04:03.13 | *** join/#asterisk infobot (ibot@rikers.org) |
04:03.13 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.24.1 (2018/12/26) 16.1.1 (2018/12/26), Security Only: 15.7.1 (2018/12/26); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
04:33.28 | *** join/#asterisk heckler (~epicurus@unaffiliated/epicurus) |
18:54.06 | *** join/#asterisk infobot (ibot@rikers.org) |
18:54.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.24.1 (2018/12/26) 16.1.1 (2018/12/26), Security Only: 15.7.1 (2018/12/26); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
19:19.43 | *** join/#asterisk life_of_e__ (~life_of_e@108-95-189-245.lightspeed.irvnca.sbcglobal.net) |
19:20.12 | life_of_e__ | Anyone know how often freenode purges nicks? I keep getting booted. |
19:28.04 | igcewieling | life_of_e__: I assume you mean purges idle nicks? I'm on for weeks at a time without problems, so must not have a maximum connection duration |
19:28.35 | life_of_e__ | Yes, exactly, it seems I get purged once a week or so |
19:29.05 | life_of_e__ | I have to go through the registration process each time |
19:37.44 | file | they aren't purging your nick, each time you reconnect you gain a _ at the end |
19:37.54 | file | so you currently have life_of_e, life_of_e_, and life_of_e__ registered to yourself |
19:38.24 | life_of_e__ | I never noticed the extra lines. I better see if there's a "do not reconnect" option. My ISP drops out a lot. |
19:38.45 | life_of_e__ | Thanks for that discovery |
19:42.09 | *** join/#asterisk life_of_e (~life_of_e@108-95-189-245.lightspeed.irvnca.sbcglobal.net) |
19:42.39 | life_of_e | So the IRC client was adding the underscore on reconnect |
19:42.55 | life_of_e | I don't know why, it doesn't have an option to disable that |
19:50.58 | driz | samot your insight is amazing.. i wouldn't fathom that asterisk has grown in 17 years. thanks for your valued input |
19:54.29 | Samot | I wasn't the one who made it sound like they just watched a 17 year old video and discovered how much they don't know about Asterisk. |
19:55.06 | driz | I clearly stated it was 17 years ago and i learned i didnt understand things. Work on your reading comprehension boss. again, thanks for your valued input |
19:55.22 | Samot | No, you asked if any watched the video from 2002 |
19:55.32 | driz | please show where i mentioned a video |
19:55.40 | driz | this is text, should be a simple matter for you |
19:55.48 | Samot | 11:17:18 AM <driz> did any of you watch mark spencer's presentation at phreaknic (nashville, tn) back in 2002? |
19:55.58 | driz | and in there the word video is? |
19:56.16 | Samot | Oh you're asking if people WHERE at a specific location. |
19:56.21 | driz | clearly |
19:56.37 | driz | glad I could assist you, have a nice day |
19:56.59 | Samot | And what would be the relevance of being at a presentation 17 years ago? |
19:57.24 | driz | you're clearly new to asterisk, it was a great presentation back when asterisk was still in its infancy |
19:57.32 | Samot | No, I am not. |
19:58.00 | driz | my apologies, i just assumed you were young and inexperienced due to the reading comprehension. either way, i'm done speaking to you now, have a nice day |
19:58.23 | Samot | OK. |
19:58.38 | Samot | Glad you got that out of your system. I'm going back to real things. |
19:58.48 | driz | asterisk for dummies is a good starter :) |
19:58.51 | driz | enjoy |
20:16.55 | sibiria | life_of_e: it does it because your old user (with the old nick) is still online waiting to time out - you're probably dropping because of your network connection going stale |
20:20.45 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
21:07.53 | *** join/#asterisk youtmon (~yout@c-98-242-250-233.hsd1.fl.comcast.net) |
21:19.02 | life_of_e | sibiria: yeah, my ISP connection occasionally drops because they poorly configured the modem. I didn't know my IRC client would append underscores to the nick on reconnect, though. |
21:24.57 | driz | it just does it if when you reconnect, irc hasn't noticed you dropped yet, so the nick is in use and your client appends the underscore |
21:25.24 | life_of_e | Right, but there was no option to disable that either |
21:25.50 | driz | you can't really disable it, but you can use nickserv to ghost the original name when yo ureconnect |
21:25.56 | driz | if you register the nick, that is |
21:32.20 | *** join/#asterisk rpifan (~rpifan@ipb218f0ee.dynamic.kabel-deutschland.de) |
21:33.38 | *** join/#asterisk pts0 (~pts0@cpe-45-37-8-40.nc.res.rr.com) |
21:33.40 | life_of_e | Given the reconnect option you'd think that another one to add or not add an underscore would be there. I disabled the reconnect anyway so hopefully it won't keep happening |
21:33.56 | pts0 | If you only want to allow opus do you just do a disallow=all allow=opus |
21:35.11 | *** join/#asterisk rockman37 (~rockman37@122-60-43-242-adsl.sparkbb.co.nz) |
21:40.32 | sibiria | life_of_e: it only does it if your configured nickname isn't available |
21:40.44 | sibiria | when you've timed out, it won't be, because the old user is still lingering behind for a few minutes |
21:41.04 | sibiria | you reconnect, nick is not available, client makes a simple "fix" in order to finalize the connection |
21:41.33 | sibiria | pts0: yes |
21:42.14 | sibiria | with pjsip you can simplify it as allow !all,opus - not sure if chan_sip allows that syntax |
22:15.25 | *** join/#asterisk sotoz (~ssss@51.247.195.35.bc.googleusercontent.com) |
22:16.08 | sotoz | Hello, on the Application Dial the documentation says "z - On a call forward, cancel any dial timeout which has been set for this call.". Does this `timeout` refers to the TIMEOUT(absolute) that I may have set in the dialplan? |
22:16.37 | sotoz | or to the `Dial(Technology/Resource&[Technology2/Resource2[&...]],[timeout,[options,[URL]]])` <- timeout here ? |
22:16.41 | sotoz | or maybe both?? |
22:18.06 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
22:25.14 | rockman37 | sotoz: It refers to the latter. |
22:25.40 | sotoz | ok thank you :) |
22:26.07 | sotoz | so, I encounter a weird situation that the TIMEOUT(absolute) that I set in a channel sometimes is not being honored |
22:26.18 | sotoz | and the call lasts for more |
22:26.39 | sotoz | anyone knows why that could happen? |
22:30.06 | rockman37 | Can you show us a log from a call where this happens? |
22:44.57 | sotoz | well, I don't have an asterisk log but I've seen it on my application logs and in my database (and also in the SIP traces) |
22:45.04 | sotoz | it's very difficult to replicate it |
22:45.24 | sotoz | and especially in the very busy production environment that it happens |
22:47.09 | sotoz | so I was wondering maybe there is a situation that this TIMEOUT(absolute) gets reseted or something |
22:53.42 | rockman37 | Mmkay. I would suspect it's to do with TIMEOUT(absolute) not getting set in the first place... Can you narrow it down to a particular part of the dialplan responsible? |
23:07.12 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
23:08.50 | sotoz | hmmm, I'll try, although the dialplan is pretty straightforward. |
23:09.05 | sotoz | no IF statements or strange logic going on |
23:22.15 | *** join/#asterisk Pasha (~Cory@unaffiliated/cory) |