, Stop allowing anonymous SIP connection from the outside.  that's "allowguest=no" in chan_sip and there is a similarly named on for PJSIP
13:02.19SamotNo there isn't.
13:02.20[TK]D-Fender"see. Some are internatl transfers i believe" <-
13:02.25grandfsoKobaz: thanks, I already am happy about the data I'm seeing. Can't understand much of it.
13:02.33[TK]D-Fender"international transfer" isn't a thing
13:02.36SamotThere is no allowguest variation for PJSIP.
13:02.56[TK]D-FenderIf you have calls to numbers that you don't recognize then most likely it's the typical toll fraud attempts
13:03.44Kobazdefinitely a possability
13:03.59grandfso[TK]D-Fender: naah, it's set to allowguest=no  :)
13:04.17Samotgrandfso: So this is a Chan_SIP peer?
13:04.19grandfsobut we use asterisk as the backend for another system
13:04.29grandfsoSamot: yes
13:04.45[TK]D-Fendergrandfso, So it could be coming from the front-end?
13:05.39grandfso[TK]D-Fender: you mean like initiating a call from the front-end? yes
13:07.28grandfsoit seems like i don't have the CEL. Would that mean that we can't track abadoned connections directly from the cdr history? (e.g. theres no enough data) ?
13:09.04SamotWhy do you mean abandoned?
13:09.07SamotHung up?
13:09.07grandfsocorrection: CEL Status: enabled  ;)
13:09.09SamotCanceled?
13:10.14grandfsoSamot: abandoned --> received by asterisk, was not picked up by operator and got disconnected before operator was available.
13:10.41SamotOK, so an incomplete call.
13:10.45SamotA canceled call.
13:11.23*** join/#asterisk brad_mssw (~brad@66.129.88.50)
13:12.00grandfsoSamot: thanks for correcting my nomenclature :)
13:12.14SamotWell Asterisk has Queues
13:12.37SamotAbandoned Calls is something queues track when someone exists the queue will still holding.
13:12.45Samotexits
13:12.47*** join/#asterisk scgm11_ (~scgm11@r186-50-34-181.dialup.adsl.anteldata.net.uy)
13:16.53grandfsothanks for the suggestion
13:19.05SamotSo how do you check the CDRs?
13:20.14grandfsoRIght now I'm just looking into the CDR table in database, trying to figure out why I have more than 1 record for each number called
13:22.08SamotBecause it does that.
13:22.51Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification
13:22.55*** join/#asterisk scgm11_ (~scgm11@r186-50-34-181.dialup.adsl.anteldata.net.uy)
13:23.00Samotv12 saw an overhaul to the CDRs.
13:24.19SamotI call into your PBX and I hit the IVR, that's record 1
13:24.41SamotI press 3 to go to a Ring Group that dials three endpoints, that's record 2
13:24.51grandfsowe're on v.11
13:25.01SamotOh.
13:25.13SamotWell it's EOL.
13:25.18grandfsoRight :/
13:25.26SamotYou really shouldn't be on it.
13:26.00*** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl)
13:27.12file11 had no defined spec, and it evolved a few times, so it's a bit of a... who knows
13:27.53SamotA C.F. ?
13:28.20SamotRhymes with Luster Buck?
13:29.27grandfsoSamot: you got gud rhymes :)
13:29.53SamotYeah, Jay-Z has me on speed dial.
13:30.30grandfsoYeah, the company is running pretty old software, can update everything at once.. you know techno debt
13:32.00Kobazmmm techno debt.... i more prefer lo-fi hip hop
13:33.09grandfsomusic references all the way, nice
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14:27.35aberriosanyone been playing with the SFU and MS Edge? Got a problem with DTLS initialisation. From what i've garnered, MS Edge has an issue with how the certificate is fragmented. Anyone else seen a similar problem?
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14:56.48igcewielingFreePBX 12.0.76, in Ring Groups, the option Send Progress re-checks itself when saving the ring ground.  Any idea on how to fix?
14:57.02igcewielingring group, not ring ground
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15:07.18SamotYou would have to look at the code.
15:07.28Samot12 is EOL and there is no support for it.
15:07.57SamotSo if this is a bug, it's there to stay.
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