| 00:20.35 | *** join/#asterisk infobot (ibot@c-174-52-60-165.hsd1.ut.comcast.net) |
| 00:20.35 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.27.0 (2019/5/30) 16.4.0 (2019/5/30), Security Only: 15.7.2 (2019/2/28); DAHDI: 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 10:48.27 | *** join/#asterisk hemmond (~tom@mail.artisys.cz) |
| 10:52.24 | hemmond | Hi. I am currently trying to set up Asterisk server in LAN in a way, that it accepts any REGISTER requests without password challenge (and without the need to configure account to every URI). I thought that "allowguest" will do the trick - in default id should be "yes", but it doesnt. I'd like to ask you for any pointers where to look next. |
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| 11:08.44 | *** join/#asterisk emsjessec (~emsjessec@96.56.225.51) |
| 11:09.58 | hemmond | (repost from 12:52) Hi. I am currently trying to set up Asterisk server in LAN in a way, that it accepts any REGISTER requests without password challenge (and without the need to configure account to every URI). I thought that "allowguest" will do the trick - in default id should be "yes", but it doesnt. I'd like to ask you for any pointers where to look next. |
| 11:18.34 | *** join/#asterisk hemmond (~tom@mail.artisys.cz) |
| 11:40.20 | hemmond | Hi. I am currently trying to set up Asterisk server in LAN in a way, that it accepts any REGISTER requests without password challenge (and without the need to configure account to every registered user). I thought that "allowguest" will do the trick - in default it should be "yes", but it doesnt. I'd like to ask you for any pointers where to look next. |
| 11:40.47 | hemmond | Has anyone here any idea where to look next? |
| 11:53.43 | *** join/#asterisk FuriousGeorge (ad3f24d4@pool-173-63-36-212.nwrknj.fios.verizon.net) |
| 11:54.08 | FuriousGeorge | hey all |
| 11:54.56 | FuriousGeorge | im trying to come up with a way to test if a line from a channel bank to an analog phone is damaged. i have askterisk installed in a building where vermin chew on the cables |
| 11:55.33 | FuriousGeorge | when they break the cable, it still show up as "on hook" in the channel bank's web interface. i guess they'd have to be courteous enough to short it for me to know in that way |
| 11:56.31 | FuriousGeorge | im assuming that trying to call the phone would result in it just ringing, given the hook status, but i forgot to verify that |
| 11:57.09 | FuriousGeorge | which, if im not mistaken solves my problem, because the phones autoanswer |
| 11:57.18 | FuriousGeorge | so if it rings it must be busted |
| 11:57.27 | Reinhilde | i guess? |
| 11:57.37 | FuriousGeorge | i guess too |
| 12:00.05 | Reinhilde | rings more than 20sec |
| 12:01.28 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 12:21.25 | FuriousGeorge | Reinhilde: then send email subject: fix me |
| 12:21.57 | Reinhilde | FuriousGeorge: reinstall the cables with anti-mouse devices |
| 12:22.21 | FuriousGeorge | Reinhilde: they actually exist |
| 12:22.44 | *** join/#asterisk twanny796 (~user@antazzo.com) |
| 12:23.21 | FuriousGeorge | i've seen some with a chew resistant sheathings, and sheathings with repellents, and then another sheathing on top of that, so it doesn't repel you |
| 12:23.41 | Reinhilde | just has to be chew resistant, really |
| 12:24.03 | FuriousGeorge | i think those need to be added on to the cable |
| 12:24.13 | Reinhilde | something like that yeah |
| 12:24.29 | FuriousGeorge | but there are probably some that come preinstalled. i doubt the repellents work |
| 12:25.13 | FuriousGeorge | the best idea, i think, might be to unleash Jerry the cat into the risers, and sit back while hilarity ensues |
| 12:26.05 | FuriousGeorge | these are two different kinds "i've seen some with a chew resistant sheathings, and sheathings with repellents, and then another sheathing on top of that, so it doesn't repel you " |
| 12:26.20 | FuriousGeorge | i wasn't clear |
| 12:26.45 | Reinhilde | i'd install rat traps too, yeah |
| 12:26.59 | Reinhilde | and elevate the phone cables off the grund |
| 12:27.01 | Reinhilde | ground* |
| 12:27.10 | FuriousGeorge | it's happening in the walls |
| 12:27.19 | FuriousGeorge | there is no exposed cable |
| 12:27.49 | Samot | Clearly, they are exposed. |
| 12:28.00 | Reinhilde | They're exposed to something. |
| 12:28.01 | Samot | Regardless of being behind a wall where people can't see them. |
| 12:29.02 | FuriousGeorge | it's a high rise, and every column of units (101, 201... 1801, for example) has a riser, with the cables in it. they connect to junction boxes, where a callbox is mounted ooutside the wall |
| 12:29.19 | FuriousGeorge | the vermin are getting in the j-boxes |
| 12:29.40 | Reinhilde | FuriousGeorge: replace the J boxes, or you don't have a phone system |
| 12:30.24 | FuriousGeorge | they are extra large, custom for a prior system from a bygone company from a bygone era, and we retrofitted them call boxes onto them. |
| 12:31.08 | FuriousGeorge | alas, I did not get that scope of work, nor did anyone. this is a construction contract that the contractor is trying to wrap up. and so is this sub |
| 12:31.31 | FuriousGeorge | but maybe they will give it to me now |
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| 12:33.06 | FuriousGeorge | what im wondering about is... what do they want with the cable? do they use the jacket for their nest? |
| 12:33.31 | Samot | Does it matter? |
| 12:33.57 | FuriousGeorge | Samot: of course it does. there are people who dedicate their lives to studying the behavior of animals |
| 12:34.08 | Samot | Are you one of them? |
| 12:34.14 | Reinhilde | FuriousGeorge: but not mattering to YOU, RIGHT NOW. |
| 12:34.25 | Samot | Does that solve your issue? |
| 12:34.26 | FuriousGeorge | no, but it matters to me because i'm curious, otherwise i wouldn't have asked |
| 12:34.30 | Reinhilde | Samot: seemingly, but this is not his current scope |
| 12:34.32 | FuriousGeorge | i don't have an issue |
| 12:34.34 | FuriousGeorge | the owner does |
| 12:34.50 | Samot | And you need to solve it? |
| 12:35.01 | Samot | Thats your part, yes? |
| 12:35.01 | Reinhilde | FuriousGeorge: tell the owner to replace the J boxes with something rat-tight |
| 12:35.09 | FuriousGeorge | i'd love to, but no one is talking about a contract for that. i've suggested it. |
| 12:35.16 | FuriousGeorge | i just fix them as they come |
| 12:35.34 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 12:35.35 | FuriousGeorge | im going to start using a different cable though |
| 12:35.42 | Reinhilde | you don't have a phone system until you fix the J boxes and, yes, the cables |
| 12:36.16 | FuriousGeorge | they don't have one. mine still works |
| 12:37.28 | Samot | FuriousGeorge: Rats chewed for one simple reason. I learned this back in elementary school. |
| 12:37.49 | Samot | Just like other animals, Beavers, etc. Their teeth never stop growing. |
| 12:38.03 | Samot | They chew to help keep their teeth trimmed. |
| 12:38.42 | *** join/#asterisk FuriousGeorge99 (ad3f24d4@pool-173-63-36-212.nwrknj.fios.verizon.net) |
| 12:41.00 | FuriousGeorge | Samot: I had heard that before. Thank you for clarifying. |
| 12:41.40 | Samot | Now does that help you with your original question? No. |
| 12:41.46 | seanbright | i think we've all learned something today |
| 12:41.56 | seanbright | /join #animalfacts |
| 12:42.36 | FuriousGeorge | Samot: Reinhilde: ofc it's more frustrating for me than for you guys. i'd rather it just work than have to prove why it isn't. |
| 12:42.49 | FuriousGeorge | but what would you have me do? go godfather part 1 on them? https://memegenerator.net/img/instances/46275048/either-your-brains-or-your-signature-are-going-to-be-on-this-contract.jpg |
| 12:43.41 | FuriousGeorge | im going to write a script that detects line trouble, and sends them an email, so we can skip to the end every time it happens |
| 12:44.32 | Samot | How will it detect line trouble? |
| 12:49.50 | FuriousGeorge | Samot: fortuitously, the call boxes auto answer by default. so if call status is anything but "answered" then "something is terribly wrong" |
| 12:50.36 | Samot | But just because the line is answered doesn't mean there isn't trouble. |
| 12:50.52 | Samot | There could be static or other issues due to your rat friends chewing. |
| 12:51.07 | FuriousGeorge | Samot: let's call it a continuity test then |
| 12:51.37 | Samot | So you're going to write a script that calls each line? |
| 12:51.44 | FuriousGeorge | once a week or so |
| 12:55.20 | FuriousGeorge | i don't think there's much i can do beyond that. is there an electrical engineer in the house? couldn't i hook up an amphenol cable to a breakout board, and test the leads with a multimeter |
| 12:55.52 | FuriousGeorge | periodically, not that i expect they will request this service, but it's more useful to me than knowing why mice chew cable |
| 12:56.18 | FuriousGeorge | if resistance is high, i bet the line has static, for instance |
| 12:56.24 | FuriousGeorge | maybe i'm betting wrong |
| 12:56.46 | *** join/#asterisk scgm11_ (~scgm11@r186-50-47-122.dialup.adsl.anteldata.net.uy) |
| 12:58.50 | FuriousGeorge | aside from resistance... if i can test it inline, i bet the latent voltage can tell me something. when analog phones are on hook, i know one circuit is opened, but another is not, otherwise it would not be connected to any thing when it is time to ring |
| 13:03.14 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:03.57 | FuriousGeorge | i can put a butt an analog phone on it, and use a switch like on a butt set to alternatively test inline, answer, or take it off line to do a resistance test.... |
| 13:05.06 | FuriousGeorge | you could, i guess, calculate resistance knowing the output of the port, but that assumes the port is not malfunctioning... *shrug* |
| 13:09.52 | Samot | Do you know the mV you should be at? |
| 13:11.10 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 13:14.15 | hemmond | Hi. I am currently trying to set up Asterisk server in LAN in a way, that it accepts any SIP REGISTER requests without password challenge (and without the need to configure account to every registered user). I thought that "allowguest" will do the trick - in default it should be "yes", but it doesnt. I'd like to ask you for any pointers where to look next. |
| 13:20.45 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:22.08 | DanFromUK | Hi. Is it possible to disable three way calling across the whole of Asterisk? I've got a client whose staff keep accidentally conferencing customers together causing problems and embarrassment. |
| 13:30.48 | FuriousGeorge | Samot: i have not even started looking into the technicalities, though i'm intrigued enough to |
| 13:32.19 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 13:32.56 | file | DanFromUK: unless you are referring to chan_dahdi, then Asterisk doesn't do three way calling |
| 13:33.33 | Samot | FuriousGeorge: Well you were talking about putting a meter on it. So how would you know what reading is right? |
| 13:33.48 | DanFromUK | @file: the external callers are chan_dahdi, but the staff use chan_sip and that's what's triggering the three way calling. |
| 13:34.02 | file | chan_sip isn't doing the three way calling, the endpoint is |
| 13:34.10 | Samot | ^^^ |
| 13:35.28 | DanFromUK | is it? I thought the endpoint was sending a SIP request to asterisk to bridge all the calls. I didnt think the endpoint was actually handling the bridge itself. |
| 13:35.42 | Samot | For local conferences, yes. |
| 13:35.56 | Samot | Unless you are some how managing to put people in a confbridge. |
| 13:35.56 | DanFromUK | If the sip endpoint hangs up, the two chan_dahdi parties are left talking to each other, with no sip endpoint around. |
| 13:36.13 | Samot | DanFromUK: That is also phone dependent. |
| 13:36.34 | Samot | Whether or not to join the other to parties when the user leaves the call. |
| 13:37.10 | DanFromUK | ok. i'll tell the client it isn't possible to disable threeway calling. |
| 13:37.18 | Samot | That's untrue. |
| 13:37.41 | Samot | Until you look at the actual phone and it's capabilities that statement is not true. |
| 13:37.55 | Samot | Because many phones off the ability to turn of N-way calling |
| 13:38.15 | DanFromUK | Already emailed the sip hardware distributor to see if they know whether it can be disabled on the device. |
| 13:38.22 | Samot | What phones? |
| 13:38.26 | Samot | What models of phones? |
| 13:38.34 | DanFromUK | Gigaset SL750H with N510IP base |
| 13:38.41 | Samot | Oh, no clue. |
| 13:38.49 | Samot | What did the admin manual say? |
| 13:39.21 | DanFromUK | No mention of disabling it and, from the way the three way calling instructions read, it doesnt appear to be possible to disable. |
| 13:41.53 | Samot | And how are they ending up conferencing calls into a 3-way? |
| 13:42.54 | twanny796 | Can I have a dialing tone audible after I press 9 to do external calls? |
| 13:43.19 | twanny796 | audible dialing tone in the handset. |
| 13:44.41 | Samot | For SIP? Not unless the phone does it. |
| 13:44.52 | Samot | Otherwise 9 is just a prefix. |
| 13:45.41 | DanFromUK | Samot: when the staff answer call-waiting, the first call gets put on hold in order to answer the 2nd call. As the SL750H is now handling two calls, the soft-button changes to "Conference", when usually it's "Hold", "Resume". They press it without realising it's changed function, and that creates the 3-way call. |
| 13:46.31 | Samot | OK and the phone doesn't allow you to program or disable those buttons? |
| 13:47.17 | DanFromUK | I couldnt find the setting anywhere and never seen it before. I've emailed the distributor to see if they know. |
| 13:48.25 | Samot | Again, the admin manual as nothing on this? |
| 13:52.50 | *** join/#asterisk jkroon (~jkroon@165.16.203.105) |
| 14:06.42 | DanFromUK | Nope. |
| 14:06.53 | DanFromUK | No mention of options to disable it |
| 14:18.21 | hemmond | Hi, I need to use Asterisk as SIP Location service (random SIP URIs in LAN) only and don't want to put all possible SIP Uri's in configuration. Is there any way how to tell Asterisk to accept every SIP REGISTER packet for any number? |
| 14:20.36 | hemmond | Right now I am trying to use sip.conf but anything I do has the same result: Register, 401 (auth required), Register (with auth), 403 (unauthorized), "Wrong password" in the log. |
| 14:23.44 | *** join/#asterisk pchero_work (~pchero@87.213.247.82) |
| 14:24.05 | Samot | hemmond: Why would you want to accept anything and allow your box to be open like that? |
| 14:28.49 | *** join/#asterisk lbazan (~lbazan@fedora/LoKoMurdoK) |
| 14:30.47 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-qbhibnwfjcwvpyhp) |
| 14:30.47 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:31.37 | hemmond | Samot: It is box in LAN, separated from the outside world and registrations go from specific software phone and numbers are distributed by different program and dynamic. |
| 14:31.51 | hemmond | *software phones |
| 14:32.54 | [TK]D-Fender | And whatès the trouble in actually configuring the phones correctly to have a proper passwordà |
| 14:33.00 | [TK]D-Fender | ? |
| 14:33.39 | Reinhilde | [TK]D-Fender: are you quebecois? |
| 14:33.57 | Samot | hemmond: You can't register unknown endpoint/peers. |
| 14:34.03 | [TK]D-Fender | Yes. Had my keyboard in English until recently and the defaul keeps flipping on me... |
| 14:34.07 | Reinhilde | ha |
| 14:34.08 | Samot | What would they be registering for? |
| 14:35.14 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-kmsqikdabusxconp) |
| 14:35.15 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:36.32 | *** join/#asterisk lbazan (~lbazan@fedora/LoKoMurdoK) |
| 14:39.28 | hemmond | Samot: I need to use Asterisk as SIP Location service (to map SIP URIs to IPs for INVITEs... RTP streams go directly after that). |
| 14:41.12 | Samot | hemmond: Then you need to setup all the user accounts. That's how Registrar/Location services work. |
| 14:41.52 | hemmond | [TK]D-Fender: That the network is changing dynamically by configuration in different program and softphones reconfigures themselves. Usualy when part of the system is reconfigured to "simulate" different environment. |
| 14:42.39 | *** join/#asterisk cusco (~tralala@neptune.tretas.eu) |
| 14:42.57 | cusco | hello, anyone using mongodb stuff from https://github.com/minoruta/ast_mongo ? |
| 14:43.39 | Samot | hemmond: What I said doesn't change. As well you have to remember something RTP is always proxying through Asterisk. |
| 14:43.50 | Samot | It will take over the media when it needs to. |
| 14:44.01 | hemmond | Samot: Hmmm... Configure all possible accounts, that means 8 milion possible URIs, atleast. |
| 14:44.11 | Samot | Why? |
| 14:44.15 | Samot | How many users? |
| 14:44.24 | Samot | Why do you need to change the user accounts? |
| 14:44.37 | Samot | They don't have to be in direct relation to the numbers. |
| 14:46.28 | hemmond | Well... There are about 40 stations, but in the configuration they can be assigned pretty much any SIP URI and every station can have 1-16 SIP URIs. And depending on the configuration, each can have any number. And the software uses SIP URI also as login credentials. |
| 14:46.59 | Samot | Well if you have 8 million possible user accounts and have no way to manage it, that's your first problem. |
| 14:49.47 | Samot | hemmond: Asterisk is the wrong solution for this as a Registrar/Location service. |
| 14:50.01 | Samot | It can be part of the solution but it can't be the whole solution. |
| 14:50.25 | Samot | You need something like a SIP Proxy/SBC that can do the Registrar/Location services and then use Asterisk for any media. |
| 14:50.43 | hemmond | Main problem is, that users will have "simulated environment" which is used for training purposes and uses same technologies as real environment. Problem is, different sites have different numbers and someone who will be doing training preparation will put there numbers used on real site (so trainees can use the same numbers as they would use on site). I have no idea which numbers will be used in wich training scenarios, so I take into account all possible |
| 14:51.21 | Samot | When you say numbers you mean DIDs? |
| 14:52.23 | hemmond | As far as I kow, yes. |
| 14:52.43 | Samot | Then you won't have 8 million. |
| 14:52.45 | Samot | At all. |
| 14:53.20 | Samot | You would have as many DIDs as you own. |
| 14:55.20 | *** join/#asterisk lbazan (~lbazan@fedora/LoKoMurdoK) |
| 14:56.46 | hemmond | Oh... My mistake, no DIDs there. The Asterisk is not connected to any other SIP, PTSN or any other network. It should work only inside separated LAN - between stations whch can simulate different endpoints - even DIDs. Asterisk should be Location service and only endpoints should be the stations which will register to it. |
| 14:57.04 | Samot | hemmond: Your 8 million is just unrealistic. |
| 14:57.15 | Samot | hemmond: You should have control over what is created and why it's created. |
| 14:57.58 | Samot | hemmond: How big is this company? What is the average employee count? |
| 15:00.30 | Samot | hemmond: And you must have some method of storing or checking the accounts. How will anyone know what is already in use or even what shouldn't be in use? |
| 15:01.58 | *** part/#asterisk FuriousGeorge (ad3f24d4@pool-173-63-36-212.nwrknj.fios.verizon.net) |
| 15:02.21 | *** join/#asterisk FuriousGeorge (ad3f24d4@pool-173-63-36-212.nwrknj.fios.verizon.net) |
| 15:02.22 | hemmond | Samot: In use, there will be about 40 stations (about 3 registered numbers each), but problem is, for first training scenario they will use "simulated DIDs" from site 1, in second from site 5... And there can be any kind of phone numbers inserted in the training scenario. Distribution of who should register as which number is part of the trainig scenario and is distributed to stations along with other scenario data. |
| 15:02.40 | Samot | OK. |
| 15:02.46 | Samot | Again, it's not 8 million. |
| 15:03.59 | Samot | So you're saying they can create "456823523" as a test DID to receive incoming calls? |
| 15:04.07 | Samot | Or that they can make calls to? |
| 15:06.44 | hemmond | For example: Station A can register with number "456823523", Station B registers with number "456823008" |
| 15:06.44 | hemmond | Station A then wants to call "456823008", sends INVITE for sip uri "456823008" to Asterisk, Asterisk sends this INVITE to station B and call is made... |
| 15:09.32 | hemmond | Someone can even act as Emergency services (fire guard, police, ambulance, etc...) and trainee can use their station to dial that emergency number as part of training scenario. This number then shall be routed to station simulating emergency services. |
| 15:09.36 | Samot | hemmond: OK so what about the dialplan part of this? |
| 15:10.06 | Samot | How will A call B and Asterisk process it? |
| 15:10.27 | Samot | Because really A is call Asterisk, Asterisk is processing the call which results in Asterisk calling B |
| 15:10.34 | Samot | And then bridging the two calls together. |
| 15:12.41 | hemmond | Samot: From what I knew about SIP (and seen in wireshark when we used small Grandstream PBX for real system), PBX is bridging only SIP requests, which is how stations A and B know which IP has their counterpart and after placing the call, the call can be routed directly or via PBX. |
| 15:14.38 | hemmond | We were able to configure the PBX to communicate as on this diagram: https://www.tutorialspoint.com/session_initiation_protocol/images/sip_call_flow.jpg |
| 15:16.15 | hemmond | If you know about any other SIP PBX/Proxy which is maintained and able to run on Gentoo, I would be grateful. We used to use Opensips, but I am unable to install it for some reason. |
| 15:22.39 | *** join/#asterisk scgm11_ (~scgm11@r186-50-47-122.dialup.adsl.anteldata.net.uy) |
| 15:25.15 | Samot | hemmond: Asterisk is a Back-to-Back User Agent. |
| 15:25.33 | Samot | hemmond: So when device 100 registers to Asterisk, it's sending it's requests to Asterisk. |
| 15:25.58 | Samot | hemmond: Asterisk needs to know what to do with the digits presented to it. So you need dialplan to handle that. |
| 15:26.24 | Samot | hemmond: So if 100 wants to call 200, there needs to be dialplan that MATCHES for 200 to then DIAL 200. |
| 15:27.04 | Samot | hemmond: Asterisk is not a Proxy. |
| 15:28.35 | Samot | 10:49:48 AM <Samot> hemmond: Asterisk is the wrong solution for this as a Registrar/Location service. |
| 15:28.35 | Samot | 10:50:02 AM <Samot> It can be part of the solution but it can't be the whole solution. |
| 15:28.35 | Samot | 10:50:27 AM <Samot> You need something like a SIP Proxy/SBC that can do the Registrar/Location services and then use Asterisk for any media. |
| 15:31.54 | hemmond | Samot: Thank you for your knowledge. Do you know which SIP Proxy could be used for this use-case and preferably is simple to configure? And if it will work on Gentoo Linux, that would be a big plus. :) Someone here used to use opensips sip proxy here, but for some reason we are not able to install it at all... |
| 15:32.09 | hemmond | That's why we tried to use Asterisk. |
| 15:38.17 | Samot | KAmailio or OpenSIPs would be my suggestion. |
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| 16:06.51 | hemmond | Samot: Thank you, I will give Opensips a try. :) |
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| 18:15.42 | FuriousGeorge | i was logged off but now i'm not |
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| 21:00.00 | IpSo | Anyone know why I can't get Asterisk queues to make the holdtime announcement sooner than 90 seconds into the call? I have announce-frequency = 30, min-announce-frequency = 29 set, but when testing it always makes the first announcement about 1min 30secs into the call, never earlier than that. |
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| 21:04.22 | IpSo | n/m, just discovered the bit in the documentation about the queue retry period, I was testing when one handset was available and just being ignored. |
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