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| 00:27.39 | drmessano | No more ATAs? |
| 01:01.22 | Samot | drmessano: Just no more SPA's. |
| 01:01.42 | drmessano | What is the replacement? |
| 01:01.47 | Samot | drmessano: The Yeastar FXS devices. |
| 01:02.05 | Samot | 1, 2, 4, 8, 16, 32 port options. |
| 01:02.10 | Samot | Better call control. |
| 01:04.26 | drmessano | oh |
| 01:04.45 | Samot | I can do trunking |
| 01:05.18 | Samot | Or hunt the ports. |
| 01:06.24 | Samot | They run old school Asterisk but gives some benefits. |
| 01:06.28 | Samot | Like call records. |
| 01:06.38 | drmessano | I need to come up with a new home phone standard. Already been asked about backlit phones |
| 01:08.08 | Samot | Analog? |
| 01:10.51 | drmessano | No, SIP |
| 01:11.29 | drmessano | Been doing a mix of ATAs and phones.. I have enough SPA112's in the closet to last a decade |
| 01:11.52 | drmessano | So Just need some new desk or dect cordless or something |
| 01:12.46 | Samot | B/W or color? |
| 01:12.56 | drmessano | I don't think it matters |
| 01:14.22 | Samot | VVX150 |
| 01:14.56 | Samot | They're like $60 or so depending on PSU needs. |
| 01:15.39 | Samot | 2 lines, B/W backlight |
| 01:15.50 | Samot | er monochrome backlit |
| 01:16.37 | drmessano | That's not too bad |
| 01:17.43 | drmessano | $49 on Amazon |
| 01:17.43 | Samot | The VVX250 is now my go to low/entry phone for people. |
| 01:17.47 | Samot | Yup. |
| 01:17.53 | Samot | Even better, probably PoE only. |
| 02:08.39 | Nivex | The datasheet for the VVX250 seems to indicate it's IPv4 only :( |
| 02:10.36 | Nivex | I have a couple of Yealink T21P E2 handsets kicking around the network. They're not fancy, but they get the job done. |
| 02:14.25 | drmessano | No OPUS either |
| 02:30.08 | Samot | I don't use OPUS |
| 02:30.25 | Samot | So it's not a requirement for my phone needs. |
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| 12:44.44 | fredcwbr-GO | Good morning, |
| 12:44.44 | fredcwbr-GO | Greetings from Brasil (BR) |
| 12:44.45 | fredcwbr-GO | I1m looking for guidance on what mailing list should use to ask about queue.app on asterisk |
| 12:44.45 | fredcwbr-GO | Dug into the code and was thnking in a fork for a specific need regarding a associaton of a tag to the queue name on the connected instance . Think got already, just some confirmations about side effects. |
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| 14:20.58 | teitoklien | Hey there Everyone ! |
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| 15:51.26 | fredcwbr-GO | Howdy, Greetings from Brasil. I'm looking for guidance on what mailing list should use to ask about queue.app on asterisk |
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| 15:55.30 | jrun | hi, if uac sends a REGISTER with expire=0; what is expected of uas? send back Expire: <old_value> or Expire: 0 ? |
| 15:56.04 | jrun | server i'm having problem with built off asterisk-1.8 series |
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| 16:09.06 | Samot | jrun: Generally expire=0 will remove the location. Also, 1.8 is EOL. |
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| 16:17.54 | jrun | Samot: i just realise that i'm sending 'Expires: 0' and don't append ';expires=0' to the contact, which one is supposed to take precedent; or asterisk honors basically? |
| 16:18.13 | jrun | Samot: i just wanna go back to the sysadmin with this; they will hopefully upgrade. |
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| 16:20.35 | Samot | Why are you sending expire=0? |
| 16:20.44 | Samot | Are you trying to remove a location? |
| 16:21.25 | jrun | when i say, i'm sending; it's basically a copy-paste of pjsip's pjsua... |
| 16:21.41 | Samot | That doesn't tell me anything. |
| 16:22.17 | jrun | yes, i'm removing a localtion. |
| 16:22.35 | Samot | So what's the actual issue? |
| 16:23.20 | jrun | pjsip_regc_unregisteri'm sending 'Expires: 0' |
| 16:23.23 | jrun | opps! |
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| 16:24.22 | jrun | i'm using pjsip_regc_unregister() and it adds Expires:0 header; nothing is appended to the 'Contact:' |
| 16:24.37 | Samot | OK so this isn't an Asterisk issue. |
| 16:41.17 | jrun | hmm, now sending this: |
| 16:41.45 | jrun | https://gist.github.com/b01ee7811fc49e5c9a1ce2403f039d41 |
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| 16:42.41 | jrun | this is what server sends back: |
| 16:42.44 | jrun | https://gist.github.com/cf55433681beb556c89172e9b2506998 |
| 16:43.43 | jrun | normal? |
| 16:44.52 | jrun | if this rings a bell of yesteryear bugs; a pointer would be helpful. |
| 16:45.12 | jrun | looks up in jira |
| 16:45.55 | Samot | 1.8 is EOL. |
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| 17:12.21 | igcewieling | I hate hate hate upgrading and even I am using at least Asterisk 11 on all my Asterisk servers. |
| 17:12.39 | igcewieling | Most of my Asterisk servers, I'm using Asterisk 16 on a few of them. |
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| 18:02.22 | DanFromUK | Our SIP provider is dropping calls that enter into a Record command within an AGI. (I've not yet tested if they are also dropping calls that enter Record via a normal dialplan). |
| 18:02.36 | DanFromUK | They've said that we aren't sending RTP data when the record starts |
| 18:02.45 | DanFromUK | Which causes them to drop the call. |
| 18:03.14 | DanFromUK | Is there any way to cause Asterisk to continue sending silent RTP packets during the Record command? |
| 18:03.25 | file | https://github.com/asterisk/asterisk/blob/master/configs/samples/asterisk.conf.sample#L60 |
| 18:04.53 | DanFromUK | @file Amazing as always! Whats the best way to reload asterisk.conf ? |
| 18:05.28 | file | I do not know if there is such an ability |
| 18:05.32 | DanFromUK | @file I assume the default is no ? |
| 18:06.58 | DanFromUK | Would 'core reload' do it, or could that cause pjsip channels and registrations to drop? |
| 18:07.13 | file | that won't cause pjsip channels and registrations to drop |
| 18:07.17 | file | I don't know if it will reload asterisk.conf |
| 18:07.27 | file | the asterisk.conf file is generally core changing stuff, so restart likely required |
| 18:08.19 | file | you could just do it and then consult "core show settings" |
| 18:09.07 | seanbright | restart is required |
| 18:09.53 | DanFromUK | oh dear. ok, will have to figure that one out. |
| 18:10.15 | DanFromUK | fyi. core show settings doesn't appear to show the status of transmit_silence |
| 18:10.23 | seanbright | hmm |
| 18:10.39 | file | "Transmit silence during rec:" |
| 18:10.51 | seanbright | i wonder if you could write a module in C that you could load that could change that value |
| 18:10.54 | file | if does in 16.9.0 at least |
| 18:10.58 | seanbright | like, as a worst case scenario thing |
| 18:11.41 | DanFromUK | I'm currently on an old v13. Not had the time to upgrade unfortunately. |
| 18:11.57 | DanFromUK | ok, i'll look at it. thanks again. |
| 18:12.22 | file | October creeps ever closer for when 13 goes security fix only |
| 18:12.51 | seanbright | or maybe you could attach with gdb and set it that way |
| 18:13.03 | DanFromUK | @file thanks for the heads up |
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| 18:25.19 | DanFromUK | @file It appears that setting transmit_silence doesn't cause RTP to be sent on Record. At least in v13.17.1. |
| 18:25.33 | DanFromUK | The arrow in this screenshot shows where the Record command was executed https://www.dropbox.com/s/6r6wg95wbnnvbwe/annotation%202020-04-13%20192355.png?dl=0 |
| 18:25.57 | file | do you have a timing module loaded? |
| 18:26.13 | file | otherwise dunno |
| 18:26.45 | DanFromUK | Not unless it's enabled by default. Where would I check that? |
| 18:26.51 | file | "timing test" on the CLI |
| 18:27.14 | DanFromUK | voip1a*CLI> timing test |
| 18:27.14 | DanFromUK | Attempting to test a timer with 50 ticks per second. |
| 18:27.15 | DanFromUK | Using the 'timerfd' timing module for this test. |
| 18:27.15 | DanFromUK | It has been 1000 milliseconds, and we got 50 timer ticks |
| 18:27.41 | file | I don't recall any issues in the area since then, but that's been years |
| 18:28.35 | DanFromUK | Does PJSIP have anything that would also need enabling, in addition to transmit_silence in asterisk.conf? |
| 18:28.45 | file | no |
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| 18:30.22 | DanFromUK | The upstream provider suggested either continuing to send RTP, or set the call to recvonly. Is settings recvonly possible within asterisk? |
| 18:30.30 | file | no |
| 18:31.25 | DanFromUK | Ok. I'm going to test outside of AGI to see if using a normal dialplan causes transmit_silence to be used successfully. |
| 18:31.53 | DanFromUK | I'll also set up a test machine with the latest release of asterisk to see if that's any better. |
| 18:31.56 | DanFromUK | Thanks for your help. |
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| 18:56.18 | DanFromUK | @file FYI. Tests show that (in v13.17.1), a Record initiated by an AGI script ignores transmit_silence. When Record is executed within a normal extensions.conf dialplan, transmit_silence works perfectly. I guess it's a bug. I need to re-write this AGI script using a normal dialplan as I need to fix it fairly urgently. And then I'll try to get a new asterisk installation up and running and will file a bug report if |
| 18:56.18 | DanFromUK | the same happens in the latest release. |
| 19:10.36 | seanbright | yeah, Record() starts a silence generator |
| 19:10.44 | seanbright | 'RECORD FILE' does not |
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| 19:37.06 | DanFromUK | @seanbright I assume from what you are saying, there isn't a workaround to start a silence generator before RECORD FILE? |
| 19:38.11 | DanFromUK | Quick question, does Record() create directories if they don't exist? My AGI script checks for and creates the directories before 'RECORD FILE', but not sure how best to mkdir within a normal asterisk dialplan. |
| 19:43.49 | DanFromUK | Never mind. just going to do System(mkdir -p ..... |
| 19:45.52 | seanbright | > Quick question, does Record() create directories if they don't exist? |
| 19:45.55 | DanFromUK | RECORD FILE retains the recording on hangup. Is it possible to enable the same functionality within Record() ? |
| 19:45.56 | seanbright | yes, it does |
| 19:46.31 | seanbright | *CLI> core show application Record |
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| 19:47.13 | seanbright | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Record |
| 19:47.14 | DanFromUK | ah. i misread the 'k' switch comments. Thought it said 'keep recordING file upon hangup'. Not sure why I even thought that made any sense! |
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| 19:57.51 | DanFromUK | Sorry, one final question, if the destination file is in use by a Playback() or ControlPlayback(), will Record() still be able to update it? It's fine if anyone using Playback() or ControlPlayback() gets their channels dropped and have to redial as it'll only occur once a day. |
| 20:03.20 | DanFromUK | Ignore that. Tested and seems to be fine if it's already in use. |
| 20:03.25 | DanFromUK | Thanks for all your help today! |
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