IRC log for #asterisk on 20210414

00:00.59*** join/#asterisk Janos (~textual@201.204.94.76)
00:02.33SamotSo I'm confused.
00:02.47SamotIf they send them to voicemail, then use the GOTO: option
00:03.05SamotIf they hangup on them, also use GOTO and send them to a hangup
00:03.41ShaunRI 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.
00:08.52SamotOK so setup the timeouts properly then.
00:09.44ShaunRThe timeouts are set properly, if the caller is in the queue too long they are sent to vm.
00:09.54SamotWhat is too long?
00:10.03ShaunR5 minutes.
00:10.18SamotAnd what's the retry?
00:10.24Samotand what's the timeout to the agent?
00:11.08*** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir)
00:11.40ShaunRagent timeout is 2 minutes.
00:15.59SamotAnd the retry?
00:19.41ShaunR1
00:19.55ShaunRwell freepbx says 0, but retry=1 in the conf.
00:25.21SamotSo 1 second.
00:25.51SamotI dont think you can adjust the timeout once they enter the queue.
00:29.39SamotYou might be stuck with the gosub result
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00:58.34ShaunRI 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.
01:04.15SamotIt's queues.
01:04.18SamotIt doesn't change.
01:04.41SamotThe timeout in .conf is how long to ring an agent. the retry is how long to wait to retry agents.
01:05.03SamotThe timeout set in queues() is how long the caller sits in the queue before being kicked.
01:05.26SamotApp timeout is checked between conf timeout and retry
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13:49.51DanFromUKHi. 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?
13:52.30*** join/#asterisk mvanbaak (~mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak)
13:53.02filethere is no built in mechanism, only using a Local channel is it done
13:54.40DanFromUKOk, 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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14:11.30DanFromUKIs there any way to use the linear queue strategy without requiring a restart in v13 ?
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15:50.44SamotShould I be shocked that after almost 8 years people still think providers need to support chan_pjsip on their side for it to work?
15:51.56igcewielingPeople still think the earth is flat.
15:52.04fileSIP it, SIP it real good
15:55.22Samotigcewieling: Well I'd be just as unimpressed by atlas makers that didn't know the Earth was round.
15:55.54SamotI would have to question any world map that had "Edge of World" on it in 2021.
15:56.30igcewielingfile is an asterisk Devo-tee?
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18:20.37alrsI saw mention of a new Asterisk website go by somewhere today
18:20.54alrsit brought me back here for old times' sake
18:21.05alrsand I went to go look at the site
18:21.11alrsderrrrp. http://digium.com/
18:22.08alrs(found my way to asterisk.org, still laughed)
18:22.23SamotI don't think it was ever digium.com
18:23.20alrsIt's been over a decade since I needed to care, I'm sure you're right.
18:23.34SamotOh, wait.
18:23.36SamotJFC.
18:24.22clarjon1www.digium.com redirects to sangoma.com, but it looks like the forwarder was skipped for the non-www subdomain
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19:06.34*** join/#asterisk sinaowolabi (~Sina@102.134.114.1)
19:06.52nbjoergSamot: well, given that there are devices that don't work with chan_pjsip...
19:07.28filesuch as/
19:07.29file?
19:07.55nbjoergfile: snom m(3)25 :(
19:08.25Samotnbjoerg: That is entirely false.
19:08.49Samotnbjoerg: PJSIP has been around for almost as long as Asterisk. It is a SIP driver used in software and hardware SIP applications/devices.
19:09.01fileexcept for the use of tel URIs, every time I've seen someone having problems with a device it's been configuration
19:09.08Samot^^^^^^
19:10.31*** join/#asterisk pchero (~pchero@211.178.226.108)
19:11.06SamotPJSIP 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.
19:11.36SamotSo this "providers don't support Chan_PJSIP" mantra is completely false.
19:12.39nbjoerggiven that there is a lot of commercial software around that only supports Internet Explorer: what has age to do with willingness to support something?
19:17.03Samotnbjoerg: You only took part of the point away from that.
19:17.24Samotnbjoerg: It's been around for over a decade and been used in software and hardware SIP applications/devices.
19:17.51SamotSo there's a high chance that many have connected to providers with PJSIP and not known it.
19:19.43nbjoerganyway, 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...
19:20.04SamotWell the problem is most likely a configuration issue.
19:21.26nbjoerg*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
19:21.42SamotIt's not.
19:21.47SamotIt's actual experience.
19:22.03SamotBut you want to lean on it being "chan_pjsip" is unsupported by providers. That's fine.
19:22.08SamotIt's wrong, but fine.
19:22.50igcewielingnbjoerg: don't worry, chan_sip is going away
19:22.55nbjoergigcewieling: I know
19:23.29nbjoergigcewieling: I wouldn't care if it worked...
19:23.41SamotWell let's hope the providers catch up by then.
19:23.49seanbrightdid you post some errors or logs somwhere?
19:23.57nbjoergseanbright: I did, it's been a while
19:23.58seanbrighti feel like i'm missing the front end of this conversation
19:24.01Samotnbjoerg: It works. If you're having actual problems please describe them and show some debugs.
19:25.15seanbrightif there are legitimate interoperability problems that are 100% on asterisk, obviously we would want to fix those
19:25.25nbjoergat 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...
19:25.37SamotI've gotten Polycom, Yealink, Yeastar, SNOM, Cisco SPA series, Matrix PBXes, Asterisk, FreePBX and others.
19:25.44SamotAll working on chan_pjsip.
19:25.46nbjoergI'm not even saying that they are bugs on the asterix/pjsip side
19:26.11nbjoergthe fun part with interop issues is that there are two sides to blame and that doesn't make things communicate either
19:26.29Samotnbjoerg: I've done interopping of PBXes and phone devices.
19:26.35Samotnbjoerg: I work at providers, it's part of what I do.
19:26.39SamotSo yes, this works just fine.
19:26.42seanbrightso... is there an actual problem?
19:26.47Samot^^
19:27.02seanbrightwhat are we talking about?
19:27.20nbjoergseanbright: 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
19:27.28seanbrightok
19:27.58nbjoergSamot: you don't have a support account for snom btw? I have an issue with the phonebook feature in newer firmware generations
19:28.10SamotNo, I don't.
19:28.16SamotI don't like snom's.
19:28.36nbjoerggrumbles about companies that don't want bug reports by only dealing with integrators
19:32.20nbjoerghttps://dpaste.org/pPO7 that was my chat log from when I looked at it
19:32.40SamotWell if you've ever used MicroSIP or csipsimple, you've used PJSIP.
19:33.37SamotOh wait, I remember this
19:33.41SamotThis is an issue with one phone.
19:34.28SamotIs that snom on the same network?
19:34.35nbjoergyes
19:34.49SamotI would need to see a real sip debug of this.
19:35.13SamotBecause based on that convo, you call from the snom it sends back a register to the auth challenge but no reply.
19:35.34SamotSo need to see what happens when it replies to the auth challenge.
19:35.46*** join/#asterisk mvanbaak (~mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak)
19:35.53SamotBut if it can receive calls, per this convo, I doubt it's a pjsip issue.
19:36.07*** join/#asterisk Janos (~textual@201.204.94.76)
19:36.22nbjoergwell, something(TM) was completely ignoring some of the packets
19:36.36SamotNeed to see an outbound call in action.
19:37.12SamotBecause it seems like it can register just fine and accept calls just fine.
19:37.19SamotSo it's something with not being able to send calls.
19:37.35nbjoergas I said, need time to set up a test bed so that I don't have to kill the main pbx :)
19:37.49nbjoergand spare time was rare the last few month
19:37.57nbjoerg"works for now"
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20:06.59*** join/#asterisk ShaunR (~Shaun@freenode/sponsor/NDChost.com)
20:42.20ShaunRHmm... 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))'
20:42.20ShaunRexit.  Popping routine return locations.'
20:43.25ShaunRI 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
20:43.30igcewielingShaunR: what dial option are you using to trigger the gosub?
20:43.56ShaunRigcewieling: exten = _X.,n,Dial(PJSIP/${EXTEN},30,U(screen-callee,s,1))
20:45.11igcewielingI 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."
20:45.18ShaunRwondering if i should use read instead.
20:50.43ShaunRgonna give that a try i guess.
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21:25.33ShaunRWell so far so good. Have to stop for a bit so to be continued;
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21:47.55igcewielingLOL!  "The 'S' in IOT stands for Security."
21:58.52ketashahaha
22:00.04ketasin addition to s, it needs d, and second i
22:06.55SamotHuh.
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