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| 05:51.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.0, 16.16.0 (2021/01/21) Final Bugfix: 13.38.1, 17.9.1 (2020/12/22); 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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| 15:39.46 | nbjoerg | I'm trying to get a snom m25 to play nicely with pjsip in asterisk 16, but calling e.g. a simple moh extension fails |
| 15:40.23 | nbjoerg | what I see in the logs is that it sends the first invite, gets a 401 and sends another invite |
| 15:41.05 | nbjoerg | and the second invite seems to just hang in the air according to the phone's log? |
| 15:41.42 | nbjoerg | what also confuses me is that the m25 will try to register first, which is the d725 doesn't do |
| 15:43.30 | grummund_ | mbecroft: > pjsip set history on |
| 15:43.39 | grummund_ | mbecroft: > pjsip show history |
| 15:43.56 | grummund_ | mbecroft: > pjsip show history entry <n> |
| 15:45.18 | nbjoerg | history and pcap agree that pjsip is ignoring the authorized invite? |
| 15:45.53 | grummund_ | the response to the 2nd invite? |
| 15:46.01 | nbjoerg | there is no response |
| 15:48.04 | grummund_ | that's beyond my limit i'm afraid. |
| 15:53.37 | Samot | nbjoerg: What does Asterisk actually show? |
| 15:56.17 | nbjoerg | https://dpaste.org/2JH7 <-- this is the minimally sanitized packet trace |
| 15:57.11 | file | do other endpoints work with pjsip? |
| 15:57.34 | nbjoerg | yes, I have a different phone that works fine |
| 15:57.44 | nbjoerg | I can even call the m25 from that one, but not the other direction |
| 16:11.50 | nbjoerg | TCP vs UDP doesn't make a difference either |
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| 17:09.28 | nbjoerg | so, any way to get pjsip to tell me why it doesn't handle the message? |
| 17:15.19 | letsgo | hi guys, is it possible to see silence in wireshark with the payload? |
| 17:15.21 | letsgo | https://prnt.sc/xk3kw4 |
| 17:15.28 | letsgo | looking at this screenshot |
| 17:15.50 | letsgo | there was no audio on those packets, only silence |
| 17:20.00 | nbjoerg | letsgo: silence supression is normally used |
| 17:20.37 | nbjoerg | so normally "no" |
| 17:25.53 | ckb | okay, pjsip show contacts doesn't show "inactive" contacts, how can I get a better formatted list rather than pjsip show endpoints? |
| 17:29.33 | Samot | What do you mean by "inactive"? |
| 17:31.30 | ckb | Samot, no devices registered on it |
| 17:32.05 | letsgo | nbjoerg then it is possible to see the silence supression as it is static |
| 17:32.10 | Samot | Why would there be a list of those in the contacts? |
| 17:32.14 | letsgo | nbjoerg like in my screenshot |
| 17:33.49 | ckb | Samot, I'm just trying to get a list of PJSIP "extensions" |
| 17:33.51 | Samot | ckb: An Endpoint can be associated to multiple AoRs which in turn could have N amount of contacts under them. |
| 17:34.23 | Samot | ckb: No such thing. |
| 17:35.20 | Samot | ckb: An Endpoint *does not* have to be associated to an AoR. |
| 17:35.24 | nbjoerg | letsgo: silence is supressed by not sending data |
| 17:35.43 | Samot | ckb: But you could use X endpoint to call A, B, C and D AoRs. |
| 17:37.39 | ckb | Samot, this I know (I don't know what AoR is, but I get it's 1:n) |
| 17:37.58 | ckb | I just want a LIST of them. |
| 17:38.21 | ckb | Whether 1 device is online or not. |
| 17:39.05 | Samot | pjsip show endpoints will show you all your configured endpoints |
| 17:39.19 | Samot | It will show you their related AoRs and any contacts under said AoRs. |
| 17:39.36 | ckb | Samot: can you explain what AoR means? |
| 17:39.43 | Samot | You can do: pjsip show aors and that will show you all the configured AoRs and their online/static contacts. |
| 17:39.50 | Samot | Address of Record. |
| 17:40.00 | Samot | 200@domain |
| 17:40.12 | Samot | 200@domain can have N contacts registered to it. |
| 17:40.49 | ckb | Right, I understand this all... I think this is much easier to parse than show endpoints |
| 17:42.12 | ckb | someone told me if I'm trying to use machine to parse I should use AMI (but I'm already too deep into parsing the asterisk CLI at this point) |
| 17:42.54 | Samot | I'm not sure why you would do that when AMI or ARI could provide easier methods. |
| 17:43.48 | ckb | I have only known of the CLI for a short period |
| 17:45.21 | ckb | if you have any beginner resources for the AMI (no idea what ARI is; rest API?) I would love them |
| 17:49.49 | ckb | I thought my only choice was the CLI, but the AMI and ARI are only available to what versions of asterisk? |
| 18:03.30 | Samot | Well AMI has been around forever. ARI since like Asterisk 12. |
| 18:04.12 | ckb | Samot: my boss has no idea that he needs to give me this information. |
| 18:06.32 | Samot | ckb: What are you try to do with all this? |
| 18:07.34 | ckb | Samot: it's not what I'm trying to do. |
| 18:08.46 | ckb | My boss has 100 PBX servers. I need access to all information (call logs, firewalls, extensions, etc). I'm building a VoIP web application. |
| 18:09.37 | Samot | Ahhh. |
| 18:09.44 | ckb | I need to create/edit/delete dialplans, but I have (after 1 year of being on board; after doing SMS) before he told me about CLI/ARI |
| 18:09.55 | ckb | so I need a CRASH COURSE |
| 18:10.27 | ckb | I didn't know SIP/PJSIP was 1:1 and 1:n respectively until recently |
| 18:10.47 | ckb | I have just now learned*** |
| 18:11.22 | ckb | he told me there wasn't documentation on the ARI so he thought it was "outdated" |
| 18:12.19 | ckb | He told me that there was just now docs on the rest and management after 16, when all of ours are 13+ |
| 18:13.21 | ckb | the CLI gave me a "omfg why didn't you tell me about this to begin with" moment. |
| 18:13.40 | ckb | now I learn I have a rest API I can use to do these things. |
| 18:13.53 | ckb | HELP |
| 18:30.10 | drmessano | https://wiki.asterisk.org |
| 18:44.41 | Samot | Wait, what? |
| 18:44.49 | Samot | Your boss thought that ARI was outdated? |
| 19:17.20 | Kobaz | ARI is new and improved |
| 19:17.29 | Kobaz | 33% more new than your leading new and improved |
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| 20:31.47 | igcewieling | If anyone has a PCAP of a connected line update sent after CONNECTEDLINE(name) is set, please /msg me. |
| 20:43.38 | Kobaz | speaking of connected line updates |
| 20:43.52 | Kobaz | i didn't dig deep into this one yet, but how do you disable them per-call |
| 20:45.02 | Kobaz | Dial with 'I' will disable updating from connected line updates received from the peer/endpoint you're talking to, but it doesn't stop asterisk from sending them |
| 20:45.44 | Kobaz | many devices/carriers will return a 500 internal server error when sending connected line updates |
| 20:45.58 | Kobaz | and then others will actually drop the call if you send something that causes a 500 |
| 20:50.01 | igcewieling | best I imagine you could do is disable per endpoint (i.e. disable it on your carrier trunks) |
| 20:50.37 | igcewieling | I didn't have send_pai enabled on my test endpoint, so it wasn't sending out the updates. |
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