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| 02:29.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.23.0 (2018/09/05) 16.0.0-rc1 (2018/08/08), Standard: 15.6.0 (2018/09/05); 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 -=- Logs: bit.ly/1s4AKKu |
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| 05:02.53 | bigbadpig | Is there a page command for confbridge like for meetme? Suppose I want to automatically put 10 people in a conference and they all have sip phones/extensions? |
| 05:03.22 | bigbadpig | Can I call a number that does that--perhaps the extension dials them and puts them in the conference? |
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| 05:30.22 | [TK]D-Fender | Does page not work for you? |
| 05:30.41 | [TK]D-Fender | Otherwise originate the call-outs and have them individually dump to the conference you're doing this for |
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| 18:06.33 | jploh | Hi, I had an incident earlier where a page with SIP channels caused asterisk to reach the opened file limit. Now there is one call that's been orphaned - sip show call says Owner channel ID: <none>. How can I hang up this channel? |
| 18:12.25 | jkroon | hangup request ? |
| 18:13.19 | jploh | i tried, but it's not in core channels, only in sip show channel |
| 18:13.45 | jkroon | jploh, core restart when convenient then :(. |
| 18:14.01 | jploh | :( okay. Thanks jkroon |
| 18:14.08 | jkroon | and I'd increase the number of file descriptors too if I were you. |
| 18:15.04 | jploh | is the default limit too low? |
| 18:15.32 | jploh | it's on an aws t2.micro |
| 18:24.36 | jkroon | jploh, default is distro dependend, but I typically run asterisk on 8192, and on high volume systems with 16384 |
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| 18:25.30 | jploh | okay thanks. how many calls do you get on that? i know it varies on what modules you have, just want to get an idea |
| 18:26.10 | jkroon | default on my system at least seems to be 1024, which is probably fine for a small number of ongoing channels but runs out surpringly quickly. |
| 18:26.43 | jkroon | idle asterisk 13 system i'm debugging on right at the moment has just under 50 descriptors open. |
| 18:36.15 | igcewieling | For the most part, each call takes two file descriptors. |
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| 18:44.20 | jploh | okay thanks |
| 18:48.35 | igcewieling | On a lightly loaded system I see Asterisk using about 20 fd. |
| 18:50.24 | file | I think these days for a SIP call it would be 3 minimum at all times |
| 18:51.36 | igcewieling | On a system with 25 calls I see about 50 fd. |
| 18:51.49 | igcewieling | file: what is the extra one? |
| 18:52.00 | file | oh maybe 4... |
| 18:52.06 | file | RTP, Alert Pipe, Timer |
| 18:54.23 | igcewieling | 30 calls / 321 fds used, but I assume each AGI also uses an FD? |
| 18:54.35 | igcewieling | fds for MySQL access too? |
| 18:55.29 | jploh | that's without cdr/cel? |
| 18:56.08 | igcewieling | jploh: that is: ls -l /proc/`pidof asterisk`/fd | wc -l |
| 18:56.18 | igcewieling | so it should include cdr and cel fds |
| 18:56.49 | jploh | i see, thanks |
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| 19:30.50 | tdonahue | hi all, does anyone here have any experience with building asterisk for Yocto or openembedded? |
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| 21:13.56 | bigbadpig | Is there a way to use Google Voice with Asterisk since they stopped the XMPP access? |
| 21:19.59 | rmudgett | bigbadpig: No. However there is a patch in the works for it: https://gerrit.asterisk.org/#/c/asterisk/+/9505/ |
| 21:28.19 | bigbadpig | Thanks a lot! |
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