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| 00:02.33 | Samot | So I'm confused. |
| 00:02.47 | Samot | If they send them to voicemail, then use the GOTO: option |
| 00:03.05 | Samot | If they hangup on them, also use GOTO and send them to a hangup |
| 00:03.41 | ShaunR | I could, i'm trying to not have to hardcode things. Since in freepbx i set a timeout destination for the queue to a voicemail box i'd like to just timeout that caller so that they hit the correct mailbox that was setup in freepbx. |
| 00:08.52 | Samot | OK so setup the timeouts properly then. |
| 00:09.44 | ShaunR | The timeouts are set properly, if the caller is in the queue too long they are sent to vm. |
| 00:09.54 | Samot | What is too long? |
| 00:10.03 | ShaunR | 5 minutes. |
| 00:10.18 | Samot | And what's the retry? |
| 00:10.24 | Samot | and what's the timeout to the agent? |
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| 00:11.40 | ShaunR | agent timeout is 2 minutes. |
| 00:15.59 | Samot | And the retry? |
| 00:19.41 | ShaunR | 1 |
| 00:19.55 | ShaunR | well freepbx says 0, but retry=1 in the conf. |
| 00:25.21 | Samot | So 1 second. |
| 00:25.51 | Samot | I dont think you can adjust the timeout once they enter the queue. |
| 00:29.39 | Samot | You might be stuck with the gosub result |
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| 00:58.34 | ShaunR | I need to look at how freepbx handles the timeout (reading the extension code is painfull) but maybe I could return them to the queue context "t" extension. |
| 01:04.15 | Samot | It's queues. |
| 01:04.18 | Samot | It doesn't change. |
| 01:04.41 | Samot | The timeout in .conf is how long to ring an agent. the retry is how long to wait to retry agents. |
| 01:05.03 | Samot | The timeout set in queues() is how long the caller sits in the queue before being kicked. |
| 01:05.26 | Samot | App timeout is checked between conf timeout and retry |
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| 13:49.51 | DanFromUK | Hi. I just realised that Queue Members don't work when using multiple PJSIP contacts. For non-queue calls, I use PJSIP_DIAL_CONTACTS. But is there a way to do that with Queue Members when using Realtime queues? |
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| 13:53.02 | file | there is no built in mechanism, only using a Local channel is it done |
| 13:54.40 | DanFromUK | Ok, I think Local channels might work. I'm already using those for reaching public mobile phones that are queue members. I'll give it a test. Thanks. |
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| 14:11.30 | DanFromUK | Is there any way to use the linear queue strategy without requiring a restart in v13 ? |
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| 15:50.44 | Samot | Should I be shocked that after almost 8 years people still think providers need to support chan_pjsip on their side for it to work? |
| 15:51.56 | igcewieling | People still think the earth is flat. |
| 15:52.04 | file | SIP it, SIP it real good |
| 15:55.22 | Samot | igcewieling: Well I'd be just as unimpressed by atlas makers that didn't know the Earth was round. |
| 15:55.54 | Samot | I would have to question any world map that had "Edge of World" on it in 2021. |
| 15:56.30 | igcewieling | file is an asterisk Devo-tee? |
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| 18:20.37 | alrs | I saw mention of a new Asterisk website go by somewhere today |
| 18:20.54 | alrs | it brought me back here for old times' sake |
| 18:21.05 | alrs | and I went to go look at the site |
| 18:21.11 | alrs | derrrrp. http://digium.com/ |
| 18:22.08 | alrs | (found my way to asterisk.org, still laughed) |
| 18:22.23 | Samot | I don't think it was ever digium.com |
| 18:23.20 | alrs | It's been over a decade since I needed to care, I'm sure you're right. |
| 18:23.34 | Samot | Oh, wait. |
| 18:23.36 | Samot | JFC. |
| 18:24.22 | clarjon1 | www.digium.com redirects to sangoma.com, but it looks like the forwarder was skipped for the non-www subdomain |
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| 19:06.52 | nbjoerg | Samot: well, given that there are devices that don't work with chan_pjsip... |
| 19:07.28 | file | such as/ |
| 19:07.29 | file | ? |
| 19:07.55 | nbjoerg | file: snom m(3)25 :( |
| 19:08.25 | Samot | nbjoerg: That is entirely false. |
| 19:08.49 | Samot | nbjoerg: PJSIP has been around for almost as long as Asterisk. It is a SIP driver used in software and hardware SIP applications/devices. |
| 19:09.01 | file | except for the use of tel URIs, every time I've seen someone having problems with a device it's been configuration |
| 19:09.08 | Samot | ^^^^^^ |
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| 19:11.06 | Samot | PJSIP is a SIP driver that has been around for over a decade and the Asterisk project ported in as opposed to building another SIP driver like they did with Chan_SIP or rebuilding Chan_SIP from the ground up. |
| 19:11.36 | Samot | So this "providers don't support Chan_PJSIP" mantra is completely false. |
| 19:12.39 | nbjoerg | given that there is a lot of commercial software around that only supports Internet Explorer: what has age to do with willingness to support something? |
| 19:17.03 | Samot | nbjoerg: You only took part of the point away from that. |
| 19:17.24 | Samot | nbjoerg: It's been around for over a decade and been used in software and hardware SIP applications/devices. |
| 19:17.51 | Samot | So there's a high chance that many have connected to providers with PJSIP and not known it. |
| 19:19.43 | nbjoerg | anyway, I can relate to FuriousGeorge's problem - I have the very same issue. I run out of time debugging it and left it for now... |
| 19:20.04 | Samot | Well the problem is most likely a configuration issue. |
| 19:21.26 | nbjoerg | *shrug* the same config on the phone side works with chan_sip and the asterix side works with a different snom model; so "it's a config issue" is IMO somewhat insulting hand-waving |
| 19:21.42 | Samot | It's not. |
| 19:21.47 | Samot | It's actual experience. |
| 19:22.03 | Samot | But you want to lean on it being "chan_pjsip" is unsupported by providers. That's fine. |
| 19:22.08 | Samot | It's wrong, but fine. |
| 19:22.50 | igcewieling | nbjoerg: don't worry, chan_sip is going away |
| 19:22.55 | nbjoerg | igcewieling: I know |
| 19:23.29 | nbjoerg | igcewieling: I wouldn't care if it worked... |
| 19:23.41 | Samot | Well let's hope the providers catch up by then. |
| 19:23.49 | seanbright | did you post some errors or logs somwhere? |
| 19:23.57 | nbjoerg | seanbright: I did, it's been a while |
| 19:23.58 | seanbright | i feel like i'm missing the front end of this conversation |
| 19:24.01 | Samot | nbjoerg: It works. If you're having actual problems please describe them and show some debugs. |
| 19:25.15 | seanbright | if there are legitimate interoperability problems that are 100% on asterisk, obviously we would want to fix those |
| 19:25.25 | nbjoerg | at the very least "we don't support XXX" in the literal sense often means "we don't want to help you if you have problems"; assuming that everything works all the time is just ivory tower mentality that has little backing in reality... |
| 19:25.37 | Samot | I've gotten Polycom, Yealink, Yeastar, SNOM, Cisco SPA series, Matrix PBXes, Asterisk, FreePBX and others. |
| 19:25.44 | Samot | All working on chan_pjsip. |
| 19:25.46 | nbjoerg | I'm not even saying that they are bugs on the asterix/pjsip side |
| 19:26.11 | nbjoerg | the fun part with interop issues is that there are two sides to blame and that doesn't make things communicate either |
| 19:26.29 | Samot | nbjoerg: I've done interopping of PBXes and phone devices. |
| 19:26.35 | Samot | nbjoerg: I work at providers, it's part of what I do. |
| 19:26.39 | Samot | So yes, this works just fine. |
| 19:26.42 | seanbright | so... is there an actual problem? |
| 19:26.47 | Samot | ^^ |
| 19:27.02 | seanbright | what are we talking about? |
| 19:27.20 | nbjoerg | seanbright: enough of a problem that I'm still running with chan_sip for now until I have time to setup a debug version of asterisk to trace the invite calls |
| 19:27.28 | seanbright | ok |
| 19:27.58 | nbjoerg | Samot: you don't have a support account for snom btw? I have an issue with the phonebook feature in newer firmware generations |
| 19:28.10 | Samot | No, I don't. |
| 19:28.16 | Samot | I don't like snom's. |
| 19:28.36 | nbjoerg | grumbles about companies that don't want bug reports by only dealing with integrators |
| 19:32.20 | nbjoerg | https://dpaste.org/pPO7 that was my chat log from when I looked at it |
| 19:32.40 | Samot | Well if you've ever used MicroSIP or csipsimple, you've used PJSIP. |
| 19:33.37 | Samot | Oh wait, I remember this |
| 19:33.41 | Samot | This is an issue with one phone. |
| 19:34.28 | Samot | Is that snom on the same network? |
| 19:34.35 | nbjoerg | yes |
| 19:34.49 | Samot | I would need to see a real sip debug of this. |
| 19:35.13 | Samot | Because based on that convo, you call from the snom it sends back a register to the auth challenge but no reply. |
| 19:35.34 | Samot | So need to see what happens when it replies to the auth challenge. |
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| 19:35.53 | Samot | But if it can receive calls, per this convo, I doubt it's a pjsip issue. |
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| 19:36.22 | nbjoerg | well, something(TM) was completely ignoring some of the packets |
| 19:36.36 | Samot | Need to see an outbound call in action. |
| 19:37.12 | Samot | Because it seems like it can register just fine and accept calls just fine. |
| 19:37.19 | Samot | So it's something with not being able to send calls. |
| 19:37.35 | nbjoerg | as I said, need time to set up a test bed so that I don't have to kill the main pbx :) |
| 19:37.49 | nbjoerg | and spare time was rare the last few month |
| 19:37.57 | nbjoerg | "works for now" |
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| 20:42.20 | ShaunR | Hmm... so this gosub that runs on the dialed extension has a background() followed by a WaitExten. The issue i'm seeing is that if i press a number while in background the order flows correctly and it moves onto the extension that was pressed. If I wait for background to finish and try and hit the same key during WaitExten(5) the gosub exits and I see 'Abnormal 'Gosub(screen-callee,s,1(s,1))' |
| 20:42.20 | ShaunR | exit. Popping routine return locations.' |
| 20:43.25 | ShaunR | I noticed in the wiki it says that waitexten doesnt function as expected inside macros. I'm wondering if that is the same case with gosubs |
| 20:43.30 | igcewieling | ShaunR: what dial option are you using to trigger the gosub? |
| 20:43.56 | ShaunR | igcewieling: exten = _X.,n,Dial(PJSIP/${EXTEN},30,U(screen-callee,s,1)) |
| 20:45.11 | igcewieling | I suspect waitexten won't work in U. "You cannot use any additional action post answer options in conjunction with this option. Also, pbx services are run on the peer (called) channel, so you will not be able to set timeouts via the TIMEOUT() function in this routine." |
| 20:45.18 | ShaunR | wondering if i should use read instead. |
| 20:50.43 | ShaunR | gonna give that a try i guess. |
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| 21:25.33 | ShaunR | Well so far so good. Have to stop for a bit so to be continued; |
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| 21:47.55 | igcewieling | LOL! "The 'S' in IOT stands for Security." |
| 21:58.52 | ketas | hahaha |
| 22:00.04 | ketas | in addition to s, it needs d, and second i |
| 22:06.55 | Samot | Huh. |
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