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| 00:19.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12), Standard: 15.5.0 (2018/07/12); 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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| 02:45.35 | dan_j | Is it possible that AGI Record_File does not close the file immediately after hangup, but only after a few moments? |
| 02:46.52 | dan_j | Very occasionally, code that checks the recording length immediately after hangup is unable to read the file. I was wondering whether I need to introduce some sort of delay after AGI Record completes, before attempting to read the new recording? |
| 02:47.53 | drmessano | Sure it takes the file system some small amount of time to close the file, right? |
| 02:49.06 | drmessano | I don't ever assume a file operation is instant |
| 02:49.34 | dan_j | What do you consider the maximum 'small amount of time' would be to guarantee that Asterisk has closed the file? |
| 02:50.14 | dan_j | Surely AGI Record_File should not complete until the file has been released? |
| 03:02.54 | drmessano | Beats me.. I'm saying that it doesn't surprise me to have an operation complete and not have an instant unlock of the file |
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| 03:18.57 | forgotmynick | hello. im trying to create an alerting system. right now im using pushover to send email alerts to zapier and zapier calls a webhook which triggers a call from twilio. i need the call to ring for a long time but you have to answer the trunk which means outgoing call charges. is it possible to make asterisk call an extension through PHP? im hoping then i can set zapier to call the php script directly |
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| 04:14.57 | masoudd | Can you make asterisk call an extension through php? Yes. |
| 04:16.08 | masoudd | forgotmynick: You can either use Asterisk Call Files api or use asterisk manager php api |
| 04:16.35 | masoudd | I'd go with the Call files api |
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| 09:42.23 | danielyk | Hi all, how can I set for a pjsip contact a SDP Attribute. Requirement from our ISP: ... with the SDP a=3ge2ae:applied" attribute |
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| 10:01.24 | file | you can't without modifying the source code |
| 10:06.46 | danielyk | @file This is a requirement of Deutsche Telekom to set this SDP attribute to use SDES. Is something like that unusual? |
| 10:06.58 | file | yes, it appears to be for IMS purposes |
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| 17:00.55 | igcewieling | Has anyone seen this before? channel 3: open failed: connect failed: Connection refused |
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| 17:54.47 | Frod | hello, i have a weird issue, i have a sip trunk connected |
| 17:55.11 | Frod | but i i call from one service provider i get the SIP invite |
| 17:55.39 | Frod | but if i call from anothet even that i get the sip invite the asterisk box does not see it or acceptit |
| 17:55.52 | Frod | any one has an idea on what to look for |
| 18:16.24 | tuxd00d | What does the error actually say? |
| 18:16.49 | tuxd00d | What do the packets from the 'non-working' provider look like? |
| 18:17.25 | tuxd00d | Asterisk is set up using IP auth? |
| 18:19.44 | tuxd00d | Frod: You have one ITSP connection, and calls from one number works, but calls from another number do not? Is that the situation? |
| 18:24.58 | Frod | yes |
| 18:25.12 | Frod | i see that one packet is sligtly larget |
| 18:25.25 | Frod | https://pastebin.freepbx.org/view/4affbf06 |
| 18:25.30 | Frod | the one that works |
| 18:25.59 | Frod | https://pastebin.freepbx.org/view/b6a0a04e |
| 18:26.06 | Frod | donst even get detected |
| 18:26.10 | Frod | by asterisk |
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| 19:28.48 | degenerate | In an older version of asterisk we were hooking into the AMI on a Join event, this event seems to be gone in 13.21.1 is there an equivalent event? what is it called? |
| 19:30.00 | file | https://github.com/asterisk/asterisk/blob/master/UPGRADE-12.txt#L279 |
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| 19:40.54 | Micc | if I use INC on a shared variable, is that an atomic increment? |
| 19:42.01 | file | no. |
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| 20:01.03 | Micc | Shit, is there anyway to do a decrement of a shared variable that would be atomic to asterisk channels, not to the computer, but just between other asterisk dialplan running on other channels using that same shared variable |
| 20:08.10 | Micc | INC/DEC don't seem to even work on shared variables. |
| 20:13.12 | igcewieling | Micc: "core show functions like LOCK" |
| 20:15.27 | igcewieling | If you have an AGI which already uses MySQL, you can use the MySQL locking which should work across different Asterisk servers, as long as they are all talking to the same DB server. |
| 20:16.20 | igcewieling | select get_lock() and select release_lock() |
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