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| 00:18.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); 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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| 11:58.47 | dritskalle | pjsip - is there an equivilant to the last line of chan+_sip "sip show peers" to get a total/off/on? i cant find anything.. parsing or event/log/update seems like lots of work |
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| 12:07.07 | Samot | pjsip show endpoints |
| 12:07.11 | Samot | pjsip show aors |
| 12:07.15 | Samot | There are quite a few. |
| 12:16.37 | Samot | Am I missing something? I'm have a switch send back a 408 after 1XX replies and Asterisk treats it like a 408 before 1XX replies and considers it "Busy/Congested". If I send back a 480 which Asterisk has listed as "No Answer from User", same issue. Asterisk considers it BUSY. I'm sure because 480 means BUSY/DND. |
| 12:17.08 | Samot | So I can I get Asterisk to just accept a standard "No Answer" without the Dial() timer expiring? |
| 12:17.18 | Samot | -I |
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| 12:19.29 | Samot | I'm also kinda finding it where that Asterisk also considers a 483 as a "No Answer from user (user alerted)". |
| 12:19.40 | Samot | 483 is Too Many Hops, it means it never made it to the user. |
| 12:19.45 | Samot | So the user would not have been alerted. |
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| 12:33.19 | dritskalle | Samot: yeah buut pjsip show/list endpoints/contacts dont give totals.. seems dirty having to parse output |
| 12:34.57 | Samot | Well Endpoints probably isn't the right choice. |
| 12:35.09 | Samot | You can have 1 endpoint but 20 contacts. |
| 12:35.27 | Samot | With PJSIP you need to look at the contacts an AOR has. |
| 12:35.34 | Samot | Chan_SIP is 1:1 |
| 12:36.11 | dritskalle | in my case all contacts are 1 for 1.. i can do list contacts for online users and list endpoints for all users - then a bunch of manual parsing stuff wich is rather annoying |
| 12:36.18 | Samot | OK |
| 12:36.34 | Samot | The fact you are using it 1:1 doesn't mean it should automagically just report on that |
| 12:36.39 | Samot | It's a 1:many thing. |
| 12:36.46 | Samot | Look at the other PJSIP options. |
| 12:36.57 | Samot | pjsip show <tab> |
| 12:37.12 | dritskalle | i am just looking for a quick way to get counts.. all users, online/offline users |
| 12:37.39 | dritskalle | this application only allows one contact per endpoint - with replace |
| 12:38.01 | Samot | OK. |
| 12:38.07 | Samot | That doesn't matter. |
| 12:39.33 | dritskalle | the output of the pjsip show commands are annoyingly formatted for parsing |
| 12:39.50 | Samot | OK. |
| 12:39.55 | Samot | That is what you have to work with. |
| 12:40.00 | [TK]D-Fender | Go look at AMI then |
| 12:40.01 | Samot | Or AMI commands |
| 12:40.21 | Samot | Asterisk is a tool kit. |
| 12:40.28 | Samot | It's up to those using it to do what they want with it. |
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| 12:42.05 | Samot | pjsip show aors will show all the AORs and how many online contacts they have. |
| 12:42.06 | dritskalle | actually - maybe AMI is a good idea |
| 12:42.25 | Samot | pjsip show contacts will output all the online contacts with their AOR |
| 12:43.30 | Samot | I don't think this is very hard. |
| 12:43.34 | Samot | You have a 1:1 setup. |
| 12:43.43 | Samot | So how many Endpoints/AORs do you have? |
| 12:43.53 | Samot | Vs. how many contacts are online. |
| 12:44.26 | Samot | AMI or using the CLI commands, you're parsing data that is outputted to you. |
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| 12:47.41 | dritskalle | i never used AMI - was assuming it format the data in some structured way or no? |
| 12:52.23 | [TK]D-Fender | Don't assume |
| 12:52.34 | [TK]D-Fender | That's why they invented documentation |
| 12:52.51 | [TK]D-Fender | And actual usage + eyes |
| 13:04.09 | dritskalle | relevant docs not all that easy to find.. did a test with ami PJSIPShowEndpoints and it is structured better than show commands for sure |
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| 13:11.37 | [TK]D-Fender | Yeah docs are a little lighter, so that's what testing is for. Every command has its own way of presenting. AMI doesn't truncate like the "show" commands often do, and gets you more info typically. This is so you can more properly scan the activity on your server to have something solid to act on. "Show" commands are often quick dumps for basic stuff so the focus was on other avenues |
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| 13:12.56 | file | AMI is for applications, CLI is for human eyes |
| 13:13.23 | [TK]D-Fender | CLI sometimes comes with its own spoon ;) |
| 13:13.38 | dritskalle | :p |
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| 13:34.00 | ntt | Hello, it is possible to read all extensions in a ring group using agi (phpagi)? (Obviously I know the ring group number) |
| 13:35.36 | sibiria | are ring groups an asterisk thing at all? |
| 13:37.14 | sibiria | if your "group" is just a set of channels stored in a variable, you can obviously read that variable over AGI |
| 13:39.05 | sibiria | if it's implemented via the DIALGROUP function you should be able to read that as well |
| 13:39.16 | sibiria | (at least i think you can read a DIALGROUP) |
| 13:41.58 | [TK]D-Fender | ntt, No. |
| 13:42.18 | [TK]D-Fender | actually .... depends on a point of view |
| 13:43.15 | [TK]D-Fender | AGI offers nothing in terms of interfacing with Asterisk. Ring Groups are a dialplan construct invented by FreePBX's GUI. The closes you could get is to use your script to directly look at the FreePBX DB tables and process that |
| 13:43.44 | ntt | [TK]D-Fender: thank you. The problem is that I'd like to avoid one db query each time a call arrives |
| 13:44.06 | [TK]D-Fender | Then code accordingly |
| 13:44.32 | ntt | I have to notify through an api when a call arrives and which phones are ringing |
| 13:44.57 | [TK]D-Fender | That'll be trickier.... |
| 13:45.10 | [TK]D-Fender | especially if you have concurrent phone calls trying to ring |
| 13:45.12 | ntt | my idea is to use a custom destination (with freepbx) as inboud route and implement [my-trigger] in extensions_custom.conf |
| 13:45.20 | ntt | is this approach correct? |
| 13:45.35 | [TK]D-Fender | I could dial 400 and while it's ringing another call comes in and hits a ring group that 400 is a member of. |
| 13:45.43 | [TK]D-Fender | I can't assume the ring-group is why |
| 13:46.55 | [TK]D-Fender | This sounds like a solid case for custom dialplan tagging the created channels so they can be tracked and an AMI tracker for the ringing status capture |
| 13:47.30 | ntt | ok... but I'm trying to "store" the channel to I know when a call is answered or when a call hangs up without answer (example: RECEIVING CALL FROM xxx on CHANNEL SIP/conv-0828xxxxxxx-in-00000062) |
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| 13:52.56 | [TK]D-Fender | I just gave you the path |
| 13:57.00 | [TK]D-Fender | You might be able to avoid AMI if you aren't looking to see it in run-time, only logged results. This would be by dialing nested local channels with macros on answer and having the local channel track immediate dial failures (phone DND, call setup failure) that prevented ringing, etc |
| 13:57.16 | [TK]D-Fender | Doesn't actually confirm they rang though |
| 13:57.29 | [TK]D-Fender | only if enough time passed before someone answered. |
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| 14:21.50 | lonepurplesmurf | What keys do I need to receive to verify this? http://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-13.18-cert3.tar.gz.asc |
| 14:21.52 | lonepurplesmurf | thanks |
| 14:33.13 | Samot | lonepurplesmurf: Do you have a support contract or any sort of an agreement with Digium? |
| 14:33.13 | [TK]D-Fender | You shouldn't be running Cert unless you're paying Digium for a support contract |
| 14:33.20 | Samot | If the answer is no, then cert is the wrong choice. |
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| 14:35.56 | Sepultura | Hallo by default Asterisk is using UDP? |
| 14:36.16 | Samot | Depends on what it is doing. |
| 14:36.23 | Samot | For SIP, yes. |
| 14:36.30 | Sepultura | and IAX? |
| 14:36.34 | Samot | Yes. |
| 14:36.40 | [TK]D-Fender | IAX is only UDP |
| 14:36.45 | Samot | Because those are the standards. |
| 14:36.46 | Samot | And that. |
| 14:37.56 | Sepultura | afaik FreePBX let you choose SIP over TCP |
| 14:38.11 | Samot | Yes. |
| 14:38.14 | Samot | You could. |
| 14:38.22 | Samot | Because SIP works over TC |
| 14:38.24 | Samot | Because SIP works over TCP |
| 14:38.31 | Samot | But that is not the standard default. |
| 14:39.14 | Samot | You can do SIP over TCP, TLS, WS, WSS |
| 14:40.53 | Samot | But that's just the signalling. |
| 14:40.58 | Samot | The RTP is always UDP. |
| 14:40.59 | Sepultura | TLS is a transport layer? |
| 14:41.14 | Samot | Yes. |
| 14:41.20 | Samot | Just as it always has been. |
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| 14:42.47 | Sepultura | RDP is TL too :O |
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| 14:59.16 | lonepurplesmurf | Samot No, I just wanted to be able to verify the files are signed. thanks |
| 14:59.36 | Samot | Signed? |
| 14:59.48 | Samot | It regular stable release is just fine. |
| 14:59.55 | Samot | s/It/The |
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| 15:09.44 | lonepurplesmurf | I linked to a signature file. Someone signed it. It proves the file is by the authors. I was wanting to know their public keys so I can verify it. |
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| 15:21.18 | Samot | If you need all that then you need to call Digium and get a contract. |
| 15:21.33 | Samot | Otherwise the stable releases of 13 or 15 are just fine. |
| 15:23.07 | Samot | Asterisk/Digium is just Josh or Sean or Mark... |
| 15:23.41 | Samot | It's a corporation. This isn't some app you're downloading from a random website. |
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| 15:33.40 | lonepurplesmurf | Lol...that's now how digital signatures work. |
| 15:33.53 | lonepurplesmurf | It's not "all that". They sign it for a reason. |
| 15:34.22 | lonepurplesmurf | If you're in an environment where you're concerned about security like a remote part of the world, you want to check those. |
| 15:34.50 | lonepurplesmurf | Anyhow thanks. They publish them somewhere. I'll just have to figure out where. |
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| 17:48.34 | peacetreaty | i have a strange problem. dahdi is loaded, but /proc/dahdi is empty. and dahdi_scan doesn't show anything too. dahdi_hardware and dahdi_span_assignments list both give me results. does anybody have a clue? |
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| 18:07.14 | dritskalle | ..I am detective John Kimble... ..who is you dahdi and what does he do? |
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| 18:57.08 | nny | we are testing what happens when a client goes UNREACHABLE and asterisk seems to keep the channel open when one side of the channel goes offline. Any settings to ensure this scenario causes asterisk to end the open channel? |
| 18:59.26 | file | There are options in both SIP channel drivers which can use lack of flowing RTP to terminate the call, or session timers which send SIP messages and wait for a response |
| 18:59.52 | nny | @file thanks, I'll look for that option |
| 19:00.00 | file | Going unreachable itself doesn't terminate active calls |
| 19:00.20 | nny | @file got it thanks |
| 19:02.00 | nny | rtptimeout |
| 19:02.03 | nny | thanks |
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| 20:51.51 | damania | hi i am on freepbx using bria. the xmpp username is different than the extension so the setup is not good. i want to be able to txt and call on the same extension |
| 20:54.38 | damania | is there any easy solutions? |
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| 22:31.44 | dritskalle | pjsip - is there command (cli or ami) to get a contacts current time-to-expiration? with chan_sip it was ship show peer peername |
| 22:32.04 | dritskalle | pjsip show contact contactname does not list that info |
| 22:35.05 | dritskalle | time to incoming register expiration that is |
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| 22:49.23 | dritskalle | found one i can use.. ami PJSIPShowRegistrationInboundContactStatuses has a Regexpires: utime |
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