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10:54.17 | bolyon | Hi guys. |
10:54.37 | bolyon | Please , I have an issue with my patton registration on my asterisk Pbx |
10:54.56 | bolyon | chan_sip.c:17309 check_auth: Correct auth, but based on stale nonce received from '50' |
10:55.38 | bolyon | I have previously add a GOIP gateway with username '50' and the patton later with username '51' |
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10:56.10 | bolyon | asterisk seems to reassigne the same nonce it give to the goip gateway. |
10:56.24 | bolyon | can someone help please? |
10:56.34 | bolyon | Sorry for my broken english, thanks |
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11:52.34 | p1ke_ | Is it possible to set SIP Session Timers settings per trunk or only globally? |
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11:56.51 | Samot | Yes |
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12:37.45 | Kobaz | is there an option to have yelink hilight the current line you're on? |
12:38.01 | Kobaz | if i have say, three registrations |
12:38.18 | Kobaz | i answer a call that's for reg 2 (attached to button 2) |
12:38.28 | Kobaz | there's no visible indication at all that my call is on button 2 |
12:40.06 | Kobaz | https://imgur.com/a/xD5Z54L |
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12:40.15 | Kobaz | all the buttons look the same |
12:42.48 | Kobaz | also... this is for receiving a call.... when placing a call it does show what line key you're using |
12:42.58 | Samot | What is the line that is red? |
12:43.21 | Kobaz | BLF |
12:43.31 | Kobaz | monitoring the station that called me |
12:44.23 | Samot | And what happens when you hold the call? |
12:44.38 | Kobaz | nothing interesting |
12:44.43 | Kobaz | it goes on hold |
12:44.50 | Kobaz | doesn't light up any buttons |
12:45.54 | Kobaz | using line type 15 for the top left three buttons |
12:46.08 | Kobaz | oh crap |
12:46.10 | Kobaz | i think i know what this is |
12:46.15 | Kobaz | linekey.2.line = 1 |
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12:46.19 | Kobaz | thaaat's why |
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12:48.30 | Kobaz | (I hope) |
12:49.05 | Kobaz | makes sense though |
12:49.33 | Kobaz | also, yealinks can't dial numbers with dashes in them |
12:49.35 | Kobaz | those punks |
12:50.06 | Kobaz | yay cool, line buttons work now like they are supposed to |
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12:55.48 | Samot | Why would there be dashes in the numbers? |
12:55.57 | Kobaz | because, why not? |
12:56.04 | Kobaz | polycom supports it |
12:56.14 | Kobaz | what if i want to dial 1001-1 and 1001-2 |
12:56.38 | Kobaz | it converts it to 10011 and 10012 |
12:56.49 | Kobaz | so that's why i'm using 1001a and 1001b, etc |
12:57.09 | Kobaz | need some sort of distinction from an actual 'normal' number |
12:57.14 | Samot | So those are actually three different accounts? |
12:57.36 | Kobaz | yeah |
12:57.45 | Samot | OK. |
12:58.07 | *** join/#asterisk grandfso (~quassel@185.180.205.67) |
12:58.08 | Kobaz | and i want to map an exten to those accounts, 1:1 |
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12:58.20 | Kobaz | fun times |
12:59.12 | Samot | Just out of curiosity, why is there 1001a, 1001b |
12:59.24 | Kobaz | doing custom stuff for shared lines |
12:59.43 | Kobaz | go direct to line 2, sort of thing |
12:59.49 | grandfso | I'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 ? |
13:00.01 | Kobaz | and... asteirsk 11 chan_sip... so no AOR |
13:01.01 | Kobaz | grandfso: transfers... you'll need CEL... and good luck! |
13:01.07 | Kobaz | following transfers is so much fun |
13:01.12 | [TK]D-Fender | grandfso, Sounds like you haven't secured your server |
13:01.58 | Kobaz | [TK]D-Fender: what makes you say that? he hasn't described what the data looks like at all |
13:02.06 | [TK]D-Fender | grandfso, Stop allowing anonymous SIP connection from the outside. that's "allowguest=no" in chan_sip and there is a similarly named on for PJSIP |
13:02.19 | Samot | No there isn't. |
13:02.20 | [TK]D-Fender | "see. Some are internatl transfers i believe" <- |
13:02.25 | grandfso | Kobaz: 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 |
13:02.36 | Samot | There is no allowguest variation for PJSIP. |
13:02.56 | [TK]D-Fender | If you have calls to numbers that you don't recognize then most likely it's the typical toll fraud attempts |
13:03.44 | Kobaz | definitely a possability |
13:03.59 | grandfso | [TK]D-Fender: naah, it's set to allowguest=no :) |
13:04.17 | Samot | grandfso: So this is a Chan_SIP peer? |
13:04.19 | grandfso | but we use asterisk as the backend for another system |
13:04.29 | grandfso | Samot: yes |
13:04.45 | [TK]D-Fender | grandfso, So it could be coming from the front-end? |
13:05.39 | grandfso | [TK]D-Fender: you mean like initiating a call from the front-end? yes |
13:07.28 | grandfso | it 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) ? |
13:09.04 | Samot | Why do you mean abandoned? |
13:09.07 | Samot | Hung up? |
13:09.07 | grandfso | correction: CEL Status: enabled ;) |
13:09.09 | Samot | Canceled? |
13:10.14 | grandfso | Samot: abandoned --> received by asterisk, was not picked up by operator and got disconnected before operator was available. |
13:10.41 | Samot | OK, so an incomplete call. |
13:10.45 | Samot | A canceled call. |
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13:12.00 | grandfso | Samot: thanks for correcting my nomenclature :) |
13:12.14 | Samot | Well Asterisk has Queues |
13:12.37 | Samot | Abandoned Calls is something queues track when someone exists the queue will still holding. |
13:12.45 | Samot | exits |
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13:16.53 | grandfso | thanks for the suggestion |
13:19.05 | Samot | So how do you check the CDRs? |
13:20.14 | grandfso | RIght 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 |
13:22.08 | Samot | Because it does that. |
13:22.51 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
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13:23.00 | Samot | v12 saw an overhaul to the CDRs. |
13:24.19 | Samot | I call into your PBX and I hit the IVR, that's record 1 |
13:24.41 | Samot | I press 3 to go to a Ring Group that dials three endpoints, that's record 2 |
13:24.51 | grandfso | we're on v.11 |
13:25.01 | Samot | Oh. |
13:25.13 | Samot | Well it's EOL. |
13:25.18 | grandfso | Right :/ |
13:25.26 | Samot | You really shouldn't be on it. |
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13:27.12 | file | 11 had no defined spec, and it evolved a few times, so it's a bit of a... who knows |
13:27.53 | Samot | A C.F. ? |
13:28.20 | Samot | Rhymes with Luster Buck? |
13:29.27 | grandfso | Samot: you got gud rhymes :) |
13:29.53 | Samot | Yeah, Jay-Z has me on speed dial. |
13:30.30 | grandfso | Yeah, the company is running pretty old software, can update everything at once.. you know techno debt |
13:32.00 | Kobaz | mmm techno debt.... i more prefer lo-fi hip hop |
13:33.09 | grandfso | music references all the way, nice |
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14:27.35 | aberrios | anyone 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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14:56.48 | igcewieling | FreePBX 12.0.76, in Ring Groups, the option Send Progress re-checks itself when saving the ring ground. Any idea on how to fix? |
14:57.02 | igcewieling | ring group, not ring ground |
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15:07.18 | Samot | You would have to look at the code. |
15:07.28 | Samot | 12 is EOL and there is no support for it. |
15:07.57 | Samot | So if this is a bug, it's there to stay. |
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