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| 01:51.25 | darkunderlord | lvlinux: nope, sorry. |
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| 02:59.14 | darkunderlord | finally got my main server from 11 to 13 and chan_sip to chan_pjsip. :) Hopefully all goes well tomorrow morning. /me keeps fingers crossed |
| 03:00.32 | avb | darkunderlord: :) congratulations. any perks that you have noticed? |
| 03:00.46 | darkunderlord | ARI, is the one i'm the most excited about. |
| 03:01.03 | darkunderlord | man I can't wait to code some ish for htat |
| 03:01.04 | darkunderlord | that |
| 03:01.29 | avb | ari yes |
| 03:01.35 | avb | whats about chan_pjsip? :) |
| 03:01.43 | avb | darkunderlord: |
| 03:01.58 | avb | any benefits in comparison with chan_sip? |
| 03:02.20 | darkunderlord | oh, not really, but I do want to allow people to use iphone sip client, and other phones on the same extension. The 1 to many is nice about pjsip |
| 03:02.21 | avb | maybe its time for me to move as well |
| 03:02.37 | avb | oh, they finally made that happen |
| 03:02.45 | avb | this is indeed nice |
| 03:02.56 | darkunderlord | so now my work phone, cellphone sip client, and home phone on VPN can't have same extension. :) |
| 03:03.02 | avb | i hope nat handling is not broken :) |
| 03:03.14 | darkunderlord | no idea, I don't have that issue. :) |
| 03:03.27 | avb | jj |
| 03:03.37 | darkunderlord | cool |
| 03:04.04 | darkunderlord | now I want to find out what's cool and new in 15 that isn't in 13 :) |
| 03:04.23 | avb | is still with 11 :) |
| 03:04.36 | darkunderlord | though I do have one more site that is still on 11. But now that I've worked the kinks out, it'll be easier hopefully |
| 03:05.05 | darkunderlord | 11 is good, but I just really wanted to keep up with versions, and moreso get the ARI hotness |
| 03:05.45 | avb | well, keep us updated if you will find something new :) |
| 03:06.20 | avb | were happy like a baby when found out about new hangup handlers |
| 03:22.41 | Samot | avb: What do you mean? |
| 03:23.20 | Samot | CDR specs redone, ARI, PJSIP and there are other things, including fixes. |
| 03:23.29 | Samot | Like 11 is done. |
| 03:23.31 | Samot | Dead. |
| 03:23.36 | Samot | No more updates or fixes. |
| 03:25.17 | Samot | Will 11 continue to work? Of course, it sure has. Can you still run 11 and it will work, of course. |
| 03:25.46 | Samot | But the days of "Oh man, I hope they fix this or add that in 11" passed last year. |
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| 08:30.39 | dnit | Hi I am using AMI originate action to dial some number |
| 08:31.00 | dnit | the moment I bridge the above channel, I get OriginateResponse failure. |
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| 09:59.11 | defswork | if I have a call recording that includes bad call quality from the incoming caller side, can the data issue only exist of the caller side outwards ? I.e. could it be affected by the endpoint side network ? |
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| 11:32.10 | dnit | defswork: Yes it could be only at the incoming side only. |
| 11:33.03 | defswork | cheers |
| 11:33.15 | defswork | just trying to debug this issue |
| 11:33.33 | dnit | You should look for jitter and loss % |
| 11:35.36 | defswork | can you see that on very old asterisk (1.4) ? |
| 11:36.10 | dnit | I though you had raw packet capture. |
| 11:37.26 | defswork | no |
| 11:37.44 | defswork | is there a 1.4 equiv of show channelstats ? |
| 11:38.01 | defswork | (don't ask why still on 1.4 - it's a vendor supplied solutions :| ) |
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| 11:40.07 | catphish | does the queue application always call all agents at the same time, or can it retry in a staggered manner? the reason for my question is that i want to set a very long timeout, but i want devices to be retried immediately if they were previously busy |
| 11:43.48 | defswork | I've turned on rtcp stats |
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| 12:58.44 | [TK]D-Fender | catphish, "strategy" <- that is what it can do + queuerules.conf |
| 12:59.31 | [TK]D-Fender | dnit, guess we'd have to see your actual attempt |
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| 14:26.29 | ryan42 | hi all. i have a strange problem i can't seem to solve. I am using asterisk with aastra 6869i phones. when i place outbound calls, after a few seconds, the 'name' part of the call onscreen gets filled with CID:<my outbound did> |
| 14:26.50 | ryan42 | this makes it so that all the outbound call log entries show that instead of the actually-dialed number |
| 14:27.10 | ryan42 | i can't quite figure out where this is comign from or how to make it stop happening |
| 14:27.13 | darkunderlord | damn I'm so pissed. Third time trying to do my upgrade from 11 to 13 and all hell breaks loose. No rhyme or reason either. I'm thinking it has more to do with my phones (cisco 525g2) than asterisk. Can't even see errors on asterisk side. Not a good day. |
| 14:30.00 | ryan42 | ahh i think i finally found it |
| 14:30.15 | Samot | What problem are you have with the Cisco 525? |
| 14:30.19 | ryan42 | "Display CallerID on Calling Phone" -- a FreePBX thing. |
| 14:30.21 | Samot | s/have/having/ |
| 14:31.23 | Samot | ryan42: Your phone display has nothing to do with the CDR results. |
| 14:31.43 | Samot | ryan42; My phone displays both the number I am calling and the callerID it is calling from... |
| 14:32.05 | Samot | Pretty helpful when I realize, "Oh crap, I'm calling out from the wrong line" |
| 14:32.30 | ryan42 | eh i guess i can see the purpose but here it is breaking the call logs pretty badly on this device |
| 14:32.45 | Samot | What call logs? |
| 14:32.50 | Samot | How are they breaking the call logs? |
| 14:33.01 | Samot | The phone has nothing to do with what ends up in the CDR |
| 14:33.02 | Samot | At all. |
| 14:33.04 | Samot | Zero. |
| 14:33.05 | Samot | Nadda. |
| 14:33.10 | ryan42 | it is a display issue |
| 14:33.13 | Samot | Outside of the digits it presents. |
| 14:33.23 | Samot | What type of phone? |
| 14:33.26 | ryan42 | Aastra |
| 14:33.41 | ryan42 | the phone prefers to show CNAM data in the outbound call log if it is provided |
| 14:34.08 | ryan42 | since asterisk/freepbx was populating CNAM for all the outbound calls as CID:<outbound CID> all the outbound calls look the same |
| 14:34.10 | Samot | You mean the phones call log? |
| 14:34.12 | ryan42 | yes |
| 14:34.23 | Samot | OK, helps to be specific. |
| 14:34.31 | ryan42 | sorry |
| 14:35.17 | Samot | Sorry, The aastra 6869i is not displaying the numbers that were dialed in it's "Dialed Number" history? |
| 14:35.58 | Samot | The Aastra's internal call history when it comes to the numbers it's dialed..and how those are displayed on the phone.. |
| 14:36.06 | Samot | Is an Aastra thing. |
| 14:36.13 | Samot | Not an Asterisk/FreePBX thing. |
| 14:36.50 | [TK]D-Fender | Hard to say |
| 14:36.58 | [TK]D-Fender | We'd have to actually look at the call to be sure |
| 14:36.59 | dnit | What exactly does ChanSpy does ? Can I spy a single channel from an agent , so that agent is able to hear to that channel as soon as it is created ? |
| 14:37.05 | ryan42 | ok. i found the setting to fix it though, so i'm okay now |
| 14:37.09 | Samot | What does that have to do with the call history on the phone? |
| 14:37.10 | Samot | Really? |
| 14:37.16 | [TK]D-Fender | But "looking" never seems to be a solution on the table much these days |
| 14:37.34 | [TK]D-Fender | dnit, Does exactly what it says it does. |
| 14:37.37 | Samot | But how does the call log on Asterisk.... |
| 14:37.40 | ryan42 | KEYWORD:OUTBOUND_CID_UPDATE |
| 14:37.41 | ryan42 | When set to true and when CONNECTEDLINE() capabilities are configured and supported by your handset, the CID value being transmitted on this call will be updated on your handset in the CNAM field prepended with CID: so you know what is being presented to the caller if the outbound trunk supports and honors setting the transmitted CID. |
| 14:37.50 | Samot | Have anything to do with how the phone stores the dialed digit history and displays it? |
| 14:37.54 | [TK]D-Fender | dnit, you spy on a channel. What option does it have? The instructions tell you |
| 14:38.13 | Samot | ryan42: CONNECTEDLINE() shows the Presence of the destination CALLED |
| 14:38.33 | dnit | Yes I read all the instructions but failing to get it to work. :( |
| 14:38.34 | Samot | ryan42: So if I call Gary's extension... |
| 14:38.40 | Samot | It tells me "Gary is Busy" |
| 14:38.50 | Samot | Or "In Meeting" |
| 14:38.54 | [TK]D-Fender | dnit, Try following them then. That usually helps... |
| 14:39.05 | [TK]D-Fender | #garybusey |
| 14:39.14 | Samot | Well in that case |
| 14:39.22 | Samot | The Presence status would be "Insane" |
| 14:39.44 | Samot | 'Gary is Insane. You sure you want this conversation?" |
| 14:39.50 | ryan42 | this only really affects calls outside our organization |
| 14:40.43 | ryan42 | anyway, setting the above setting to no seems to make it work as desired so everything is good now. |
| 14:40.44 | Samot | So the CDRs are correct? |
| 14:40.49 | [TK]D-Fender | continued description is worthless alternative to actually looking #justsayin |
| 14:40.55 | Samot | The actual call history at FreePBX/Asterisk is correct? |
| 14:41.09 | Samot | The only issue is the phone's internal history? |
| 14:41.16 | ryan42 | the CDRs were never the issue. they were and are correct |
| 14:41.34 | Samot | So this is an issue with Aastra's display and internal history? |
| 14:41.39 | ryan42 | i just needed Asterisk/FreePBX to stop passing CID:<outbound DID> as the CNAM |
| 14:41.49 | Samot | What CNAM? |
| 14:41.54 | Samot | Passing it WHERE? |
| 14:41.56 | ryan42 | the CNAM filed of the caller ID |
| 14:41.57 | ryan42 | to the phone |
| 14:42.17 | ryan42 | its telling the phone that the name of the person I'm calling is literally CID:outbound DID |
| 14:42.20 | Samot | What does it pass when you make an "internal" call? |
| 14:42.29 | ryan42 | internal calls were and still are fine |
| 14:42.36 | ryan42 | the setting only affects calls routed over a trunk |
| 14:42.38 | Samot | What does it show the caller on their phone? |
| 14:42.57 | ryan42 | now it just shows the number dialed, and not CID:foo |
| 14:43.10 | ryan42 | before it was showing CID:foo as the name of the person being called |
| 14:43.43 | ryan42 | because the call history on the aastra devices perfers to show CNAM if the data is there, it was showing CID:foo for all outbound calls (and not the number called) |
| 14:43.43 | Samot | But when you call 1NXXNXXXXXX instead of keeping that on the display... |
| 14:43.53 | Samot | It's showing YOUR own CallerID? |
| 14:43.59 | ryan42 | it was |
| 14:44.06 | ryan42 | now it isn't |
| 14:44.31 | ryan42 | the setting that affects this.... |
| 14:44.33 | ryan42 | When set to true and when CONNECTEDLINE() capabilities are configured and supported by your handset, the CID value being transmitted on this call will be updated on your handset in the CNAM field prepended with CID |
| 14:45.36 | Samot | OK |
| 14:45.37 | Samot | So. |
| 14:45.39 | Samot | Turn it off. |
| 14:45.48 | ryan42 | i did that |
| 14:45.53 | Samot | Because while it seems Aastra supports it, you don't like how it functions. |
| 14:46.08 | Samot | I do not have this issue on snom, Polycoms or Cisco SPAs. |
| 14:46.32 | Samot | Their call history is just fine despite the CONNECTEDLINE() details being passed back. |
| 14:46.58 | Samot | And I do have some Aastra's that were BYOD by clients and they don't seem to have this issue either. |
| 14:47.13 | Samot | So I'm going to go back to this is an Aastra thing. |
| 14:47.14 | ryan42 | idk |
| 14:47.22 | Samot | Probably a configuration setting for the phone. |
| 14:47.25 | ryan42 | here's a BR i found about it that led me to the solution: <https://issues.freepbx.org/browse/FREEPBX-11981> |
| 14:47.44 | Samot | ryan42: You should really be in #freepbx for this. |
| 14:48.04 | Samot | Since you are dealing with pre-generated dialplan that you don't control. |
| 14:48.09 | ryan42 | okay |
| 14:48.19 | ryan42 | if i have any issues in the future i'll go there first; thanks. |
| 14:48.20 | Samot | And in here, the help will be to actually make changes at the wrong level for FreePBX. |
| 14:48.21 | [TK]D-Fender | And we're not looking at YOUR call so everything is hypotheitcal nonsense until then |
| 14:48.38 | Samot | What would you like to see in the call TK? |
| 14:48.39 | [TK]D-Fender | Show where it is happening |
| 14:48.53 | Samot | And how will that show the call history on his phone? |
| 14:48.57 | [TK]D-Fender | Someone else's blood test doesn't do me any good at all... |
| 14:48.59 | Samot | And what the phone is displaying? |
| 14:49.11 | [TK]D-Fender | It'll prove if */FreePBX was TOLD to set it <- |
| 14:49.20 | Samot | And then? |
| 14:49.30 | [TK]D-Fender | And then I can start giving a shit and seeing if that can be changed |
| 14:49.31 | Samot | Because if he disable CONNECTEDLINE() sounds like it goes away |
| 14:49.57 | ryan42 | it is definitely a freepbx thing |
| 14:50.05 | [TK]D-Fender | I don't see a call |
| 14:50.06 | Samot | So.. |
| 14:50.07 | ryan42 | sorry, should have asked about it there; but i have it working now |
| 14:50.10 | Samot | So what? |
| 14:50.17 | Samot | Not everything is solved with a call log. |
| 14:50.19 | [TK]D-Fender | and this should be taken over to the appropriate channel |
| 14:50.27 | [TK]D-Fender | Solved, no. Proven, yes |
| 14:50.46 | darkunderlord | I'm googling, but anyone do any capacity and functionality testing with tools? My own spot testing works fine, but when everyone comes in, ish goes sideways. |
| 14:50.47 | [TK]D-Fender | Don't give me a diagnosis without an EXAM taking place |
| 14:50.51 | [TK]D-Fender | that's nonsense |
| 14:51.17 | Samot | ryan42: Show him a call log. |
| 14:51.47 | [TK]D-Fender | Take this to #freepbx first |
| 14:51.52 | [TK]D-Fender | and continue there. |
| 14:51.56 | [TK]D-Fender | It should never have started here |
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| 14:54.21 | dnit | <PROTECTED> |
| 14:54.38 | dnit | Infro about how I am using ChanSpy |
| 14:54.42 | dnit | and the error too |
| 14:55.18 | [TK]D-Fender | ERROR[6961]: chan_pjsip.c:2166 request: Unable to create PJSIP channel with empty endpoint name |
| 14:55.32 | [TK]D-Fender | Well you're never getting off the ground in the first place |
| 14:55.39 | [TK]D-Fender | Forgetting chanspy you aren't calling OUT |
| 14:55.45 | [TK]D-Fender | And it's telling you directly you screwed up |
| 14:56.00 | [TK]D-Fender | Unable to create PJSIP channel with empty endpoint name <- |
| 14:57.51 | dnit | Sorry I did not understand, I am trying to make the PJSIP channel to spy on the channel SPY_ON_CHANNEL |
| 14:58.39 | [TK]D-Fender | Well you didn't call your channel properly |
| 14:58.49 | [TK]D-Fender | "spying" isn't the problem |
| 14:59.08 | [TK]D-Fender | You failed to provide something that could be dialed |
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| 15:11.03 | dnit | Thanks, will look into it. |
| 15:11.13 | dnit | [TK]D-Fender |
| 15:18.52 | dnit | I have been trying to do this for few weeks but failing with all the tools / applications I tried. |
| 15:19.05 | dnit | Something or otherthings breaks. |
| 15:19.15 | dnit | I have a simple requirement. |
| 15:19.26 | dnit | There are agents logged in. |
| 15:19.46 | dnit | Whenever the system makes an outgoing call. |
| 15:19.56 | dnit | Whenever the * makes an outgoing call. |
| 15:20.09 | dnit | We assign an agent to this call. |
| 15:20.56 | [TK]D-Fender | rewind all of that |
| 15:21.12 | [TK]D-Fender | because all of this back-story does nothing until you even have a clue how to DIAL. |
| 15:21.41 | [TK]D-Fender | If you can't dial the guy properly in the first place the resot of this won't mean anything |
| 15:22.34 | dnit | I am able to dial |
| 15:22.42 | [TK]D-Fender | <[TK]D-Fender> Unable to create PJSIP channel with empty endpoint name <- |
| 15:22.46 | [TK]D-Fender | It begs to differ |
| 15:23.08 | dnit | This is a part of something else I was trying to accomplish. |
| 15:23.27 | dnit | I just want my agent to hear the early media of the outgoing call. |
| 15:23.46 | dnit | But I don't have an agent unless the dial begins. |
| 15:24.07 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:24.19 | dnit | Means I don't know which agent has to be assigned to this call unless Dial is reached in dialplan |
| 15:24.20 | eric_hill | dnit, you want the agent to hear the call before the agent is connected to the call? |
| 15:24.47 | dnit | No I want the agent to hear to the outgoing call before it is picked up. |
| 15:25.05 | eric_hill | So call the agent first. |
| 15:25.17 | eric_hill | When the agent picks up, place the outgoing call. |
| 15:25.54 | dnit | This is what I can't do. because unless the outgoing call is placed. I don't have the agent |
| 15:26.14 | [TK]D-Fender | Stop saying "agent" |
| 15:26.27 | eric_hill | dnit, https://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 15:27.29 | eric_hill | Channel goes to your internal extension. When that extension answers, the active call is sent to Context/Extension, which can be an outside number. |
| 15:27.39 | [TK]D-Fender | <dnit> We assign an agent to this call. |
| 15:27.50 | [TK]D-Fender | You effectively could be making that determination at the same time |
| 15:28.03 | [TK]D-Fender | so there is no excuse to not knowing them right from the start |
| 15:28.21 | eric_hill | ^ what TKD said. |
| 15:32.38 | dnit | agent is assigned to call once it is dialled, this business decision I cannot take, |
| 15:33.01 | eric_hill | *how* is the agent assigned. |
| 15:33.34 | eric_hill | random? latest clock-in time? nearest birthdate? lowest social security number? |
| 15:34.19 | dnit | Currently we set a varible via AMI on the outgoing channel, so when the outgoing calls channel is answered we rring the agent. |
| 15:34.53 | dnit | eric_hill agent is chosen based on users score / and other factors . this is tricky. |
| 15:34.59 | eric_hill | so your called party will hear the agent's phone ringing. |
| 15:35.22 | eric_hill | Every call has two legs |
| 15:35.35 | eric_hill | Even if one of those legs is asterisk. |
| 15:35.51 | dnit | No agent is already logged in. I use AgentRequest(agentnum) here. |
| 15:36.43 | dnit | Sorry about that "ring the agent" part |
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| 15:38.31 | eric_hill | An agent being logged in is orthognal to an agent being on the phone |
| 15:40.10 | eric_hill | And until the outside call is delivered to an agent, you aren't going to send anything to an agent. |
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| 15:43.46 | dnit | Ok so a basic question, lets say I called an external endpoint via Dial(PJSIP/${Number}@External_SIP_trunk) , now as soon as the channel for call is created. How can I make my agent spy on this channel. |
| 15:46.27 | eric_hill | Using on-the-phone agents is incompatible with calling Chanspy since no dialplan event occurs. |
| 15:47.58 | eric_hill | I would like to hold the hammer in my left hand and have the nail inserted into the board by using my right hand not holding a hammer. |
| 15:48.34 | [TK]D-Fender | dnit, have them call chanspy.... |
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| 16:22.11 | drmessano | 11:38:32 <eric_hill> An agent being logged in is orthognal to an agent being on the phone |
| 16:22.24 | drmessano | eric_hill: Thanks for reminding me how stupid I am |
| 16:23.59 | eric_hill | ? |
| 16:24.10 | eric_hill | Not calling anyone stupid.... |
| 16:24.58 | Samot | You used a mathy word. |
| 16:25.14 | eric_hill | oh. lol. sorry about that. |
| 16:25.35 | drmessano | eric_hill: I had to google it |
| 16:25.37 | drmessano | Twice |
| 16:25.39 | drmessano | and even then |
| 16:25.41 | drmessano | :( |
| 16:25.42 | eric_hill | takes off propeller hat with "i math" written on it. |
| 16:26.15 | eric_hill | I just meant that those two concepts are two different directions from each other. Logged in or not does not mean on the phone or not. |
| 16:26.34 | Samot | I know what it means. |
| 16:26.38 | drmessano | Had you said "Perpendicular" or "Tacos", I would have been "RIGHT ON!" |
| 16:26.48 | drmessano | But instead |
| 16:26.52 | drmessano | Sadness |
| 16:27.18 | eric_hill | TACOS! Woot! It's about lunch time. |
| 16:27.34 | Samot | I mean having a large and wide vocabulary is awesome. |
| 16:27.35 | drmessano | eric_hill: In the south, we just say "Oh, it's not square" |
| 16:27.48 | Samot | But this is a rule about using big words when little ones can do the job. |
| 16:27.57 | Samot | s/this/there/ |
| 16:28.00 | drmessano | If I said "Orthogonal" to someone, I would be shot |
| 16:28.06 | Samot | ^^^ |
| 16:28.09 | drmessano | or hear "You sassin me, boy" |
| 16:28.10 | Samot | Know your audience. |
| 16:28.18 | eric_hill | https://xkcd.com/thing-explainer/ |
| 16:28.28 | eric_hill | I'll read up. |
| 16:29.09 | eric_hill | I work with a bunch of engineers. They use lots of big words, and an insane number of abbreviations. |
| 16:29.12 | drmessano | No Kindle edition? |
| 16:29.14 | drmessano | WTF |
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| 17:15.35 | lvlinux | I just scored 2 Hex Lite PoE Mikrotik routers for $25 each. :-D Also looking into VyOS/Vyatta as it appears to meet my needs as well. |
| 17:17.42 | learath | That's pretty awesome. |
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| 17:18.36 | Samot | RB750UPr2's ? |
| 17:20.40 | lvlinux | yes |
| 17:21.13 | Samot | Used? |
| 17:21.16 | Samot | Good price. |
| 17:21.34 | lvlinux | yes used |
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| 18:01.31 | sakhi | Hello, is the Linphone application the one that support g729 for free? |
| 18:01.51 | Samot | The softphone doesn't require the license. |
| 18:02.03 | Samot | The PBX/media server does. |
| 18:02.34 | sakhi | Samot: thanks |
| 18:08.48 | [TK]D-Fender | Both do |
| 18:09.01 | [TK]D-Fender | Any device that has to encode or decode is subject |
| 18:09.12 | [TK]D-Fender | (where/when the patent applies) |
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| 18:29.06 | overyander | i recently updated centos on my * box. one of the updates was the kernel... now i'm getting an error when trying to us meetme that the dahdi pseudo device can't be found. |
| 18:29.41 | overyander | the version of asterisk is 11.2.1. do i need to just recompile dahdi and install again or is there something else i can do to fix this? |
| 18:30.16 | overyander | if i need to recompile and install, can i use any current version of dahdi or do i have to use a specific version that's compatible with * 11.2.1? |
| 18:32.34 | rmudgett | You would need to recompile dahdi since dahdi has a kernel module and is sensitive to kernel updates. |
| 18:33.21 | salviadud | overyander, if you are using meetme, but you do not have dahdi devices, you should be using confbridge instead. |
| 18:33.40 | overyander | confbridge != meetme |
| 18:34.06 | salviadud | It's similar... |
| 18:34.20 | overyander | rmudgett, that's what I was thinking. will any version of dahdi work or is there a specific one that will work best with this older version of *? |
| 18:34.45 | [TK]D-Fender | just try the one you have |
| 18:35.05 | rmudgett | I don't know. You could start with the version you have now. |
| 18:35.12 | overyander | i don't have the source of the original one i used. |
| 18:35.48 | overyander | was just going to download the tarball and go through config/make/makeinstall then reboot |
| 18:36.00 | [TK]D-Fender | then go DL another |
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| 19:06.54 | Samot | confbridge > meetme |
| 19:06.59 | Samot | meetme == deprecated |
| 19:07.52 | Samot | So when suggesting confbridge over meetme it is because confbridge != meetme due to it being the replacement and better. |
| 19:09.06 | Samot | 1) It doesn't require DAHDi |
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| 19:33.46 | Samot | file, et al: The schedule graphic needs to be updated to reflect the v15 update. FYI. |
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| 19:36.05 | lvlinux | Is asterisk 15 not going to be an LTS release? |
| 19:36.16 | lvlinux | The download page says "Standard" |
| 19:36.38 | file | it isn't |
| 19:36.44 | file | Samot: comment on the page |
| 19:37.01 | Samot | Ks. |
| 19:37.27 | Samot | Huh. |
| 19:37.35 | Samot | I guess I never created an account |
| 19:37.52 | Samot | Not that I've really needed to submit an Asterisk ticket. |
| 19:40.59 | Samot | Does that mean the LTS on odds is changing or will it be standard until 17? |
| 19:41.47 | [TK]D-Fender | i? |
| 19:42.15 | [TK]D-Fender | pretty sure he just it isn't "standard" |
| 19:42.18 | [TK]D-Fender | Thus is LTS |
| 19:42.30 | [TK]D-Fender | and no indication of that pattern changing so far... |
| 19:42.42 | Samot | v15 was changed from TLS to Standard yesterday |
| 19:42.48 | Samot | In the release schedule. |
| 19:43.14 | Samot | 15.xStandard2017-10 (tentative)2018-10 (tentative)2019-10 (tentative) |
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| 19:43.30 | Samot | er LTS.. |
| 19:43.42 | Samot | Not TLS. |
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| 19:46.16 | drmessano | Yep, and it looks like 13 was extended |
| 19:48.02 | drmessano | I'm good with that.. 13 is the bees knees |
| 19:48.52 | Samot | Yeah, I mean 14 has been pretty solid for me. |
| 19:49.13 | drmessano | Right, but it's not L-T-S |
| 19:49.17 | Samot | But I only use it where I need to use that new Dial string feature. |
| 19:49.22 | drmessano | and we know how important acronyms arwe |
| 19:49.23 | drmessano | and we know how important acronyms are |
| 19:50.08 | drmessano | I prefer to run a solid E-L-T-S-L-O-L stack |
| 19:50.12 | Samot | If 15 is just as solid as 14 then I'll jump to it where needed. |
| 19:52.09 | [TK]D-Fender | I don't recall any branch not being "solid" in vaguely recent times |
| 19:52.17 | [TK]D-Fender | ist just a question of LTS or not |
| 19:52.41 | drmessano | Really it gets no better than the Asterisk-FreePBX-Apache-PHP-MySQL-Enterprise Stack |
| 19:52.45 | drmessano | AFAPME for life |
| 19:53.06 | drmessano | [TK]D-Fender: There is no question. It's been answered |
| 19:53.59 | drmessano | AFAPME LTS = 2019 |
| 19:55.51 | Samot | For what I need 14 to do... |
| 19:55.53 | Samot | It's solid. |
| 19:56.18 | drmessano | Sure |
| 19:56.25 | drmessano | But that's old AFAPME |
| 19:58.14 | Samot | drmessano: |
| 19:58.15 | Samot | http://www.dvd-forum.at/bilder/schnittberichte/686/sb019.jpg |
| 19:59.33 | drmessano | No, that's the old stack |
| 20:00.01 | Samot | That was such a good movie. |
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