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| 14:07.39 | michael_work | i would like to hear advices on what most stable ast 11 version you use if you do in production? |
| 14:08.04 | michael_work | i have 11.6cert and 11.5 and both seems to work fine while 11.11 seems to have problem with possible memory leak |
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| 15:08.11 | WIMPy | michael_work: That depends on what you want to do with it. |
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| 15:17.01 | michael_work | WIMPy, emmm? |
| 15:18.17 | michael_work | WIMPy, what do you mean by to do with it. We have a lot of production servers, mostly 1.8. i tried to push 11 to production and as few versions work fine, i faced certain problems with 11.11. i wonder if anyone lese faced the problem as well |
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| 15:53.21 | WIMPy | michael_work: Since 11.7.1 it usually locks up for me when trying to transfer calls that came in via IAX, but all the places where that doesn't happen, 11.11 or 11.13 have been fine. |
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| 15:58.46 | michael_work | i had problems with IAX2 on 11 as it did not work with 1.8 for some reasons. it seems to work now. but i prefer not to use IAX@ |
| 15:59.32 | michael_work | the problem is that i have server with about 50 concurrent calls (mostly with recording) and i see that CPU of asterisk jumps high after 15 hours and memory is about 1.5 gib :( |
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| 16:02.00 | WIMPy | Sounds evil. Since which version? |
| 16:05.05 | michael_work | iax problem or memory? |
| 16:05.12 | michael_work | memory problem is from 11.11 |
| 16:05.35 | michael_work | i have new server and thinking if to roll back to 11.5-6 or to upgrade 10 11.13 |
| 16:05.45 | michael_work | s/10/to |
| 16:06.55 | michael_work | i was thinking to set new version for servers as 11.11 coz of it being minimal version with worjing webrtc for me |
| 16:12.13 | file | working webrtc, for now |
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| 16:18.47 | WIMPy | memory |
| 16:21.01 | michael_work | file, \o |
| 16:21.33 | michael_work | file, i wanted to talk to you about something. somebody told me to write to maillist but afaik things from review board are going to ml anyway. |
| 16:22.58 | michael_work | file, https://reviewboard.asterisk.org/r/3867/ - the problem of that if i copy default settings over before loading peer, if system by default is dtls_enabled it would set dtls enabled for peer. |
| 16:23.23 | michael_work | i think the right logic is to use default settings only in case when dtls is enabled on peer |
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| 16:25.19 | michael_work | let's say that global setting is to enable dtls as if asterisk is capable of dtls in general and per peer is to turn it on and off per peer |
| 16:25.46 | michael_work | and why is working webrtc for now? what is going to be changed again? |
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| 19:19.27 | P-NuT | Hi all, do you know when Asterisk 13 will leave beta and go to GA? |
| 19:20.05 | WIMPy | GA? |
| 19:20.22 | P-NuT | General Availability |
| 19:20.37 | P-NuT | General Release... Not Beta. |
| 19:25.10 | P-NuT | Any idea? |
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| 20:22.32 | ChannelZ | When it's golden brown on top |
| 20:43.37 | mjordan | hint: the Asterisk Versions page shows historically when we've released prior versions |
| 20:43.44 | mjordan | you can probably infer when 13.0.0 is going to come out |
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| 21:08.51 | jdzielny | Hello everyone. I'm trying to set up a virtual extension so it rings for a bit and then goes to voicemail, but instead it's automatically going to voicemail |
| 21:09.32 | jdzielny | Any suggestions? |
| 21:17.00 | mjordan | jdzielny: Ringing + Wait? |
| 21:17.44 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Application_Ringing |
| 21:17.53 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Application_Wait |
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| 21:23.52 | jdzielny | hmm |
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| 21:24.19 | jdzielny | mjordan, thought there'd be an easy way to do it with the GUI |
| 21:24.25 | jdzielny | I'm using FreePBX, btw |
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| 21:32.39 | mjordan | probably should ask that in #freepbx then |
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| 21:36.05 | jdzielny | Did, didn't get any response |
| 21:36.16 | jdzielny | That's why I asked here to figure out how to do it with asterisk itself |
| 21:36.22 | jdzielny | Since FreePBX is just a frontend |
| 21:40.06 | WIMPy | A frontend that takes over your asterisk. If you go in between it will have its revenge. |
| 21:41.27 | jdzielny | haha yeah that's true, it does tend to do whatever it wants |
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| 22:11.56 | apb1963 | * 1.8 I've got one way audio ... using meetme to conference the two channels... |
| 22:12.25 | apb1963 | I've flushed the firewall |
| 22:12.50 | apb1963 | what else should I be looking at? |
| 22:18.45 | jdzielny | apb1963, I'm a new Asterisk user, so this might not be relevant to your situation, but I had no audio and i solved it by unblocking ports 10001-20000 for inbound UDP traffic |
| 22:19.37 | apb1963 | jdzielny: thanks. flushing the firewall generally accomplishes that :) Right now I'm looking to see if I did something dumb with my mic. |
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| 22:24.55 | apb1963 | nope, that seems ok |
| 22:27.23 | apb1963 | so my asterisk 11 is happy, works fine... but 1.8, hosted elsewhere... one sided audio |
| 22:27.47 | apb1963 | scratches his foot |
| 22:29.27 | mjordan | apb1963: NAT. |
| 22:30.54 | apb1963 | mjordan: right... but the firewall is flushed |
| 22:32.48 | apb1963 | mjordan: unless.... Hmmm... the system my softphone is on may be blocking RTP from outside |
| 22:33.07 | apb1963 | checks the other firewall |
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| 22:36.13 | apb1963 | nope... all good there |
| 22:36.23 | apb1963 | accepts everything from that system |
| 22:37.01 | ruben23 | hi guys |
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| 22:42.58 | mjordan | apb1963: NAT != firewall. You need to check the IP addresses advertised in the SDPs. |
| 22:43.35 | apb1963 | well that's true... but I did check the IPs earlier, and they looked ok |
| 22:44.43 | apb1963 | thing is... it started working a minute ago... the only thing I did different was add the IP for 1.8 to my softphone system...A SECOND TIME. There are now two entries for it. Very, very strange. But, going back to the 1.8 firewall to restore it to verify it still works. |
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| 22:49.10 | apb1963 | yeah... still works. weird. |
| 22:49.18 | apb1963 | oh well, I'm happy :) |
| 22:49.23 | apb1963 | mjordan: thank you! |
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