21:12.27 | *** join/#asterisk infobot (ibot@rikers.org) |
21:12.27 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.11.2 (2016/09/09), 11.23.1 (2016/09/08), Standard: 14.0.0-rc2 (2016/09/22); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
21:35.31 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
21:54.12 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
22:04.05 | *** join/#asterisk tzafrir (~tzafrir@bzq-82-81-175-197.red.bezeqint.net) |
22:38.56 | *** join/#asterisk zackychan (~zackychan@AToulon-651-1-241-86.w83-113.abo.wanadoo.fr) |
22:39.38 | zackychan | hello everybody , I changed my asterisk server and now I get very often Auto fallthrough, channel 'SIP/my_phone-00000026' status is 'CHANUNAVAIL' |
22:39.53 | zackychan | what can cause that ? |
22:46.37 | zackychan | anybody here ? |
22:52.12 | [TK]D-Fender | Show us the call |
22:52.38 | [TK]D-Fender | ~pb |
22:52.38 | infobot | hmm... pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
22:52.39 | [TK]D-Fender | ^^^ |
22:53.02 | [TK]D-Fender | autofallthough says your call finished all the steps it had to execute |
22:57.24 | *** join/#asterisk puzzola (~puzzola@unaffiliated/puzzola) |
23:06.27 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
23:11.48 | zackychan | I added nat=yes |
23:11.57 | zackychan | it seems to solve the problem |
23:12.15 | zackychan | crazy |
23:32.53 | zackychan | do you understand why ? |
23:35.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
23:40.27 | zackychan | ok bye |