| 00:22.14 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:22.14 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); 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 |
| 01:06.56 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 01:26.57 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 01:27.58 | tripleslash | I have/had a stuttering noise on only outbound calls. Only the interior party could hear it. It would sound like the person was talking through a fan. Very short, interrupted pauses that if the person used short words, you wouldn't notice, but holding a word or listening to music would make it painfully obvious. After testing all sorts of configurations, even installing a new, very simply |
| 01:27.59 | tripleslash | box with no change, I tried switching from ast13 to ast14. That made no change. Switching to ast11 however made my issue go away. It appears there's something going on with the latest versions published via yum but I have no way to diagnose further. If there's something someone wants me to run to get more info, I'm happy to oblige for the cause. |
| 01:29.48 | Samot | What are the resources of the VM again? |
| 01:30.46 | tripleslash | I installed on bare metal for my testing before I changed the ast version on the vm. |
| 01:31.03 | Samot | What were the resources? |
| 01:31.05 | tripleslash | sec |
| 01:33.46 | tripleslash | i5-2400 @3.1g, 8g ram, 250gb hd |
| 01:34.20 | Samot | I had that issue once. |
| 01:34.28 | Samot | Where I couldn't upgrade to 13... |
| 01:34.36 | Samot | Because of that and other issues... |
| 01:34.54 | Samot | VM was under powered |
| 01:35.23 | Samot | i5 is what single or dual? |
| 01:35.30 | tripleslash | single I think |
| 01:35.38 | tripleslash | this isn't a new machien |
| 01:35.44 | tripleslash | probably 3 or 4 years old |
| 01:36.22 | Samot | Yeah.. |
| 01:36.37 | WIMPy | 2400 is 4 cores 3.1GHz |
| 01:36.58 | tripleslash | there we go :) |
| 01:37.33 | Samot | Well I experienced that problem once and kicking the VM up a bit fixed it. |
| 01:37.39 | WIMPy | 3.4GHz Turbo boost, no HT. |
| 01:39.09 | tripleslash | the vm was sitting on a cluster, I moved all the other vms off of it when it was first having issues, so at one point, it had 24 logical processors available. |
| 01:39.47 | tripleslash | yum history showed the last update on the 9th which was after the issue originally showed up. |
| 01:40.11 | WIMPy | It happened suddenly? |
| 01:40.16 | tripleslash | yep |
| 01:40.25 | WIMPy | Nice |
| 01:40.27 | tripleslash | users started complaining the morning of the second. |
| 01:40.51 | WIMPy | I'd look at other network components then. |
| 01:41.36 | tripleslash | well, I went and made a brand new network via dumb switch and it still happened |
| 01:41.51 | WIMPy | And at differences in the nwtwork output between the versions. |
| 01:42.11 | tripleslash | adtran, pbx, phone. GigE all around. |
| 01:42.56 | WIMPy | What's that adtran doing? |
| 01:43.08 | tripleslash | that's my sip trunk from the provider. |
| 01:43.20 | tripleslash | it passes everything to pbx via local ip |
| 01:43.56 | WIMPy | Err, what exactely does it do if it's IP? |
| 01:44.53 | tripleslash | its passing sip from my isp. Comcast doesn't do sip over the traditional data network. They bring in a different modem, put the adtran and let the customer talk to the adtran. |
| 01:45.15 | WIMPy | That sounds ... interesting. |
| 01:45.44 | WIMPy | Maybe that thing was upgraded? |
| 01:45.47 | tripleslash | it keeps things secure, assuming the adtran doesn't get owned, because adtran will only talk to a single ip on my local network. |
| 01:46.12 | tripleslash | I asked them, they said no, but I trust them about as much as I trust a 3 year old that's been left alone in a candy store for 10 minutes. |
| 01:47.12 | tripleslash | They only do 711 and 722. |
| 01:47.37 | WIMPy | Sounds ok to me. |
| 01:48.24 | tripleslash | up until this issue, things were good. |
| 01:48.54 | WIMPy | Well, that's one of the things about using SIP. You never know for how long it will work. |
| 02:10.32 | *** join/#asterisk CryptoManiac (~Crypto@unaffiliated/cryptomaniac) |
| 02:26.00 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 02:27.37 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 02:35.46 | *** join/#asterisk adnauseum (uid126405@gateway/web/irccloud.com/x-kpemosxkdtdmvbob) |
| 02:37.08 | avb | WIMPy: my friend have the same thing in the office |
| 02:37.48 | avb | not sure about a hardware brand, but its been as well shipped with some crazy UPS which is alive and kicking for last 6 years :) |
| 02:39.08 | Samot | Yeah, it's the one thing about PRIs. You never know how long it will work. |
| 02:39.41 | WIMPy | I've never had a PRI fail. |
| 02:39.58 | Samot | You've never EVER had a PRI circuit have issues? |
| 02:40.10 | Samot | EVER? |
| 02:40.19 | WIMPy | No |
| 02:40.46 | WIMPy | And I only once had a BRI fail. |
| 02:41.17 | WIMPy | The telephony shit just works unless someone cuts the cable. |
| 02:41.45 | Samot | Yeah, no. |
| 02:41.49 | Samot | It can be poorly wired. |
| 02:41.59 | Samot | It can have poor connections, splitters.. |
| 02:42.06 | Samot | It can be programmed wrong. |
| 02:42.15 | WIMPy | Or some idiot changes the wrong cable. |
| 02:42.19 | Samot | There are many, many factors that can make a PRI/T1 not work. |
| 02:42.34 | WIMPy | Sure. But since it's all automated, it just doesn't happen./ |
| 02:43.05 | Samot | Wire pairs jump connectors automatically? |
| 02:43.11 | Samot | That's impressive. |
| 02:43.34 | WIMPy | That's teh "Or some idiot changes the wrong cable." thing. |
| 02:43.49 | Samot | You just said it was automated |
| 02:44.03 | Samot | Wire pairs are often moved around for space and routing.. |
| 02:44.34 | WIMPy | That was in reply to "It can be programmed wrong.". |
| 02:44.58 | Samot | Well perhaps I just don't understand Europe. |
| 02:45.16 | Samot | But at some point someone has to wire that last mile to point Z |
| 02:45.34 | WIMPy | Not sure that has anything to do with Europe. |
| 02:46.25 | Samot | 10:42:36 PM W<WIMPy> Sure. But since it's all automated, it just doesn't happen. <-- What's all automated then? |
| 02:47.04 | WIMPy | All the configuration. |
| 02:47.38 | Samot | You mean the logical routing configuration.. |
| 02:47.55 | WIMPy | There's quite a lot you can configure (wrong). |
| 02:48.01 | Samot | Yes. |
| 02:48.04 | Samot | There is. |
| 02:48.23 | Samot | That's why I said there are many, many factors as to why a PRI/T1 circuit can fail. |
| 02:48.39 | Samot | Or have issues. |
| 02:48.46 | WIMPy | Yes, and I said there aren't because it's automated. |
| 02:49.02 | Samot | It's not ALL automated |
| 02:49.17 | WIMPy | It is. |
| 02:49.26 | Samot | How does the circuit get wired? |
| 02:49.27 | WIMPy | Except for wiring, but we already had that. |
| 02:49.55 | Samot | OK. So you've never had a single PRI circuit have issues, At all. Ever. |
| 02:49.56 | WIMPy | Repeating old arguments that already have been agreed is not getting any further. |
| 02:50.07 | WIMPy | Nope. |
| 02:50.38 | Samot | Well then you have the luck of the Irish shining down on you. |
| 02:50.53 | WIMPy | Not even a falsely removed wire. |
| 02:51.05 | WIMPy | Not really. |
| 02:51.20 | Samot | How many circuits? |
| 02:51.31 | WIMPy | Telephony used to be extremely reliable technology. |
| 02:52.17 | WIMPy | Not that many. BRIs are much more common. Let me count... |
| 02:52.35 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 02:52.41 | WIMPy | I thin it have been 9. |
| 02:52.45 | Samot | Oh. |
| 02:52.51 | WIMPy | +k |
| 02:52.51 | Samot | K. |
| 02:52.59 | Samot | 9,000? |
| 02:53.12 | WIMPy | No 9. |
| 02:53.17 | Samot | Oh. |
| 02:53.18 | Samot | OK. |
| 02:53.29 | WIMPy | What are you goi9ng to do with that many PRIs? |
| 02:53.40 | Samot | I worked at CLEC's. |
| 02:53.58 | Samot | We had circuits that never had issues. |
| 02:54.15 | Samot | We had circuits that had occasional issues |
| 02:54.27 | Samot | And we've had circuits that were chronic issues... |
| 02:54.49 | drmessano | All it takes is one tech moving a pair around in a cabinet or pulling the wrong card at the CO |
| 02:55.03 | drmessano | It's not hard to create an "Act of Human" failure |
| 02:55.10 | Samot | So yeah, if you've had 9 BRI/PRI circuits and didnt have issues with them... |
| 02:55.11 | drmessano | Even on a circuit that's been fine for years |
| 02:55.13 | Samot | I can see that. |
| 02:55.19 | WIMPy | Yes, that can and does happen. |
| 02:55.29 | WIMPy | 9 PRIs. Can't count the BRIs. |
| 02:55.35 | Samot | Well... |
| 02:55.44 | Samot | We don't have BRI's really in the US |
| 02:55.55 | Samot | People use T1 or PRI circuits for that. |
| 02:56.17 | WIMPy | Yes, but PRIs don't make sense to the residential user. |
| 02:56.18 | Samot | But a circuit is a circuit. |
| 02:56.26 | drmessano | BRI's are even worse for reliability here |
| 02:56.32 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 02:56.49 | Samot | No, of course not. |
| 02:56.58 | Samot | We didn't have residential users for that kind of stuff. |
| 02:57.06 | Samot | That's what POTS/DS0 lines were for. |
| 02:57.36 | drmessano | We did use BRI's a lot in Radio.. and to some extent, still do.. But good luck keeping one up and running |
| 02:57.40 | WIMPy | Most of our businesses are on 2-4 BRIs. |
| 02:58.02 | Samot | Sounds about right. |
| 02:58.15 | Samot | Most of ours had 2-6 circuits. |
| 02:58.17 | Samot | Depending. |
| 02:58.25 | Samot | We also did Private Line... |
| 02:58.32 | Samot | Between premises... |
| 02:59.18 | Samot | When I was there we were averaging about 50 circuit turn ups per day |
| 02:59.48 | Samot | Not counting SIP turn ups over another Internet connection. |
| 03:01.04 | Samot | Or our DS3(T3) turn ups. |
| 03:07.22 | Samot | I can discuss the pros and cons of SIP and Analog/Digital copper (wired) Telephony all day. But to sit that and say "SIP BAD!" and no treat it like Telephony (or a public utility for that matter) it does nothing more than validate basement ITSP that SIP is something easy and not really "Telephony". |
| 03:07.39 | Samot | no = not |
| 03:08.36 | WIMPy | It is ancient shit. |
| 03:09.09 | Samot | What is ancient? |
| 03:09.19 | WIMPy | SIP |
| 03:09.41 | Samot | Yes, it's been around for a while. |
| 03:09.55 | WIMPy | there have been far petter VoIP protocolls after SIP. |
| 03:10.22 | Samot | Such as? |
| 03:10.29 | WIMPy | Any. |
| 03:10.36 | Samot | IAX? |
| 03:10.43 | WIMPy | Yes. |
| 03:10.56 | Samot | H.323? |
| 03:11.33 | WIMPy | IAX is actually a really good one. If had received some TLC it could be quite excellent by now. |
| 03:11.38 | WIMPy | yes. |
| 03:12.33 | WIMPy | But even as it is, IAX is pretty good. |
| 03:12.59 | Samot | Just never adopted outside of Asterisk really. |
| 03:13.31 | WIMPy | Or things that came from Asterisk. |
| 03:16.46 | Samot | SIP was more versatile. |
| 03:17.21 | WIMPy | Yes, and not tailored towards telephony at all. |
| 03:18.12 | Samot | How do you come to that conclusion? |
| 03:18.50 | WIMPy | Because all the bit you'd want were later additions or still don't exist. |
| 03:18.57 | WIMPy | bits |
| 03:19.21 | Samot | Like? |
| 03:19.48 | WIMPy | Hundresds of additions with usually 3 different and incompatible ones for the same feature. |
| 03:20.22 | Samot | That's still not a clear example. |
| 03:20.26 | WIMPy | Was there much more than from: and to: in the original? |
| 03:20.30 | Samot | What doesn't still exist? |
| 03:20.37 | Samot | RURI |
| 03:20.43 | Samot | TURI FURI |
| 03:20.49 | Samot | DURI |
| 03:20.51 | Samot | Yeah... |
| 03:20.52 | WIMPy | Just read all the add-ons. That's lots of examples. |
| 03:21.09 | WIMPy | A remote operations protocol. |
| 03:21.11 | Samot | What doesn't still exist? |
| 03:21.26 | WIMPy | No way to control call forwarding or conferences. |
| 03:21.32 | Samot | Huh? |
| 03:21.57 | WIMPy | Where do you find that in SIP? |
| 03:22.01 | Samot | Control call forwarding in what way? |
| 03:22.23 | WIMPy | Via menu. |
| 03:22.33 | Samot | You mean * codes? |
| 03:22.44 | Samot | vertical codes? |
| 03:22.45 | WIMPy | No. |
| 03:22.54 | WIMPy | A standard protocol feature. |
| 03:23.13 | Samot | Call forwarding? |
| 03:23.57 | WIMPy | But conferences or the lack thereof are even worse. |
| 03:24.12 | Samot | What do you mean lack of conferences? |
| 03:24.31 | WIMPy | SIP doesn't have any way to control conferences, either. |
| 03:24.37 | Samot | What do you mean? |
| 03:24.44 | Samot | Control them how? |
| 03:25.31 | WIMPy | Like you do in the PSTN or PBXes. |
| 03:25.46 | Samot | What do you mean in the PSTN? |
| 03:25.58 | Samot | Conferencing is not a "Class 4" feature |
| 03:26.09 | WIMPy | What ever that means. |
| 03:26.45 | Samot | It's a level of routing between exchanges... |
| 03:26.53 | Samot | The PSTN is routing. |
| 03:27.09 | WIMPy | Nothing I care about as a user. |
| 03:27.23 | Samot | Services like conferences and such that phone companies offered were enhanced services handled by specific systems. |
| 03:27.45 | Samot | OK |
| 03:27.53 | Samot | Well as someone who does this for "users" |
| 03:28.02 | Samot | And dealt with this at the PSTN level down. |
| 03:28.09 | Samot | You are saying things that are not true. |
| 03:28.11 | WIMPy | At least thesmall conference (3PTY) is a standard feature here. |
| 03:28.26 | WIMPy | What isn't true? |
| 03:28.26 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 03:28.35 | Samot | Back in the day that was called "Citrix/Centrix" services here. |
| 03:28.47 | Samot | That SIP cannot control call forwarding. |
| 03:28.50 | Samot | Or conferences. |
| 03:29.04 | WIMPy | That's something else. And has hardly ever been implemented here. |
| 03:29.16 | WIMPy | So how can you do it with SIP? |
| 03:29.22 | Samot | It's routing. |
| 03:29.54 | Samot | You do it the same way as if you routed a regular old number on a POTS or PRI |
| 03:29.59 | WIMPy | What's the RFC for controlling CF via SIP? |
| 03:30.06 | Samot | There isn't one |
| 03:30.14 | Samot | Because controlling CF isn't a PROTOCOL thing |
| 03:30.23 | WIMPy | What SIP message are used to set up conferences? |
| 03:30.38 | Samot | Again, conferences are not a PROTOCOL thing. |
| 03:30.45 | Samot | You send the call the conference server. |
| 03:30.55 | WIMPy | ThaSo yu finally agree that SIP can't do it? |
| 03:31.01 | Samot | It's not that SIP can't do it |
| 03:31.18 | Samot | Because PRIs/POTS/copper can't do it. |
| 03:31.27 | Samot | Those are specific systems. |
| 03:31.30 | WIMPy | It surely could do it if someone came up with yet anoter addition to it. |
| 03:31.32 | Samot | That calls are ROUTED TO |
| 03:31.42 | WIMPy | No. |
| 03:31.49 | Samot | OK. |
| 03:32.02 | Samot | As someone who has done this... |
| 03:32.24 | Samot | Keep telling me how things are done aren't done that way. |
| 03:32.29 | drmessano | This is a weird argument |
| 03:32.30 | WIMPy | Those are standard features that can be conrolled with any Phone via standardized protocol features. |
| 03:32.52 | Samot | Because the phone company ALLOWED it |
| 03:32.56 | drmessano | I don't need these features built into the protocol. All the switches do it |
| 03:33.04 | WIMPy | SIP is like going back to POTS. |
| 03:33.07 | Samot | They said "Hey when you dial *67 you'll jump into a conference room" |
| 03:33.24 | Samot | SIP is a deliver method. |
| 03:33.26 | Samot | Like POTs |
| 03:33.28 | Samot | Yeah |
| 03:33.41 | Samot | Routing doesn't really change at the higher level if it's POTS or SIP |
| 03:33.53 | Samot | You still routing the call path.. |
| 03:34.14 | WIMPy | You really need to take a look at telephony that's less than 30 years old. |
| 03:34.26 | Samot | LOL |
| 03:34.32 | Samot | OK. |
| 03:34.37 | voipmonk | thanks - coffee all over my screen |
| 03:34.44 | drmessano | Like arguing that SIP doesn't support distinctive ring |
| 03:34.54 | drmessano | When uhh.. I don't need it in the protocol? |
| 03:34.56 | Samot | Or call forwarding control. |
| 03:35.14 | WIMPy | Distinctive ring does exist in SIP. |
| 03:35.17 | Samot | Or basic routing that applies to all telephony. |
| 03:35.25 | WIMPy | Unline Call forwarding. |
| 03:35.44 | drmessano | WIMPy: Literally never heard of it being used, ever |
| 03:35.45 | Samot | I call forward all the time with sip. |
| 03:35.51 | drmessano | WIMPy: Which is pretty telling |
| 03:36.05 | drmessano | Because the switch handles how the call is presented |
| 03:36.11 | drmessano | So why do I need it |
| 03:36.11 | Samot | ^^^^ |
| 03:36.25 | Samot | Call comes into DID, routing is looked up |
| 03:36.32 | Samot | If the user has CF enabled.. |
| 03:36.35 | Samot | That route exists.. |
| 03:36.40 | Samot | If it exists, use it. |
| 03:36.48 | Samot | Nothing related to SIP |
| 03:36.51 | Samot | H.323 |
| 03:36.53 | Samot | IAX |
| 03:36.57 | WIMPy | What is it you're trying to tell me about call routing? And how is it related to the topic? |
| 03:36.58 | Samot | POTS |
| 03:37.00 | drmessano | This is basically some desire to have all these features on the wire |
| 03:37.08 | drmessano | Which doesn't make sense |
| 03:37.16 | WIMPy | And how does the USER contol it? |
| 03:37.22 | voipmonk | 181 Call is Being Forwarded - there's even a provisional response for it |
| 03:37.26 | drmessano | Their equipment does |
| 03:37.32 | Samot | Because as the phone company YOU SET IT UP |
| 03:37.35 | drmessano | Thats the point |
| 03:37.53 | Samot | If my users dial *123 it sends them to the voicemail server. |
| 03:37.57 | WIMPy | Exactely. |
| 03:38.02 | Samot | But I DID THAT |
| 03:38.09 | Samot | It has no bearing on SIP |
| 03:38.22 | Samot | It has a bearing on the DIGITS the USER sent. |
| 03:38.26 | WIMPy | I don't want the phone company to set it up. I'm used to do it on the phone. |
| 03:38.37 | Samot | Then you have aPBX |
| 03:38.47 | drmessano | ^^^^ which does the work |
| 03:38.50 | Samot | That's the point of those |
| 03:39.04 | WIMPy | You don't need a PBX for that. The PSTN would offer it as well. |
| 03:39.11 | Samot | GD NO |
| 03:39.23 | Samot | They only offer it if they have a system that DOES IT |
| 03:39.55 | Samot | The PSTN in a network of routing and interconnections... |
| 03:40.10 | Samot | The LEC is responsible for the services it OFFERS |
| 03:40.16 | WIMPy | It you think that's all it is, wake up. |
| 03:40.33 | Samot | Yes. The PSTN supports voicemail boxes. |
| 03:40.41 | Samot | All of it. |
| 03:41.05 | Samot | And conference rooms. |
| 03:41.16 | Samot | That's just standard parts of the PSTN anyone can use. |
| 03:41.47 | WIMPy | At least you know those. |
| 03:42.18 | Samot | Funny how the only way the LEC could offer voicemail services was to create a voicemail system... |
| 03:42.42 | Samot | Because for some reason, no one told them where on the PSTN to make that happen. |
| 03:42.46 | drmessano | Some people are never going to be happy until they have SMS on their POTS line, so they can never use it.. But it's fun to complain about not having |
| 03:43.44 | Samot | Because SOMEHOW the PSTN knows which of the LEC users have voicemail and conference features without getting inside the LECs network to figure that out. |
| 03:43.45 | WIMPy | SMS was pretty handy for alarms. |
| 03:44.28 | drmessano | SMS is expired technology for alerting |
| 03:45.07 | WIMPy | It's pretty dead for sure. |
| 03:45.28 | drmessano | I pity people relying on SMS for alarms |
| 03:46.10 | igcewieling | um, in Europe, they have SMS on POTS. Not as popular these days as messaging moves from the SMS protocol. |
| 03:46.25 | Samot | They have it in US/Canada. |
| 03:46.38 | Samot | No body likes it because its a charge. |
| 03:46.41 | drmessano | igcewieling: I'm well aware... |
| 03:46.47 | Samot | That you didnt ask for. |
| 03:47.08 | Samot | You text a landline and if the LEC supports it, it will TTS the message. |
| 03:47.24 | igcewieling | THAT isn't SMS on POTS. |
| 03:47.30 | drmessano | No |
| 03:47.33 | WIMPy | If they (or the user) DON't support it. |
| 03:48.01 | Samot | Oh yeah, I can reply back to a text on my POTS line. |
| 03:48.11 | Samot | Yeah, I can't see why that didnt take off. |
| 03:48.48 | igcewieling | I am referring to SMS aka  ETSI ES 201 912 protocol |
| 03:49.04 | WIMPy | Funnily it was extremely popular amongst mobile users. |
| 03:49.06 | igcewieling | Where you have a nice text terminal on your phone |
| 03:49.57 | WIMPy | Not sure I'd call writing text on the keypad "nice". |
| 03:51.36 | drmessano | Solution without a problem |
| 03:52.07 | WIMPy | That must bee the various versions of video phones. |
| 03:53.27 | igcewieling | *shrug* I don't care if you agree or not. |
| 03:54.41 | lorsungcu | is offended |
| 03:55.25 | drmessano | POTS also doesn't support SHA2 hashed passwords |
| 03:55.31 | drmessano | Which means basically it's insecure |
| 03:55.35 | WIMPy | I'm not even sure if I own a phone that supports SMS. |
| 03:55.41 | Samot | Lol. |
| 03:56.01 | drmessano | Im serious |
| 03:56.07 | drmessano | My Dad called me the other day |
| 03:56.14 | drmessano | He asked me for one of his passwords |
| 03:56.15 | igcewieling | ha! you can still buy them on amazon. |
| 03:56.27 | drmessano | I gave it to him, and it was CLEAR TEXT over the wire |
| 03:56.28 | igcewieling | https://www.amazon.co.uk/BT-Relate-Corded-Telephone-Messaging/dp/B00027YETY |
| 03:56.42 | drmessano | How the fuck doesn't the PSTN hash passwords? |
| 03:57.18 | lorsungcu | the problem is that sms is stored in cleartext |
| 03:57.43 | drmessano | SMS with RADIUS is even worse |
| 03:57.48 | drmessano | Even using OpenVPN |
| 03:57.59 | Samot | Currently unavailable. |
| 03:57.59 | Samot | We don't know when or if this item will be back in stock. |
| 03:58.14 | Samot | ^^ You can *find* them on amazon, can't buy them. |
| 03:58.15 | drmessano | IRC actually masks passwords |
| 03:58.29 | drmessano | This is my GMAIL password |
| 03:58.32 | drmessano | ********** |
| 03:58.35 | drmessano | See that ^ |
| 03:58.44 | drmessano | Try it, it works |
| 04:02.03 | Samot | Nice try. |
| 04:02.24 | drmessano | Not sure I would call that "nice" |
| 04:02.29 | drmessano | It was pretty weak sauce |
| 04:02.45 | Samot | Yeah. |
| 04:02.47 | drmessano | Somewhere between "weak" and "pathetic" |
| 04:02.51 | drmessano | But not quite either |
| 04:03.59 | drmessano | Samot: Have you ever used Uptime Robot? |
| 04:06.53 | Samot | No. |
| 04:07.47 | drmessano | Im trying to monitor one of my Asterisk boxes using their "Generic Port" test |
| 04:07.59 | drmessano | I can monitor my TLS port, all day long |
| 04:08.20 | drmessano | But 5060 doesn't work even with TCP configured |
| 04:08.35 | Samot | Hrm.. |
| 04:09.54 | Samot | Give me a couple minutes |
| 04:11.49 | drmessano | well, I need to check the box.. nmap is showing no response from 5060/TCP |
| 04:12.17 | drmessano | I wasn't thinking about nmap at 3am last night |
| 04:12.48 | Samot | Where's the list of IPs they check from? |
| 04:14.09 | Samot | Got it |
| 04:14.19 | drmessano | Ok I figured it out.. was pretty simple |
| 04:14.35 | drmessano | No idea why, but TCP was disabled |
| 04:15.07 | Samot | Yeah, mine might be a firewall thing. |
| 04:15.48 | Samot | Monitis has a UDP SIP monitor... |
| 04:15.55 | drmessano | I have been using Monit for years and decided to try uptime robot |
| 04:16.07 | drmessano | I just read about Monitis yesterday |
| 04:16.18 | Samot | Yeah, I've used them on other projects. |
| 04:16.20 | Samot | I like them. |
| 04:16.31 | Samot | They didn't have the SIP monitor last year. |
| 04:16.47 | Samot | You just run a small shell script so it will send stats back. |
| 04:17.02 | drmessano | Monitis is very not free lol |
| 04:17.08 | Samot | No. |
| 04:17.27 | Samot | But depending on the monitors you use, it's not overly that bad. |
| 04:17.49 | drmessano | Uptime Robot is free until I have enough monitors that I have figured out how to monitize it |
| 04:17.56 | Samot | Right. |
| 04:18.05 | drmessano | The one thing I do like about Uptime Robot |
| 04:18.08 | Samot | But I was doing more that just checking uptime. |
| 04:18.12 | drmessano | That was a little harder with Monit |
| 04:18.38 | Samot | It was watching the health, (hdd, cpu, etc), MySQL, Apache, SMTP... |
| 04:18.43 | Samot | bandwidth usage.. |
| 04:19.03 | drmessano | Is that I can go to my customers and get a yearly monitoring fee out of them, and just add their SMS/Pushover/Email to the monitor for their site |
| 04:19.04 | drmessano | and boom |
| 04:19.48 | Samot | Plus Monitis has widgets for the monitors.. |
| 04:19.59 | Samot | So you can add them to your own dashboard interface. |
| 04:21.02 | drmessano | Yeah |
| 04:23.33 | Samot | I mean for quick and basic uptime monitoring, this looks pretty good. |
| 04:24.44 | drmessano | Yep |
| 04:25.10 | drmessano | I basically use monit on my servers |
| 04:25.21 | drmessano | and they're fine for the local monitoring |
| 04:25.43 | drmessano | I was also using monit for remote monitoring |
| 04:26.04 | drmessano | Which is ok, but the web UI sucks and the notifications are all cobbled together |
| 04:26.27 | drmessano | I have this crazy shell script for parsing and manipulating the Monit variables so I can send them to Pushover |
| 04:26.54 | drmessano | Uptime Robot is basically free at this point, has a native IOS app, and the Pushover/SMS/Email support is all built-in |
| 04:27.02 | drmessano | So less to support |
| 04:28.11 | Samot | Yeah. I'll have to add all the IPs in to my sysytems. |
| 04:28.16 | Samot | Yeah. I'll have to add all the IPs in to my systems. |
| 04:28.42 | drmessano | I'm not even sure what I ever did without Pushover |
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| 05:26.16 | lorsungcu | Samot: consider hosting a text file somewhere permanent and fetch'ing it in MT to update a running address lsit |
| 05:26.45 | Samot | Oh I mean in the cloud systems. |
| 05:26.54 | Samot | I'll have to update my iptables, etc. |
| 05:27.30 | lorsungcu | ah right. |
| 05:28.58 | Samot | I thought about just writing a script that has the MT "check-in"... |
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| 06:11.52 | drmessano | Trips me out that I can't get my UPS monitor working |
| 06:12.17 | Samot | What's the issue? |
| 06:13.29 | drmessano | I'm using nut |
| 06:13.40 | drmessano | Had it on my old server at home |
| 06:13.47 | drmessano | Have it on several boxes at work |
| 06:13.59 | drmessano | "New" server at home, no notifications |
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| 08:53.01 | wasanzy | hi |
| 08:53.42 | wasanzy | I have combined my cert and key into .pem. pastebin contains how I did it and configs. https://paste.linux.community/view/6c5b2a80 |
| 08:54.16 | wasanzy | I get the error when I tried to register sipML5: WebSocket connection to 'wss://192.168.10.39:7443/ws' failed: WebSocket opening handshake was canceled |
| 08:54.16 | wasanzy | tsk_utils.js?svn=251:116 __tsip_transport_ws_onerror |
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| 09:25.38 | wasanzy | hi any one in? |
| 09:27.43 | dokma | There's a bunch of folks. Have patience... |
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| 09:58.16 | Ast001 | Hi is it possible to record both channels in call that is not Bridged in this scenario. I call person find out with AMD it is automatic machine and playback message to it. On recording that I have I can hear only my playback but not machine from other side. I use MixMonitor without b option which should record not bridged channels right ? |
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| 10:35.30 | GeneralSpongebob | Hi, what soft phone do you use to run multiple instances on one machine when testing queues and such? I can't just connect one instance to multiple accounts because it gives busy |
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| 10:38.15 | wasanzy | hmmm am getting there. I hv the two certs working and same cert on both asterisk and apache. |
| 10:38.41 | wasanzy | this error came in now, don't know why though. chan_sip.c:10712 process_sdp: Rejecting secure audio stream without encryption details: audio 51001 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 |
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| 10:46.21 | dan_j | Hi. Is anyone aware of an asterisk compatible addon that allows for taking card payments by getting the caller to enter their card number using dtmf and getting them transmitted securely to the card provider? |
| 10:46.32 | dan_j | I'm trying to get around the issue of call recording with card numbers. |
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| 10:56.31 | Haris | guys ... in this ----> 2005 (Local/2005@from-queue/n from hint:2005@ext-local) (ringinuse enabled) (Not in use) has taken 9 calls (last was 1599 secs ago) <---- what does this portion mean ----> 2005 (Local/2005@from-queue/n from hint:2005@ext-local) ? |
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| 11:02.39 | Haris | what does (1) Local (2) 2005 (3) from-queue (4) /n (5) hint:2005 (6) ext-local mean in the above ? |
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| 11:13.00 | wasanzy | looking at this, I have all that is said in place so I am lost why the failure again |
| 11:13.04 | wasanzy | http://forums.asterisk.org/viewtopic.php?f=1&t=90167&sid=66fdf8cc4be5d955ba584e989a23442f |
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| 12:06.26 | wasanzy | hello |
| 12:25.57 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:30.21 | wasanzy | [TK]D-Fender: hi |
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| 13:04.44 | Haris | guys does ----> == Everyone is busy/congested at this time (1:0/0/1) <---- this mean that all agents are busy or all channels are busy or what ? |
| 13:05.39 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:07.03 | [TK]D-Fender | Everyone being dialed fron that channel |
| 13:07.47 | [TK]D-Fender | Haris, https://wiki.asterisk.org/wiki/display/AST/Local+Channel |
| 13:07.54 | [TK]D-Fender | Haris, https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints |
| 13:08.01 | [TK]D-Fender | for your previous question |
| 13:13.10 | wasanzy | my audio still doesn't work, tried all I could |
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| 13:26.09 | Haris | [2017-03-15 18:05:22] VERBOSE[7586][C-000002ee] chan_sip.c: -- Got SIP response 480 "Temporarily unavailable" back from x.x.x.x:5060 |
| 13:26.10 | Haris | [2017-03-15 18:05:22] VERBOSE[1068][C-000002ee] app_dial.c: -- SIP/TeleCard-000005d5 is circuit-busy |
| 13:26.10 | Haris | [2017-03-15 18:05:22] VERBOSE[1068][C-000002ee] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0) |
| 13:26.39 | Haris | If I have purchased a pri (30 channels line) from a telco/provider, should I be receiving this message with only 3-5 agents live ? |
| 13:26.46 | Haris | making simultaneous calls |
| 13:27.16 | Haris | from a sip/telco/provider |
| 13:27.36 | Haris | we don't have a 30 channel config at the provider end .. right ? |
| 13:28.19 | WIMPy | How is that message from cha_SIP related to taht PRI? |
| 13:28.41 | Haris | I'm watching this ----> https://www.youtube.com/playlist?list=PLnzEbgyK52Gu9fdVDHburrsG3KBIntXFK <---- these days |
| 13:29.07 | Haris | this msg is coming from the sip/telco's SIP server ? |
| 13:29.13 | Haris | their IP is referenced in it ? |
| 13:29.41 | WIMPy | You didn't tell us what you tried to do there. |
| 13:30.00 | Haris | if I have 3 agents, on a 30 channel line, and getting circuit busy response from remote sip/telco server, that means .. ? |
| 13:30.07 | Haris | make out going calls ? |
| 13:30.41 | [TK]D-Fender | <Haris> [2017-03-15 18:05:22] VERBOSE[7586][C-000002ee] chan_sip.c: -- Got SIP response 480 "Temporarily unavailable" back from x.x.x.x:5060 <- why are you masking your PROVIDER IP? |
| 13:30.44 | Haris | ofcourse, we are making outgoing calls. gettin this msg when we do that |
| 13:30.46 | WIMPy | Not on that PRI. |
| 13:31.12 | Haris | [TK]D-Fender: What's the point of putting it in public ? |
| 13:31.25 | [TK]D-Fender | What's the point in hiding it? |
| 13:31.28 | Haris | WIMPy: Please elaborate |
| 13:31.40 | [TK]D-Fender | Makes me think you are loking at things BACKWARDS again |
| 13:31.46 | Haris | well, I hear too much about security concerns. lol :P |
| 13:31.47 | [TK]D-Fender | Show the actual CALL. |
| 13:31.53 | WIMPy | Your log said it happened on a sip channel. |
| 13:32.03 | [TK]D-Fender | We can never tell jsut how much you've screwed up when you show us butchered scraps like that |
| 13:32.14 | Haris | hmm |
| 13:32.33 | [TK]D-Fender | <Haris> well, I hear too much about security concerns. lol :P <-= paranoia from idiots who ask why their car doesn't work and refuses to let the mechanic look under the hood |
| 13:32.53 | Haris | that was a joke |
| 13:32.53 | [TK]D-Fender | These are the kinds of people that deserver a high-five |
| 13:32.56 | [TK]D-Fender | to the face |
| 13:33.01 | [TK]D-Fender | with a closed fist |
| 13:33.07 | Haris | WIMPy: That didn't make sense |
| 13:33.51 | WIMPy | True story. So what is the actual question. |
| 13:33.53 | Haris | here's an instance ---> |
| 13:33.54 | wasanzy | what else can I do to get my audio making? |
| 13:34.14 | Haris | WIMPy: I mean that went over me. Didn't get it |
| 13:34.47 | Haris | this is a screen-full of log from the /var/log/asterisk/full log file ------> http://pastebin.ca/3780611 |
| 13:34.54 | Haris | showing this log msg |
| 13:35.22 | wasanzy | Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd. |
| 13:35.31 | wasanzy | this error kept coming |
| 13:35.41 | [TK]D-Fender | Haris, You aren't looking at SIP debug. |
| 13:35.53 | [TK]D-Fender | when your calls don't work, ALWAYS be loking at SIP debug |
| 13:35.58 | WIMPy | So that "TeleCard" at 10.20.80.2 rejected your call. |
| 13:36.10 | wasanzy | from sipML5 |
| 13:36.58 | [TK]D-Fender | And use live CLI, not "logs" |
| 13:37.41 | Haris | ? |
| 13:37.58 | [TK]D-Fender | What part of that are you having trouble with? |
| 13:38.07 | Haris | WIMPy: That was what I was saying |
| 13:38.22 | Haris | what I see in live cli is what is going in full log file |
| 13:38.50 | Haris | checking if my live cli instance has an instance of this event |
| 13:39.08 | WIMPy | No you asked about a PRI. And I don't see anything like that. |
| 13:39.29 | Haris | I mean its not a literal pri |
| 13:39.46 | Haris | its a fiber line, on which we asked the sip/telco provider to give us a 30 channel line |
| 13:39.57 | Haris | I'm using the term pri for using a short word for that |
| 13:40.23 | Haris | in essence I'm summing up the 30 channel thing in a single word |
| 13:40.34 | WIMPy | So your ITSP is refusing a call. Somethign else to debug. |
| 13:40.42 | Haris | yep! they are |
| 13:40.42 | [TK]D-Fender | <Haris> what I see in live cli is what is going in full log file <- SIP DEBUG. YOU ARE NOT LOOKING AT THE SIP DEBUG THERE. YOU ARE RUNNING BLIND. |
| 13:40.49 | Haris | so they aren't giving me 30 channels on that line .. correct ? |
| 13:40.53 | WIMPy | Ans as [TK]D-Fender said, you should look at the actual sip debug in that case. |
| 13:41.00 | Haris | ah |
| 13:41.58 | Haris | *ouch*. I forgot the debug output. its painfully long |
| 13:42.16 | Haris | didn't have that enabled today. I can get that tomorrow |
| 13:43.00 | WIMPy | Alternatively calls your ITSP and ask them what that message means. |
| 13:43.14 | Haris | already emailed them with attachment(s) from log |
| 13:43.37 | [TK]D-Fender | Place a call |
| 13:43.44 | [TK]D-Fender | PASTEBIN |
| 13:43.53 | Haris | in this case, if the remote sip server was asterisk, can I assume that they didn't config a 30 channel for me on their end ? |
| 13:43.54 | [TK]D-Fender | "didn't have" = past. |
| 13:43.57 | [TK]D-Fender | past = logs |
| 13:44.00 | [TK]D-Fender | logs = SHIT |
| 13:44.04 | [TK]D-Fender | CLI = NOW |
| 13:44.18 | [TK]D-Fender | Assumptions are for CHUMPS |
| 13:44.44 | [TK]D-Fender | If yuo aren't looking at a real call right in front of your face then all you are doing is guessing which is a waste of everyone's time |
| 13:47.20 | Haris | got it |
| 13:50.40 | TandyUK | to assume is to make an ass of u and me |
| 13:52.54 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 13:53.13 | [TK]D-Fender | ~assumptions |
| 13:53.13 | infobot | hmm... assumptions is now unless otherwise stated, most support here is based on the assumptions that: 1. You're using the box you're trying to fix now, not some other box (or OS), 2. You're using Debian, 3. You have physical access to the system (it's not hours' drive, or flight, away), 4. You understand English, 5. You will check commands given against online documentation (man, info, /usr/share/doc, HOWTOS, Google) (more in assumptions2) |
| 13:53.16 | Haris | ofcourse |
| 13:53.16 | [TK]D-Fender | ~assumption |
| 13:53.17 | infobot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 13:53.22 | [TK]D-Fender | ^ |
| 13:53.53 | Haris | I'll discuss this tomorrow, since that is when I can get sip debug output. right now, I don't have the means to reproduce this event on my own |
| 13:53.58 | Haris | thank you all |
| 13:55.21 | *** join/#asterisk phrearch (~spindle@34.tbb.spindle.osso.nl) |
| 13:55.24 | phrearch | hello |
| 13:56.00 | phrearch | im trying to figure out how to build asterisk with opus support. which libraries do i need? menuselect greys out opus codeec translation |
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| 14:35.54 | wasanzy | I have solved my problems. fun fun fun. |
| 14:38.49 | [TK]D-Fender | phrearch, As I've seen OPUS as an actual transcodable codec is only distributed as a binary so should ahve nothing to do with compiling on your side |
| 14:38.53 | [TK]D-Fender | http://blogs.digium.com/2016/09/30/opus-in-asterisk/ |
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| 14:54.31 | phrearch | <[TK]D-Fender> cool, tnx |
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| 14:56.54 | phrearch | i setup tls on http (wss) like https://paste.kde.org/p4xfatxnj, but only port 8088 seems to be exposed |
| 14:57.40 | phrearch | i already generated keys with ./ast_tls_cert -C 123.123.123.123 -O "foobar" -d /etc/asterisk/keys/ |
| 15:00.02 | phrearch | is there some module missing that may prevent using tls in the http server? |
| 15:00.29 | sotoz | In order to dial two different numbers and bridge them together (with ARI through a StasisApp) in order to talk is the following order/thinking correct: 1) create the first channel 2) create the bridge and add the first channel 3) dial the first channel, 4) create the second channel and add it on the bridge 5) Dial the second channel |
| 15:02.29 | sotoz | any ARI expert?? ^^ |
| 15:05.58 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 15:07.58 | ghoti | What are some Web UI configuration options I could use to make Asterisk configurable for people who only spend time in GUIs? FreePBX seems pretty comprehensive... but are there other options too? |
| 15:08.03 | ghoti | Priorities would include .. setting up new extensions, time-based forwarding (i.e. after hours support schedule), DISA configuration, etc. |
| 15:08.34 | igcewieling | Unless you want to write your own GUI, FreePBX is about all there is out there. |
| 15:09.00 | igcewieling | Writing your own GUI is a lot harder than it sounds. |
| 15:11.01 | DivideBy0 | phrearch: it looks reasonable. did you restart asterisk? |
| 15:11.39 | DivideBy0 | sotoz: that's fine, or add 2nd channel on its StasisStart |
| 15:11.44 | DivideBy0 | (add to bridge) |
| 15:11.56 | zaf | yeah, i just got (almost) done writing my own gui |
| 15:13.17 | drmessano | zaf: It's probably missing "every feature I use every day" |
| 15:13.27 | zaf | probably |
| 15:13.58 | zaf | it's driven by our RMM |
| 15:14.04 | drmessano | That's sort of the challenge. You'll never implement everything in the way someone would want it implemented. Just too many variables. |
| 15:14.06 | sotoz | DivideBy0 The thing is that the second outgoing channel doesn't get dialed for some reason, and I also get 2 more channels on my Stasisapp . |
| 15:14.58 | zaf | drmessano, the goal in my case is to have it do things exactly the way i would want it done, no matter which one of my minions^H^H^H^H^H^Htechnicians does it |
| 15:15.02 | drmessano | The virtue of FreePBX is years of "I want this to be able to do this too". After about 10 years of that, some people might consider it non-anemic in feature. |
| 15:15.08 | sotoz | Should I do a Originate instead of create channel/dial? |
| 15:15.10 | drmessano | Lol |
| 15:15.30 | drmessano | Well if YOU are the only audience that's a good thing |
| 15:16.01 | drmessano | Writing a GUI for yourself is great. Writing a GUI for the masses is an early heart attack |
| 15:16.15 | drmessano | I wouldn't want that burden |
| 15:21.57 | *** join/#asterisk infobot (ibot@rikers.org) |
| 15:21.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); 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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| 15:30.14 | phrearch | hm, still no https |
| 15:30.24 | phrearch | http show status also doesn't show tls bind address |
| 15:31.05 | wasanzy | Samot: I was able to solve the audio issue now |
| 15:31.15 | Samot | How's that? |
| 15:32.40 | phrearch | maybe, i needed ./configure --with-ssl |
| 15:33.52 | phrearch | aha, ` The OPENSSL installation appears to be missing or broken.` maybe this is the issue... |
| 15:34.42 | wasanzy | Samot: the issue wasn't actually with the certs I put for sipML5 |
| 15:35.32 | wasanzy | thanks for the help. |
| 15:35.56 | Samot | What was the issue? |
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| 15:36.21 | *** join/#asterisk sotoz (~sotoz@095-097-255-066.static.chello.nl) |
| 15:36.41 | phrearch | ehm, i don't get tls support while i set tls properly in http.conf |
| 15:37.05 | phrearch | installing libssl-dev worked for that error about openssl |
| 15:38.19 | wasanzy | Samot: I run ./install_prereq install and ./install_prereq install-unpackaged then recompiled asterisk and it worked |
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| 15:38.51 | DivideBy0 | sotoz: you'd have to pastebin some code |
| 15:38.52 | Samot | So this goes back to my question of "You installed all the needed things on the server" |
| 15:39.00 | phrearch | is there a way to check if the tls part in the http server is not active? |
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| 15:40.41 | phrearch | i see that menuselect/the buildscript automagically takes care of download/installing the opus codec modules, but that's only for intel chipsets. Would that work on arm somehow manually? |
| 15:41.07 | phrearch | i mean, is there a manual how to install/compile those libs manually? |
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| 15:45.15 | phrearch | there seems to be https://github.com/seanbright/asterisk-opus. is the new one opensource as well? |
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| 15:53.22 | seanbright | the new one is not open source |
| 15:53.24 | seanbright | and i wouldn't use mine |
| 15:54.06 | seanbright | phrearch: if not digium's, use this one https://github.com/traud/asterisk-opus |
| 15:55.28 | phrearch | seanbright: ah, thanks for the tip! |
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| 15:59.29 | phrearch | ok great, http tls works now |
| 15:59.55 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 16:01.38 | EmleyMoor | I'm getting this warning: [Mar 15 16:01:21] WARNING[31799][C-00000010]: ast_expr2.y:1298 op_plus: non-numeric argument |
| 16:01.59 | EmleyMoor | It is being triggered by this dialplan line: same => n(tryagain),GotoIf($[${ATTEMPTS}<3]?attempt:noclid,${ARG1},1) |
| 16:02.15 | EmleyMoor | What do I need to do to stop the warning happening? |
| 16:03.27 | rmudgett | Possibly ${ATTEMPTS} evaluates to an empty string so the expression < 3 is invalid. |
| 16:04.05 | EmleyMoor | rmudgett: No... ${ATTEMPTS} is always either 1, 2 or 3 |
| 16:04.35 | EmleyMoor | (actually, it starts at 0 but that line is not reached until it is at least 1) |
| 16:07.17 | *** join/#asterisk bof23 (~Thunderbi@185.13.183.107) |
| 16:07.53 | igcewieling | EmleyMoor: put a Noop(ATTEMPTS='${ATTEMPTS}' ARG1='${ARG1}) right before that line to be SURE. |
| 16:08.12 | igcewieling | the single quotes are cosmetic, they don't mean anything. |
| 16:08.55 | EmleyMoor | igcewieling: The section is working perfectly, apart from generating the warning... but will try that and see anyway |
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| 16:10.21 | igcewieling | What is the ACTUAL line shown in the CLI which causes that? I don't mean the dialplan, I mean the CLI output, which contains the RESULT of the $[] |
| 16:10.43 | igcewieling | EmleyMoor: expression problems are hell to troubleshoot 8-( |
| 16:15.50 | EmleyMoor | igcewieling: The actual line shown in the CLI is: -- Executing [askclid@checkpermission:22] GotoIf("SIP/aain-00000014", "1?attempt:noclid,02070434811,1") in new stack on the first twice, and 0?... etc, on the third... |
| 16:16.20 | EmleyMoor | However, your NoOp has shown that it's not putting in the variable what I expect... |
| 16:17.06 | EmleyMoor | It starts at 0, but then is $0 + 1, then $$0 + 1 + 1, then $$$0 + 1 + 1 + 1, so I guess the problem is my adding line... |
| 16:17.24 | igcewieling | There is a function INC() which might do what you want |
| 16:18.39 | EmleyMoor | igcewieling: Ah, I see - let me try that... |
| 16:19.09 | igcewieling | here is a line from my dialplan same => n,Set(loop=${INC(loop)}) |
| 16:24.24 | EmleyMoor | igcewieling: Indeed, that works well. (It's my new "ask for caller's number if withheld/unavailable" routine) |
| 16:24.37 | igcewieling | glad it helped. |
| 16:25.44 | EmleyMoor | Have used a similar routine before to get round a PSTN interface fault, but need it on all technologies now |
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| 16:33.04 | igcewieling | Do you ask them to press some DTMF to prove they are human? |
| 16:34.12 | igcewieling | my log message for our similar feature: "Caller is either human or an advanced AI possibly bent on the destruction of humanity, accepting call." |
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| 16:34.53 | EmleyMoor | igcewieling: Just to enter a number to call back on - which is then subjected to checking, like presented caller IDs. |
| 16:35.08 | igcewieling | ah, different from what I was thinking of. |
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| 16:44.14 | lvlinux | hey, if i do a git clone of asterisk -b 14 will it actually get 14.3 ?? or 14.0? |
| 16:45.44 | lvlinux | in other words will i end up with the same thing as if I downloaded the current 14 ? |
| 16:47.23 | WIMPy | It will be newer than current. |
| 16:49.48 | file | 14 is the branch for 14, it includes changes since the last release |
| 16:49.54 | file | if you want a specific release then you can check out the specific tag |
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| 17:02.39 | EmleyMoor | just realised (and fixed) an issue with his caller ID asking routine |
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| 17:27.04 | lvlinux | k thx |
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| 18:59.38 | igcewieling | Gawd, I hate Verizon. I'm so tempted to put an intercept on down numbers like "We're sorry, Verizon's lines are down again. The following are the telephone numbers of your local public utilities comission." |
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| 19:31.24 | *** join/#asterisk Prelude2004c (~Prelude20@69.57.222.18) |
| 19:31.39 | Prelude2004c | hey guys.. does anyone know if codec2 and ulaw translation works.. even opus & ulaw translation |
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| 20:08.34 | igcewieling | Prelude2004c: "core show translations" |
| 20:08.45 | igcewieling | err...l Prelude2004c: "core show translation" |
| 20:09.08 | igcewieling | that shows you everything Asterisk can transcode. |
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| 20:16.00 | Prelude2004c | just didn't work.. when i call they cant' hear me.. i can hear them but they can't hear me |
| 20:16.01 | Prelude2004c | its odd |
| 20:18.46 | Samot | Inbound or outbound call? |
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| 20:39.06 | drmessano | Opus translation exists |
| 20:39.12 | drmessano | transcoding |
| 20:39.21 | drmessano | Codec2, who knows |
| 20:39.37 | drmessano | I know of 0 people that are using Codec2 with Asterisk |
| 20:39.45 | drmessano | Why would you? |
| 20:41.10 | drmessano | I think he has me on ignore.. i've asked him this like 3 times |
| 20:46.18 | Samot | Is MulticastRTP still a valid channel type? |
| 20:46.37 | Samot | Or did it change, I thought I read that somewhere and now I can't find that page... |
| 20:48.13 | Samot | Ahh, it was 14. |
| 20:48.20 | Samot | chan_rtp (was chan_multicast_rtp) |
| 20:52.39 | Samot | drmessano: You do multicast? |
| 20:57.38 | Samot | I can set ${CALLERID(num)} and ${CALLERID(num)} on a MulticastRTP channel, correct? |
| 20:57.49 | Samot | It should still take standard channel variables? |
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| 21:26.34 | drmessano | I do not |
| 21:29.13 | zaf | anyone using Digium D80s in production/test yet? Got my first one in and it doesn't seem to be configuring like I expect |
| 21:31.24 | Samot | drmessano; I'll just have to do some testing. |
| 21:31.32 | Samot | Might have to jump up to 14.. |
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| 21:37.40 | malcolmd | zaf, re: D80. ? |
| 21:37.53 | zaf | hi malcolmd |
| 21:38.04 | malcolmd | is this (https://community.asterisk.org/t/dpma-and-d80-need-info/70052) you? |
| 21:38.13 | zaf | malcolmd, that is me, yep |
| 21:38.19 | malcolmd | k, i'll take over there |
| 21:40.53 | zaf | here is fine too :P |
| 21:41.39 | malcolmd | sure, but there's got the benefit of archive for anyone else that runs into it. i know that some people log IRC, but google does a better job with forums indexing than IRC indexing. request for info there in the form of a response |
| 21:41.53 | zaf | right |
| 21:42.08 | zaf | just responded, it's got 1_1_2 currently |
| 21:44.20 | malcolmd | and, responded ;) |
| 21:45.20 | zaf | is the D80 supposed to listen to DHCP option 66 properly? |
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