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| 00:31.06 | hdon | so i see that i can listen on, for example, PlaybackFinished. but does node-ari-client allow me to listen for when a particular playback has finished? |
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| 08:36.50 | randolfcri | Hi, I need help. When I telnet and Action: Agents. The result will only display one agent. Any idea why? |
| 08:37.14 | randolfcri | btw we have around 100 agents. |
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| 11:42.30 | a11m80 | hi all |
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| 12:10.38 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.17.0 (2017/07/12), 11.25.1 (2016/12/08), Standard: 14.6.0 (2017/07/12); 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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| 14:17.33 | sibiria | hello. in asterisk 13, how can i enable verbose call-ids in the messages and/or full log? |
| 14:17.58 | sibiria | use_callids is yes in logger.conf, but they aren't present in the output |
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| 14:29.32 | pchero_work | sibiria: |
| 14:29.32 | pchero_work | core set verbose 9 |
| 14:29.32 | pchero_work | or |
| 14:29.32 | pchero_work | core set debug 9 |
| 14:29.42 | pchero_work | probably helps you. :) |
| 14:32.43 | sibiria | pchero_work: verbose and debug is already at 10. i suspect it's not being presented in the same way as the ancient v1.8 |
| 14:32.55 | sibiria | where we get a very long id in the id@peer format |
| 14:33.21 | pchero_work | are you using the 1.8? |
| 14:33.27 | sibiria | not anymore but that's the log format i'm used to |
| 14:33.36 | sibiria | is asterisk 13's callid the shorter 4 byte hexscii value? |
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| 14:36.24 | sibiria | it just doesn't mention any details about the SIP dialogs |
| 14:36.33 | sibiria | it's as if it's ignoring the verbose/debug setting in asterisk.conf |
| 14:36.46 | pchero_work | how about sip debug on? |
| 14:38.10 | sibiria | SIP Debugging re-enabled |
| 14:38.14 | sibiria | so apparently it's on |
| 14:38.20 | sibiria | kinda confusing |
| 14:38.25 | sibiria | afk a while, gonna try something else |
| 14:38.42 | pchero_work | Hm.. |
| 14:38.51 | pchero_work | I can't find callid in Asterisk-13 either. |
| 14:39.18 | pchero_work | What callid does in Asterisk-1.8? |
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| 14:43.32 | Samot | ? |
| 14:44.14 | pchero_work | sip set debug on |
| 14:44.27 | Samot | CallID is not a setting |
| 14:44.36 | Samot | It is a SIP header. |
| 14:44.52 | pchero_work | In asterisk-14.3.0, after sip set debug on, I can see it. |
| 14:44.56 | pchero_work | Call-ID |
| 14:45.06 | pchero_work | grep the Call-ID |
| 14:45.11 | Samot | Right. It is a standard SIP header. |
| 14:45.37 | Samot | The only way to get the Call-ID is to have a debug running |
| 14:45.41 | Samot | Which is a waste. |
| 14:45.44 | pchero_work | it would be the same in the Asterisk-13 as long as you use the chan_sip. |
| 14:45.50 | Samot | No. |
| 14:45.54 | Samot | Call-ID is a SIP thing. |
| 14:46.00 | Samot | Every SIP message has a Call-ID |
| 14:46.24 | pchero_work | Agree. But, he seems like need it. ;) |
| 14:46.29 | Samot | Doesn't matter if it's Chan_SIP or Chan_PJSIP |
| 14:46.37 | Samot | Well that's not something Asterisk does |
| 14:46.52 | pchero_work | sip command is for the chan_sip isn't it? |
| 14:46.55 | Samot | That's what a softswitch/SIP router does |
| 14:47.03 | Samot | sip set deub on |
| 14:47.17 | Samot | That will show a SIP debug of SIP messages. |
| 14:47.24 | Samot | If you leave it on, it will be logged |
| 14:47.30 | Samot | But it doesn't have a format |
| 14:47.37 | Samot | It doesn't have "logids" |
| 14:47.45 | Samot | It's just logged because it hits the console. |
| 14:48.07 | Samot | You cannot just "pull/search" for full SIP messages in the log |
| 14:48.18 | sibiria | pchero_work: i got the call-ids to manifest after restarting asterisk a few times. i have no idea how that solved it but it did and it now picks up the correct debug level |
| 14:49.38 | pchero_work | Strange. |
| 14:49.59 | pchero_work | Maby... are you running the Asterisk on the RBP? |
| 14:50.05 | pchero_work | Raspberry pi? |
| 14:50.26 | sibiria | no, this is on centos running in aws - maybe related to XFS and EBS storage, who knows.. |
| 14:50.30 | sibiria | wouldn't surprise me |
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| 15:54.07 | DivideBy0 | hdon: it will have the id of the playback that finished, so you'll get all Playbackfinished events, and you'll check them all for the ID you're looking for. It's optional, but highly helpful to give each playback you queue and ID of your choosing |
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| 17:35.38 | kannan | i am not able to build asterisk 13 from soruce in debian stretch. http://paste.debian.net/983738/ |
| 17:36.15 | Samot | OK. |
| 17:36.23 | Samot | So 1) Are you actually paying Digium for support? |
| 17:36.59 | kannan | hi Samot :). No . and no, this is a vanilla download. I can try an LTS version instead also if need be. |
| 17:37.09 | Samot | Well |
| 17:37.22 | Samot | The certified version doesn't give you any benefits if you're not paying for it |
| 17:37.25 | Samot | So use LTS |
| 17:37.33 | kannan | very well i will try that. |
| 17:38.12 | Samot | You ran the prereqs step right? |
| 17:38.30 | Samot | Made sure all the needed dependencies are installed? |
| 17:39.34 | kannan | yes i am sure of that ./configure (and in make menuselect i removed res_srtp in hope that the make error will not happen ) |
| 17:39.53 | Samot | No. |
| 17:40.07 | Samot | That's not running prereqs |
| 17:41.29 | Samot | [root@server asterisk-14.X.Y]# cd contrib/scripts |
| 17:41.30 | Samot | [root@server asterisk-14.X.Y/contrib/scripts]# ./install_prereq install |
| 17:41.34 | Samot | ^^ Did you do that? |
| 17:41.44 | kannan | Samot , no - i will do |
| 17:42.05 | Samot | Do you have things like OpenSSL installed? |
| 17:42.20 | hdon | DivideBy0, thanks. it looks like my use case is well supported by node-ari-client, in fact (pretty common use case i think; responding to PlaybackFinished event for a particular Playback) |
| 17:43.14 | kannan | Samot yes, i have openssl installed |
| 17:43.28 | hdon | i've got another ARI question relating to reconnecting the web socket: can i enumerate the channels who are stasis'd into the app and then continue to operate them via ARI like nothing ever happened? (minus the events that were missed in the meantime, of course) |
| 17:43.33 | Samot | OK |
| 18:06.43 | kannan | Samot : thanks, built fine! |
| 18:07.50 | Samot | Cool |
| 18:33.56 | tuxd00d | kannan: Also make sure to use â./configure --with-pjproject-bundledâ |
| 18:34.36 | kannan | tuxd00d : i will read up and do |
| 18:34.47 | tuxd00d | if https://wiki.asterisk.org/wiki/display/AST/PJSIP-pjproject |
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| 18:39.34 | kannan | tuxd00d : thanks. i will re do. |
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| 19:18.10 | TECFALL | I have asterisk 13.9.1 and it utilizing 20+% of the CPU |
| 19:18.24 | TECFALL | I've never had this experience with my many years of Asterisk. |
| 19:18.37 | TECFALL | Is this is a known issue in this version? |
| 19:19.13 | Samot | Well.. |
| 19:19.15 | Samot | It's outdated. |
| 19:19.28 | Samot | So it might of had issues with stuff back then. |
| 19:19.53 | Samot | You are almost a year behind in updates. |
| 19:20.29 | Samot | At least 8 months if every release was on the 4 week schedule. |
| 19:20.39 | Samot | But updates are every 4-6 weeks |
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| 20:03.27 | TECFALL | Samot: What is the easiest way to upgrade to the newest version? |
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| 20:04.27 | Samot | You compile it like normal |
| 20:04.43 | Samot | Did you install from source? |
| 20:04.47 | Samot | Or from a package? |
| 20:05.33 | TECFALL | Samot: from source |
| 20:05.50 | Samot | OK, the compile like normal from the source. |
| 20:06.24 | TECFALL | Then just replace extension.conf, sip.conf, voicemail.com, queues.conf, etc. with my current? |
| 20:10.09 | TECFALL | Samot: or does it not overwrite the current files? |
| 20:10.23 | Samot | It will not override them |
| 20:10.41 | Samot | But you should always backup your stuff before you do this |
| 20:10.59 | [TK]D-Fender | configs don't get overwritten unless you tell it to |
| 20:11.22 | [TK]D-Fender | If you read the instructions you'd know which one does this |
| 20:16.18 | TECFALL | Are the instructions in the source? |
| 20:17.34 | rrittgarn | with app_queue, for the announcements, is there any way to ensure that certain announcements get played X number of times, outside just setting frequency and timeout for the queue overall? Seeming inconsistent in results (still testing though). |
| 20:18.43 | rrittgarn | specific use case: trying to get a queue to play an intro message the first time, then a 'keep holding' message 3 times, and then finally a 'sorry'. I think i can get away with the please hold message and sorry messages before and after calling queue(), but its the three in the middle that I'm questioning myself |
| 20:20.14 | [TK]D-Fender | TECFALL> Are the instructions in the source? <- Yes, and on the wiki. |
| 20:21.31 | [TK]D-Fender | rrittgarn, specify them in the exact order you expect. |
| 20:21.46 | [TK]D-Fender | And make sure your queue timeout matches the playback duration |
| 20:22.13 | rrittgarn | multiple entries, or comma delimited. I keep finding Leif's guide from 2010 when searching the wiki and it doesn't specify |
| 20:22.18 | rrittgarn | (and thanks) |
| 20:22.43 | [TK]D-Fender | rrittgarn, sample configs tell you this.. |
| 20:23.05 | [TK]D-Fender | Never fail to read the actual samples provided. they are the first line of documentation |
| 20:24.11 | rrittgarn | yeah, i did, but as most people who ask here, i didn't read far enough down |
| 20:24.24 | rrittgarn | just saw it defined above in the example, didn't make it the 345 lines in |
| 20:24.25 | rrittgarn | thanks |
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| 20:54.31 | DivideBy0 | hdon: yes you can operate on them like nothing ever happened (except your missed events are truly missed) so your states may have changed |
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| 22:25.32 | tuxd00d | [PJSIP] Is there a way to shorten the timeout for TLS connections. Replace contact isnât an option as there are multiple contacts per AOR. Other than a large |
| 22:25.43 | tuxd00d | âmax_contactsâ. |
| 22:26.51 | tuxd00d | When PJSIP is reloaded, there is a bunch of âUNVAILâ that sit for a long time, prevening the enpoints from connecting due to âover max contact limitâ |
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| 22:40.19 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:40.23 | rmudgett | tuxd00d: See ASTERISK-27147 TCP/TLS contacts created using rewrite_contacts are now removed when the connection is broken or Asterisk restarts. |
| 22:41.09 | rmudgett | Also see ASTERISK-27192 about an upcoming change to the remove_existing option. |
| 22:42.59 | tuxd00d | rmudgett: Awesome. Thank you. |
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