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| 00:50.41 | igcewieling | Asterisk and card drivers are the only things I still compile from source. |
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| 03:16.31 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.33.0 (2020/04/30) 16.10.0 (2020/04/30) Standard: 17.4.0 (2020/04/20); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 11:32.30 | Zombie | I found the configure option creating the issue. |
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| 13:02.23 | Ast001 | Hi, can you point me to some source of documentation where I can see how to make asterisk realtime mysql tables in mysql (I can see only sip example in Asterisk Admin book). I'd like to see example of queues, queue_member_table etc... ? |
| 13:02.35 | Ast001 | I try to use version 16 of Asterisk |
| 13:04.29 | Ast001 | Told simple I'd like to copy paste query for creating proper sip_friends, queues and queue_member_table. |
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| 13:35.13 | Kobaz | asterisk 16, you should be using pjsip |
| 13:35.32 | Kobaz | https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 13:35.41 | Kobaz | https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
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| 13:41.30 | Ast001 | I founded that page, but I can't see sql query for creating on example queue_member_table. |
| 13:42.00 | Kobaz | the columns are generally 1:1 with the asterisk configuration |
| 13:42.10 | Kobaz | if the config takes a member=> then the db column will be 'member' |
| 13:45.08 | seanbright | if you download the asterisk tarball the sql scripts are in contrib/ directory... i think |
| 13:45.42 | Kobaz | yeah that too |
| 13:45.57 | Ast001 | Ok I'll check that contrib directory. I installed from Debian package, but I can always dl tar ball and peek into contrib. |
| 13:46.27 | seanbright | contrib/realtime/mysql |
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| 14:17.50 | FuriousGeorge | hi all |
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| 14:18.49 | seanbright | morning |
| 14:18.58 | FuriousGeorge | im a bit confused by followme's behavior. i have a situation where my users are all mobile, and may or may not have their sip clients running on their phones. this is especially true for apple users, since asterisk doesn't really support push notifications, and ios wont just start an app on boot |
| 14:20.01 | FuriousGeorge | i wanted followme to first try the sip extension, then call the mobile number for the user. if i have followme call both the sip and pstn/mobile extensions, the mobile extension will always block the sip client from ringing, so there's no point to that |
| 14:21.02 | FuriousGeorge | but if I have it call the sip extension first, and the user is not logged into it, followme will continue to play the instructions to the user's voicemail. it doesn't seem to realize that a chanunavail and hitting voicemail means you can move on to the next number => already |
| 14:22.13 | igcewieling | If the cellphone answers, then the call was successful. |
| 14:22.24 | Samot | ^^^^ |
| 14:22.25 | FuriousGeorge | yes, but that is step 2 |
| 14:22.46 | FuriousGeorge | first it tries the sip extension, which is preferable, since i don't incur a cost for the 2nd leg of the call |
| 14:22.47 | Samot | Voicemail is an answered called |
| 14:22.49 | Samot | Voicemail is an answered call |
| 14:22.58 | Samot | So to Asterisk, the call was answered. |
| 14:24.07 | FuriousGeorge | right, but from a practical and logistical standpoint, voicemail cannot hit 1 to accept the call, and it would be nice if followme could recognize that the channel is not available and move on, but it sounds like that's not an option |
| 14:24.20 | FuriousGeorge | (even if voicemail could hit 1, you would not want it to) |
| 14:24.34 | Samot | That's not how it works. |
| 14:24.42 | Samot | You must not have something setup right. |
| 14:25.59 | FuriousGeorge | what's not how it works? "Voicemail is an answered call |
| 14:26.24 | Samot | The timeout |
| 14:26.29 | Samot | What is the last step timeout set to? |
| 14:26.35 | Samot | Is it the same as all the rest? |
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| 14:33.55 | FuriousGeorge | Samot: first attempt in followme to the sip extension is 30 seconds, then 60 seconds for the mobile phone |
| 14:34.16 | FuriousGeorge | it seems like what i should do is use followme only for the mobile number, when chan is unavail |
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| 14:35.08 | FuriousGeorge | correction: it's 30 seconds to each endpoint |
| 14:35.34 | FuriousGeorge | s/when chan is unavail/when sip chan is unavail |
| 14:39.50 | Samot | So the user has 30 seconds to make a choice. |
| 14:40.04 | Samot | If they don't make a choice in 30 seconds, they are sent back to the next number. |
| 14:40.40 | Samot | So this is easy to test. |
| 14:41.01 | Samot | You make a call, followme is triggered, you answer the call, do nothing for 30 seconds. What happens? |
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| 14:52.31 | FuriousGeorge | Samot: i called myself (so the sip client did not answer, as i was on my mobile call), and after 28 seconds of no response followme hungup, and the dialplan proceeded to the voicemail step |
| 14:53.08 | Samot | OK, now do the same thing but this time hang up the call at around 20 seconds. |
| 14:53.19 | FuriousGeorge | i looked in followme.conf, and while i can specify a digit timeout (5000 ms) is default, i cannot specifiy an overall timeout |
| 14:53.40 | Samot | Repeat the test but hang up around 20 seconds into the call |
| 14:53.47 | FuriousGeorge | doing it |
| 14:53.50 | Samot | See if followme rolls over to the next step |
| 14:54.02 | FuriousGeorge | i expect it will procede in the dialplan |
| 14:54.37 | FuriousGeorge | and it does |
| 14:55.04 | Samot | So you answer the call, don't press an option and hang up before 20 seconds. |
| 14:55.10 | Samot | It rolls to the next call... |
| 14:55.29 | FuriousGeorge | it goes to the next step in dialplan after followme, which is voicemail |
| 14:55.42 | Samot | But you said it wasn't doing that before |
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| 14:56.02 | FuriousGeorge | annoyingly, even if i hang up before leaving a voicemail, i get a 2 second voicemail from the first followme attempt, where it tries to reach the sip client |
| 14:56.43 | Samot | Because voicemail answered, nothing was done and it rolled over. |
| 14:56.51 | Samot | But a message still got recorded. |
| 14:57.18 | FuriousGeorge | the problem as i was trying to convey it is simply that the first number => followme tries is a sip extension, followed by a mobile number. if that sip extension is not registered, followme does not continue to the mobile number right away |
| 14:57.34 | Samot | Right, it dials the extension |
| 14:57.39 | Samot | Goes to the extension voicemail.. |
| 14:57.42 | FuriousGeorge | it instead has a conversation with the voicemail of the sip extension, which is long enough to leave a 2 second message |
| 14:57.44 | Samot | timesout, rolsl over. |
| 14:57.49 | FuriousGeorge | yeah |
| 14:57.54 | Samot | It's how it's going to work |
| 14:58.00 | FuriousGeorge | that's what i gathered |
| 14:58.02 | Samot | It would do that with a list of mobile phones. |
| 14:58.10 | Samot | That all went to voicemail |
| 14:58.16 | Samot | Each of them would have something in their voicemail |
| 14:58.41 | Samot | Because voicemail _answers_ the call. |
| 14:58.42 | FuriousGeorge | right, but i expected, since followme is an asterisk app, that it would be able to detect the chanunavail and be a bit smarter about it |
| 14:58.49 | Samot | Which triggers the response timeout |
| 14:59.01 | Samot | It did detect it was unavailable. |
| 14:59.05 | Samot | It sent it to voicemail |
| 14:59.08 | Samot | It did its job. |
| 14:59.42 | FuriousGeorge | i was hoping that part of its job would be to tell followme to keep going if chanunavail for a local sip extension, but no big deal if that's not the case |
| 15:00.23 | FuriousGeorge | first ill use stdexten, and on chanunavail or chanbusy, i'll go to followme, which will only attempt to call mobile numbers |
| 15:00.40 | FuriousGeorge | s/if that's/since that's |
| 15:01.29 | Samot | So you set number=>100,25&13135551212,25 |
| 15:01.37 | Samot | But you also set a context |
| 15:01.44 | Samot | So followme is going to 100@context |
| 15:01.53 | Samot | Then 13135551212@context |
| 15:01.57 | FuriousGeorge | if i do that, then the sip call will never get answered. |
| 15:02.02 | Samot | Dude. |
| 15:02.07 | Samot | I'm pointing out how this works. |
| 15:02.13 | Samot | FollowMe doesn't DIAL crap |
| 15:02.24 | Samot | It sends it to a context in the dialplan to look for a MATCH |
| 15:02.32 | FuriousGeorge | because the endpoint is the same device for both extensions, and most if not all phones will not allow the sip client to ring if a pstn call is coming in |
| 15:02.38 | Samot | FollowMe doesn't DIAL crap |
| 15:02.48 | Samot | It sends it to a context in the dialplan to look for a MATCH |
| 15:03.15 | Samot | It's your dialplan that is executing the dial() |
| 15:03.18 | Samot | Not followme. |
| 15:03.28 | Samot | Nothing states that followme has to dial() something. |
| 15:03.28 | FuriousGeorge | ok, fair enough. to be clear im doing number => 100,30 then number => 15555555555,30 |
| 15:04.17 | FuriousGeorge | there is no point to doing number => 100&15555555555,30, for the reason i stated above |
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| 15:05.00 | FuriousGeorge | i didn't mean to misspeak. i have obviously set up contexts for the extensions followme is trying to follow |
| 15:07.32 | Samot | I didn't say that you didn't. |
| 15:07.47 | Samot | You keep claiming followme should know the state of a _local peer_ |
| 15:08.16 | Samot | I'm pointing out that by sending it to a exten@context there is nothing to require it to do such a thing. |
| 15:12.56 | FuriousGeorge | is there anything excluding this as a feature request (not that i'm requesting the feature, as I can just use stdexten, as I pointed out) |
| 15:22.40 | igcewieling | it might be easier to code it all in the dialplan. |
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| 17:51.07 | Kobaz | ahahaha |
| 17:51.36 | Kobaz | customer wants a 'nice easy to memorize toll free 800 number like 800-900-9999 or similar' |
| 17:51.52 | Kobaz | I had to tell him prepare to shell out like $100k or more |
| 17:52.56 | Kobaz | Unless he wants 844, 866, etc etc |
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| 19:02.25 | froy | A PeerStatus event is still sent with pjsip if any of the Contacts come online or if all go offline -- even if pjsip doesn't really call them "peers" anymore. I like that, even though DeviceStateChange reports the same info. The thing is, I can no longer induce getting a PeerEntry.Status for all peers anymore with SIPPeers. Is there an AMI command for pjsip that can induce sending a PeerStatus for all peers like SIPPeers used to? Is PJSIPShowEndpoints |
| 19:02.25 | froy | the closest equivalent to SIPPeers? |
| 19:03.47 | Kobaz | yeah probably |
| 19:18.28 | igcewieling | "pjsip show contacts"m might be what you are looking for. |
| 19:24.04 | froy | I'm talking AMI commands, so that'd be PJSIPShowContacts -- problem is, that doesn't represent the peer, it represents any number of registrations that could be for the same "peer" |
| 19:33.08 | igcewieling | "pjsip show aors" |
| 19:35.03 | froy | PJSIPShowAors does not show its state/availability -- have have to dig into detail other detail commands one by one to get them. |
| 19:35.52 | froy | s/detail other detail/other detail/ |
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| 19:43.38 | igcewieling | run the cli command from AMI. |
| 19:43.54 | igcewieling | I see the state: https://pastebin.com/zg5prrtp |
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| 19:57.28 | froy | I'm not sure what you mean by running a cli command from AMI. The "pjsip show aors" is different from the AMI PJSIPShowAors in that PJSIPShowAors does not show the device's availability. It does give a comma-separated list of contact URLs though. |
| 20:11.07 | seanbright | i think igcewieling means: |
| 20:11.08 | seanbright | *CLI> manager show command Command |
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| 20:17.36 | igcewieling | Indeed, I do. |
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| 21:01.07 | Kobaz | Really pjsip needs like a 'peers' type thing |
| 21:01.17 | Kobaz | that combines Endpoint/contact/aor into a master list |
| 21:01.27 | Kobaz | I've been working on writing the code for that |
| 21:02.01 | Kobaz | Not sure what to call it, but basically it'll be similar to sip show peers to make people comfortable in switching to pjsip |
| 21:02.08 | froy | PJSIPShowEndpoints still seems to be the least expensive way to get what used to be provided by SIPPeers. Other than that, it's climbing a big relational tree. |
| 21:02.16 | Kobaz | yeah |
| 21:02.32 | Kobaz | Well, no matter what, you're going to have a tree, given the pjsip design |
| 21:02.47 | Kobaz | Since contacts can be used across endpoints and all that fun stuff |
| 21:02.48 | froy | Right, my beef with switching to pjsip isn't configuration -- that's easy. It's all my integration code with AMI that hurts. |
| 21:02.53 | Kobaz | yeah |
| 21:03.08 | Kobaz | all the concepts are changed, so there's no easy dropin |
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| 21:04.26 | froy | The changes make sense though. It's just work. :D |
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| 21:08.07 | froy | I wish "Not in use" device state could have just been named "Idle" ;) But that's just the anal programmer in me who doesn't like to name states Not-something. |
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| 21:48.17 | igcewieling | No! peers are evil! down with peers! |
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| 23:35.12 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:104a:f28a:68c3:ab48:87ea:28b1) |
| 23:52.13 | froy | Sigh, ContactStatus event does not have a ObjectName like ContactList does. |
| 23:53.49 | froy | Is the URI reliable enough to key on? I hope random parameters don't change willy-nilly so I can find it. |