| 00:04.48 | *** join/#asterisk lankanmon_ (~LKNnet@2607:fea8:d1f:ffcb:3da5:2251:b56:ca65) |
| 00:06.26 | *** join/#asterisk DelphiWorld (~VOIPER@openvpn/user/DelphiWorld) |
| 00:06.28 | DelphiWorld | hey everyone |
| 00:06.36 | DelphiWorld | anyone used tls with aastra phones? |
| 00:21.19 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:21.20 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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 |
| 01:39.34 | *** join/#asterisk elcontrastador (~textual@c-67-182-43-55.hsd1.ca.comcast.net) |
| 01:44.25 | *** join/#asterisk spicyramen (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 01:46.08 | elcontrastador | I'm researching using chan-sccp-b with Asterisk for an installation of ~800 ancient Cisco 79[46][01] handsets currently managed by Cisco UCM. Does anyone here have production experience large scale SCCP deployment with Asterisk? |
| 02:15.37 | *** join/#asterisk bhans (~bhans@unaffiliated/bhans) |
| 02:42.38 | *** join/#asterisk d00gster (~d00gster@unaffiliated/d00gster) |
| 02:52.56 | *** join/#asterisk lankanmon (~LKNnet@2607:fea8:d1f:ffcb:9594:ee6b:366b:f0e0) |
| 02:55.56 | spicyramen | @elcontrastador, for production I would not recommend using sccp and Asterisk, specially in those old phones which phone load itself is no longer in Development, anyways, you can take a look at https://github.com/chan-sccp/chan-sccp/graphs/contributors |
| 03:19.40 | *** join/#asterisk boris_t (~boris_t@94.190.2.146) |
| 03:31.54 | *** join/#asterisk spicyramen (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 05:05.16 | *** join/#asterisk phix (~threat@14-203-214-74.tpgi.com.au) |
| 05:23.03 | *** join/#asterisk phix (~threat@220.240.108.69) |
| 05:30.46 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:50.12 | *** join/#asterisk phix (~threat@202.53.52.209) |
| 05:58.24 | *** join/#asterisk spicyramen (~Adium@c-73-241-232-155.hsd1.ca.comcast.net) |
| 06:14.55 | *** join/#asterisk earlybird (~earlybird@unaffiliated/joj) |
| 06:32.03 | *** join/#asterisk chl_ (~chl@unaffiliated/chl/x-9330839) |
| 06:33.01 | chl_ | hello, is it possible to disable the psqlodbc log in asterisk? |
| 06:39.17 | *** join/#asterisk bhans (~bhans@unaffiliated/bhans) |
| 06:42.37 | *** join/#asterisk jkroon (~jkroon@165.16.204.34) |
| 06:42.57 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:14.55 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:16.40 | *** join/#asterisk Rasputin3711 (~Rasputin3@87.255.254.66) |
| 07:40.19 | *** join/#asterisk phix (~threat@59-102-106-151.tpgi.com.au) |
| 08:09.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:10.49 | *** join/#asterisk anpi (~crr@unaffiliated/anpi) |
| 08:15.07 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:32.13 | *** join/#asterisk Mr_Pleb_Mgoo (~jakeb@103.46.213.148) |
| 08:58.53 | *** join/#asterisk phix (~threat@148.146.220.203.dial.dynamic.acc01-aubu-gou.comindico.com.au) |
| 09:00.15 | *** join/#asterisk earlybird (~earlybird@unaffiliated/joj) |
| 09:05.18 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 09:13.10 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 09:29.16 | *** join/#asterisk phix (~threat@C-61-69-172-184.syd.connect.net.au) |
| 09:53.44 | *** join/#asterisk bl3nto (~bl3nto@cpe-109-60-6-222.st3.cable.xnet.hr) |
| 10:29.02 | *** join/#asterisk elegast (~elegast@5.79.86.129) |
| 10:36.25 | *** join/#asterisk yuka (~elegast@2001:980:3385:1:a11:96ff:fe19:64c4) |
| 10:37.39 | *** join/#asterisk nighty- (~nighty@s229123.ppp.asahi-net.or.jp) |
| 11:19.50 | *** join/#asterisk mou (~donwillia@188.228.46.39) |
| 11:31.33 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 11:35.39 | *** join/#asterisk phix (~threat@220.240.112.252) |
| 11:53.15 | *** join/#asterisk phix (~threat@61.68.139.56) |
| 12:00.30 | *** join/#asterisk mr_kyd (~Einherjer@unaffiliated/mr-kyd/x-4253709) |
| 12:20.37 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 12:51.14 | *** join/#asterisk CrummyGummy (~CrummyGum@41.161.9.27) |
| 13:00.34 | CrummyGummy | Hi, I'm setting up a support call queue. Damn voicemail is causing problems for me with a simple sequential dial plan. I have another idea though. I'd like to play a message to the caller, hangup, then dial all the members of the support list sequentially until one of them enters a code to prove they are human. I'd then like to then initiate a call from that user to the original caller to close the loop. |
| 13:00.40 | CrummyGummy | Is that doable? |
| 13:01.27 | Samot | Uhm. |
| 13:01.30 | Samot | It's a queue. |
| 13:01.54 | Samot | The point behind is that it will send calls to members based on how you define the rules. |
| 13:02.13 | CrummyGummy | My queue ends when the voicemail picks up though. |
| 13:02.36 | Samot | Where is the agent? |
| 13:02.39 | CrummyGummy | It's dialling out to peoples cell phones |
| 13:02.45 | Samot | OK. |
| 13:02.50 | CrummyGummy | A/H support |
| 13:02.57 | Samot | So send it to an extension |
| 13:02.59 | CrummyGummy | People don't wake up... |
| 13:03.12 | Samot | That uses FollowMe or something. |
| 13:03.27 | Samot | Force the agent to have to press 1 or something to accept the call |
| 13:04.05 | Samot | As a user, calling a support queue to playback a message and then hang up... |
| 13:04.15 | Samot | While I wait for an "agent" to call me back.... |
| 13:04.29 | Samot | Might as well just send me to voicemail and deal with it like that |
| 13:05.58 | CrummyGummy | What you said about FollowMe is what I was looking for. Maybe I need to get the clients to connect via sip client on their cells? |
| 13:06.06 | Samot | No. |
| 13:06.18 | Samot | FollowMe is for when they aren't available or don't answer.. |
| 13:06.29 | Samot | "Follow Me to X, Y or Z" |
| 13:07.02 | Samot | You can tell Asterisk when the call is sent, you want the callee to send a DTMF response to _accept_ the call. |
| 13:07.11 | CrummyGummy | YES |
| 13:07.14 | Samot | If there is no DTMF response, it's unanswered. |
| 13:07.17 | CrummyGummy | that's exactly what I want |
| 13:07.22 | CrummyGummy | 1st prize |
| 13:07.30 | Samot | So the agent will still get a voicemail |
| 13:07.39 | Samot | But it's going to be "Press 1 to connect" |
| 13:07.55 | Samot | It's going to be the confirmation recording not the caller. |
| 13:08.04 | Samot | The caller will be snapped back into the queue. |
| 13:08.55 | Samot | You can probably mimic this straight from the queue with the proper dialplan. |
| 13:10.58 | CrummyGummy | From a dialplan without using the queue app? |
| 13:13.11 | CrummyGummy | that would be cool |
| 13:14.38 | *** join/#asterisk pawiecki__ (~pawiecki@router.dir.pl) |
| 13:14.42 | [TK]D-Fender | You can still use Queue |
| 13:14.52 | [TK]D-Fender | Do you WANT a queue? |
| 13:14.56 | *** join/#asterisk dacod (~dacod@187.103.104.42) |
| 13:15.37 | CrummyGummy | not particularly |
| 13:16.01 | Samot | Then you at least need a Ring Group |
| 13:16.03 | [TK]D-Fender | So you don't want distribution amongst multiple destinations? |
| 13:16.33 | Samot | Sequential order really isn't screaming "need a queue" |
| 13:17.01 | [TK]D-Fender | Not if you're not intending to re-ring in a loop |
| 13:17.07 | [TK]D-Fender | or maintain a calling order |
| 13:17.22 | CrummyGummy | Sure, it'd be nice if it worked like a ring group. An AGI function that gets the next number to dial, dials it, if the user presses 1 then it would connect, otherwise dial next. |
| 13:17.39 | [TK]D-Fender | AGI doesn't offer anything special for this |
| 13:17.52 | [TK]D-Fender | No more than regular diallpan |
| 13:17.54 | Samot | CrummyGummy: Are all the other calls in queues? |
| 13:18.04 | Samot | Is there reporting related to this? |
| 13:18.29 | Samot | Like "How long do callers have to wait for a A/H tech? Did the A/H techs answer the calls in X time frames?" |
| 13:18.31 | CrummyGummy | I don't use this for anything else atm. AGI would just help because I can queue the support staff numbers in a DB. |
| 13:18.38 | [TK]D-Fender | Are you looking to dial multiple people simultaneously, or only sequentially? |
| 13:19.00 | [TK]D-Fender | You can still use DB's in dialplan |
| 13:19.03 | [TK]D-Fender | but that's a choice |
| 13:19.25 | CrummyGummy | Someone will want reporting eventually but CDRs will do for now. |
| 13:19.40 | CrummyGummy | Someone always wants reporting... |
| 13:20.02 | [TK]D-Fender | Again, simultaneous or sequential? |
| 13:20.07 | CrummyGummy | sequential |
| 13:20.48 | Samot | Again, FollowMe. |
| 13:20.52 | [TK]D-Fender | Then use AGI for the selection, and regular dialplan to Dial and check for confirmation |
| 13:20.52 | Samot | Mimic it. |
| 13:20.54 | Samot | Or use it. |
| 13:21.00 | Samot | Right. |
| 13:21.19 | [TK]D-Fender | There is no "Followme". We should be dialing a single target without a followup planned |
| 13:21.27 | [TK]D-Fender | dialplan/AGI should determine that |
| 13:21.34 | Samot | I said mimic it. |
| 13:21.35 | [TK]D-Fender | it's not like a desk user wanting to hunt them down |
| 13:21.39 | Samot | He wants human confirmation. |
| 13:21.53 | Samot | That's what I was referring to in the "mimic" |
| 13:22.14 | Samot | Call the list, wait for confirmation. No confirmation, go to next # in list |
| 13:22.56 | [TK]D-Fender | [TK]D-Fender> Then use AGI for the selection, and regular dialplan to Dial and check for confirmation <- that's the confirmation |
| 13:22.58 | [TK]D-Fender | So yup |
| 13:23.04 | [TK]D-Fender | "core show application dial" <- |
| 13:23.09 | Samot | Yup. |
| 13:23.34 | CrummyGummy | hmmm, thanks, I was over complicating this |
| 13:23.41 | Samot | Yes. |
| 13:23.44 | Samot | Bigly. |
| 13:24.41 | [TK]D-Fender | YUGE. |
| 13:24.44 | [TK]D-Fender | Believe me. |
| 13:25.12 | Samot | Really at this point, it comes down to how you want to handle the actual caller. |
| 13:25.24 | Samot | What are they going to hear while the agent is found.. |
| 13:25.34 | Samot | How will you handle them dropping out of the call... |
| 13:25.51 | Samot | Will there be options to just leave a call back number or voicemail... |
| 13:26.23 | CrummyGummy | I was just thinking the same thing about what the user will hear. I don't really want them to hear the agent answering/voicemail/press 1. |
| 13:26.42 | Samot | Well no, they wouldn't. |
| 13:26.56 | Samot | I'm talking do they hear nothing but ringing for 5 minutes |
| 13:26.56 | CrummyGummy | I didn't even think about them dropping out of the call, won't the dialplan terminate? |
| 13:27.01 | Samot | Do they hear music or announcements? |
| 13:27.12 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:27.31 | Samot | Yes but there are Dial() settings to handle certain things during certain parts of the call setup/connection.. |
| 13:27.45 | Samot | Like when the caller disconnects the line |
| 13:27.48 | Samot | or the callee.. |
| 13:27.50 | Samot | Etc, etc. |
| 13:28.36 | CrummyGummy | I'll check it out, thanks. |
| 13:28.48 | Samot | As someone who has managed support call teams... |
| 13:29.37 | Samot | I'm not going to want to get my ass chewed out because someone called in, got someone else in the company and raised a huge stink because they had an issue after hours and my guys were f'ing off and the call flow left the user hanging.. |
| 13:30.14 | Samot | So in turn after getting my ass chewed... |
| 13:30.39 | Samot | I would go straight to the person in charge of the PBX and call flows and chew their ass out. |
| 13:31.20 | [TK]D-Fender | It's the circle of striiiiiffffeeee!!! |
| 13:31.27 | Samot | Oh yeah. |
| 13:32.12 | CrummyGummy | lol, I was just going to rotate through the numbers till someone answers. The CEO is in the list so everyone is accountable |
| 13:32.13 | Samot | One pissed off customer because they called because they are down and spends 30 minutes listening to ABBA and Kenny G while an agent won't wake up. |
| 13:32.19 | Samot | Hah |
| 13:32.20 | Samot | Dude. |
| 13:32.25 | Samot | You never want the CEO on the list. |
| 13:32.33 | Samot | Oh jesus. |
| 13:32.43 | CrummyGummy | It's a small company |
| 13:32.50 | CrummyGummy | he inisists |
| 13:32.52 | Samot | OK. |
| 13:32.55 | Samot | Yes. |
| 13:32.59 | Samot | I bet I know why. |
| 13:33.01 | CrummyGummy | has to go through 10 people first |
| 13:33.05 | Samot | Yup |
| 13:33.07 | Samot | That's why. |
| 13:33.10 | CrummyGummy | someone will wake up |
| 13:33.16 | Samot | If it gets to the CEO..someone is in trouble. |
| 13:33.17 | CrummyGummy | probably me... |
| 13:33.21 | CrummyGummy | yep |
| 13:33.29 | CrummyGummy | then the shit hits the fan |
| 13:33.36 | Samot | So |
| 13:33.48 | Samot | I'd start making this more then just a simple call people and press buttons. |
| 13:34.03 | Samot | Because that first call hits the CEO and it's a bad bad thing... |
| 13:34.34 | Samot | You're CEO is going to want to know history, details, reports. Questions will be asked and you will be expected to answer. |
| 13:34.37 | Samot | So I'd be ready for that |
| 13:34.50 | CrummyGummy | True |
| 13:35.18 | CrummyGummy | Atm I'm itemising that in the asterisk log so there is a trail |
| 13:35.39 | [TK]D-Fender | So log it yourself in your dialplan as you go |
| 13:35.45 | Samot | Logs are great. |
| 13:35.49 | CrummyGummy | yep |
| 13:35.50 | Samot | Logging data is awesome. |
| 13:35.58 | Samot | How do you use that data? |
| 13:36.07 | Samot | What are you actually doing with it? |
| 13:36.13 | [TK]D-Fender | Agent called. Start of dial time. Star of confirmation request time |
| 13:36.29 | Samot | Again, logging all that data is awesome. |
| 13:36.32 | Samot | It should be done. |
| 13:36.49 | Samot | But there is more to it than just logging lines of data. |
| 13:37.33 | CrummyGummy | We rarely get A/H calls, it'll just be to see who didn't wake up really. We've had one call this year so far. Just one arsehole that didn't wake up... |
| 13:37.52 | CrummyGummy | Logs will just be for the incident report really |
| 13:38.40 | CrummyGummy | lol, the dial timeout defaults to 136 years |
| 13:38.50 | CrummyGummy | should cover it. |
| 13:47.54 | *** join/#asterisk tenspeed705 (~keiths@209.91.144.146) |
| 13:48.34 | tenspeed705 | does anyone have a script they use to generate configs for SPA500(s) extension modules? don't really feal like wasting much time on making one |
| 13:51.10 | Samot | I do but it's designed for my systems. |
| 13:51.22 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:74cb:cd82:e518:2842) |
| 13:52.11 | pawiecki | tenspeed705: make a template and then find-replace. If it's more that few devices - make a simple script to automate it. |
| 13:52.50 | Samot | pawiecki: You missed the important part. |
| 13:53.01 | [TK]D-Fender | "I don't want to do the work myself" |
| 13:53.13 | Samot | Yup. |
| 13:53.28 | tenspeed705 | I think I may have to do that...I didn't really want to script it as I have 2 in production....they just like to change it every week or so, heh |
| 13:53.33 | [TK]D-Fender | Any takers on how many multiple of the time required will be spent waiting for someone to provide one he'd be happy with? |
| 13:53.43 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 13:53.43 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 13:53.58 | *** join/#asterisk asteriskmonkey (~philip@74.116.121.202) |
| 13:54.09 | asteriskmonkey | heya anyone do coding with ARI and php in here? |
| 13:54.23 | Samot | ~polls |
| 13:54.23 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 13:54.46 | *** join/#asterisk DivideBy0x0 (~DivideBy0@unaffiliated/divideby0x0) |
| 13:54.46 | *** mode/#asterisk [+o DivideBy0x0] by ChanServ |
| 13:55.20 | asteriskmonkey | sigh. |
| 13:55.32 | Samot | If the answer is yes, what then? |
| 13:55.37 | Samot | Because it probably is. |
| 13:55.45 | Samot | There's a 178+ people in here. |
| 13:55.53 | asteriskmonkey | can you initiate calls with ARI or do they have to be stasis apps |
| 13:56.18 | Samot | Channel or Channels endpoint. |
| 13:56.31 | Samot | Wiki shows that is originates calls. |
| 13:56.33 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-oabmdenlhmyorsjm) |
| 13:56.33 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:56.48 | Samot | s/is/it/ |
| 13:57.05 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
| 13:57.32 | asteriskmonkey | but you have to join channels in code right? its not like traditional ami where you set two targets |
| 13:58.18 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Channels+REST+API#Asterisk14ChannelsRESTAPI-originate |
| 13:59.01 | Samot | In fact, it's almost just like AMI Originate. |
| 13:59.21 | asteriskmonkey | oh i went down the wrong rabit hole |
| 13:59.27 | asteriskmonkey | was reading too much phpari docs |
| 13:59.36 | asteriskmonkey | that looks even similer |
| 13:59.41 | asteriskmonkey | simpler :) |
| 13:59.54 | Samot | doesn't even use ARI |
| 14:00.35 | Samot | Just for the record... |
| 14:01.06 | Samot | 9:55:53 AM A<asteriskmonkey> can you initiate calls with ARI or do they have to be stasis apps <-- That's the way to ask a question to get a faster response. |
| 14:01.12 | asteriskmonkey | samot : you have a better method for feeding 1000's of calls and getting reliable status back ? |
| 14:01.17 | asteriskmonkey | lol thanks :) |
| 14:01.33 | Samot | 9:54:08 AM A<asteriskmonkey> heya anyone do coding with ARI and php in here? <-- My answer to that is No but I still had your answer. |
| 14:01.57 | Samot | If I was to do ARI, that would be how I did it. |
| 14:02.38 | Samot | What "status" are you looking to get back? |
| 14:02.47 | Samot | A 200 OK for the fact the POST worked? |
| 14:02.55 | Samot | Or the result of the actual Dial()? |
| 14:03.01 | asteriskmonkey | I need live event status |
| 14:03.12 | asteriskmonkey | dialing, progess, connected, busy , congestion etc.. |
| 14:03.15 | Samot | Then just like with AMI, you need to watch for it |
| 14:03.41 | Samot | ARI is nothing more than AMI is a better and more RESTful format. |
| 14:03.48 | asteriskmonkey | trying to think of a sane way of diong this with the least amount of reseourse, this is probably better suited to be an api/damon using node.js |
| 14:03.59 | Samot | I don't know.. |
| 14:04.07 | Samot | I send 1800 calls via my own API |
| 14:04.21 | asteriskmonkey | your just tracking post results though right? |
| 14:04.25 | Samot | That just creates callfiles and dumps them into the outgoing... |
| 14:04.32 | asteriskmonkey | ah yeah thats simple enough |
| 14:04.42 | asteriskmonkey | but I am making something that gives realtime status of calls to people |
| 14:04.52 | asteriskmonkey | works great in ami in low volumes |
| 14:04.52 | Samot | OK. |
| 14:05.09 | Samot | I'm not sure how ARI will handle 1000 calls at once. |
| 14:05.15 | asteriskmonkey | it dosnt |
| 14:05.25 | asteriskmonkey | asterisk itself cant handle more than 50 in most boxes |
| 14:05.31 | Samot | uhm. |
| 14:05.32 | Samot | what? |
| 14:05.39 | Samot | calls? |
| 14:05.48 | asteriskmonkey | call concurrency can be way higher call setup is relativly low |
| 14:06.03 | Samot | I do 1800 calls in under two minutes. |
| 14:06.11 | asteriskmonkey | becuase your using .call files |
| 14:06.18 | Samot | On some. |
| 14:06.24 | Samot | One system is purely AMI |
| 14:06.35 | Samot | Does it just fine. |
| 14:06.41 | Samot | Now, I'm not monitoring events. |
| 14:06.51 | Samot | But it makes 1800 calls just fine. |
| 14:06.53 | asteriskmonkey | yea monitoring events beomes the issue |
| 14:06.56 | asteriskmonkey | .call files i can bomb all day |
| 14:07.09 | Samot | Well of course. |
| 14:07.11 | asteriskmonkey | ok going to test and abuse the ARI interface some see how it compares :) |
| 14:07.13 | Samot | You're making LIVE requests |
| 14:07.17 | Samot | Taking up resources. |
| 14:07.37 | asteriskmonkey | its only way to control concurrency nicley though |
| 14:07.48 | asteriskmonkey | and in a realtime fashion |
| 14:08.07 | Samot | What concurrency? |
| 14:08.14 | asteriskmonkey | call setups |
| 14:08.24 | asteriskmonkey | i wrote this in flash and ami some time ago |
| 14:08.24 | asteriskmonkey | http://massdialer.ca/ |
| 14:09.18 | asteriskmonkey | had a nice little slider to control the feed rate of the calls and keep calls in progress/connected at a certain level. |
| 14:10.22 | *** join/#asterisk newtonr (newtonr@nat/digium/x-umukcqbofiyvlptp) |
| 14:10.28 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:12.20 | Samot | Sounds like the box needs more resources. |
| 14:12.31 | Samot | Or you should be splitting this over multiple boxes. |
| 14:13.11 | asteriskmonkey | yes, i agree, main issue though intially was issue relating roung ami messaging, so ari will probably be the cure :) |
| 14:13.37 | asteriskmonkey | some people also love there 5$ vps's |
| 14:13.50 | asteriskmonkey | so have to test things to death for efficiency :) |
| 14:13.50 | Samot | I do too |
| 14:13.56 | Samot | But I don't use those for this kind of stuff. |
| 14:14.05 | Samot | I wouldn't run this on a $5 VPS |
| 14:14.07 | Samot | At all |
| 14:14.15 | asteriskmonkey | meh i know the feeling, you cant tell other people though hehe :/ |
| 14:14.23 | Samot | Sure you can. |
| 14:14.26 | Samot | I do it all the time. |
| 14:14.32 | Samot | "No, that's the wrong VPS plan for this |
| 14:14.36 | Samot | Easy. |
| 14:14.51 | asteriskmonkey | lol :) |
| 14:15.29 | Samot | Is someone paying you to provide this solution? |
| 14:15.37 | Samot | If so, tell them what to use that is right. |
| 14:16.09 | asteriskmonkey | no im making it freeware |
| 14:16.22 | Samot | Then you say "System Requirements" |
| 14:16.43 | Samot | There is no way a 1CPU 1GB RAM VM (or less) is going to sustain any high amounts of calls |
| 14:16.44 | asteriskmonkey | i tend to tourcher myself more for the sake of finding out what is the most efficient / how to make things more efficient, i believe that the increase in computing power != making something with heavy code |
| 14:16.52 | Samot | And run this monitoring software. |
| 14:17.00 | asteriskmonkey | sure, id love to be able to build a unit test for testing max box concurrency |
| 14:17.04 | asteriskmonkey | that would be brillant |
| 14:17.06 | Samot | Take the software away |
| 14:17.20 | Samot | Asterisk won't be able to sustain those amounts of calls on a 1x1 server. |
| 14:17.22 | Samot | At all |
| 14:17.39 | Samot | Running with the bare minimum, not happening. |
| 14:17.42 | asteriskmonkey | is there a good test to see how many calls with audio/record a box can do ? |
| 14:17.55 | asteriskmonkey | or does that need some sipp testing etc.. |
| 14:18.06 | Samot | You can't just run sipp testing. |
| 14:18.14 | Samot | You're doing more than just making calls. |
| 14:18.30 | Samot | Are you doing recording? |
| 14:18.35 | asteriskmonkey | would be nice if there is like an ./astbench.sh -with record .. somewhere :) |
| 14:18.51 | asteriskmonkey | no.. its a hypothetical as usually disk io is a bottleneck |
| 14:19.33 | Samot | How many overall concurrent calls will be done? |
| 14:19.41 | *** join/#asterisk kharwell (kharwell@nat/digium/x-jlngcnoiuboszeeh) |
| 14:19.41 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:20.14 | asteriskmonkey | its variable |
| 14:20.20 | Samot | I understand that |
| 14:20.23 | Samot | What is the bar |
| 14:20.31 | asteriskmonkey | ah on my bar i always made 20 max |
| 14:20.39 | asteriskmonkey | just for small sanity sakes |
| 14:20.47 | Samot | What is this going to be used for? |
| 14:21.11 | asteriskmonkey | whatever people want it for |
| 14:21.18 | Samot | What does it do? |
| 14:21.23 | Samot | Just monitor call events? |
| 14:21.50 | asteriskmonkey | no, its a little call center dialer, you load it up with messages and text to speech bubbles and it concats the info and dials/ plays the string of things to people |
| 14:21.58 | Samot | Uhm. |
| 14:22.04 | Samot | And your benchmark is 20 calls? |
| 14:22.07 | asteriskmonkey | it has a callback function to connect back to quuees thats where the slider aids people that do stuff with connect backs |
| 14:22.14 | asteriskmonkey | no, it can do more |
| 14:22.22 | asteriskmonkey | i just set that as a sane limit |
| 14:22.38 | Samot | For really small call centers. |
| 14:22.42 | asteriskmonkey | yeah |
| 14:22.43 | Samot | Like really small. |
| 14:22.58 | asteriskmonkey | im making this for 0$ like seriously |
| 14:23.04 | Samot | fop2.com |
| 14:23.08 | Samot | You should look there first. |
| 14:23.17 | Samot | Or AsterNIC's website. |
| 14:23.21 | Samot | That do this for free and paid. |
| 14:24.47 | asteriskmonkey | call center stats != call center control / setup |
| 14:24.47 | Samot | Wallboards, Popups, Reporting, Spy/Whisper/Barge-in, On-Demand recording... |
| 14:24.54 | Samot | No |
| 14:24.55 | asteriskmonkey | nice stuff though way well developed |
| 14:25.10 | asteriskmonkey | I should go dig there code base see what there doing with events |
| 14:25.13 | Samot | What do you mean by control/setup? |
| 14:25.24 | asteriskmonkey | you can load an xls list in mine and run it |
| 14:25.29 | Samot | OK. |
| 14:25.37 | Samot | So it makes outbound calls |
| 14:25.44 | asteriskmonkey | yessssss. |
| 14:25.51 | Samot | FreePBX has a module for that |
| 14:25.56 | Samot | Vicidial does that |
| 14:26.01 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-sesspzcjljqfaion) |
| 14:26.01 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:26.06 | Samot | There are various programs out there that do this. |
| 14:26.15 | asteriskmonkey | yes why not one more thats easier to use |
| 14:26.24 | Samot | But what does it do? |
| 14:26.28 | Samot | Those programs have limits |
| 14:26.33 | Samot | They have structure.. |
| 14:26.51 | Samot | They also don't care about the server. |
| 14:27.05 | asteriskmonkey | sure, they leave that to the customer to figure out |
| 14:27.20 | asteriskmonkey | i was aiming for simple dialout / mesaging playback solution for people |
| 14:27.24 | asteriskmonkey | no bigge |
| 14:27.27 | Samot | Those exist. |
| 14:27.40 | asteriskmonkey | sure so do cars, but people make new models each year |
| 14:27.46 | Samot | Right |
| 14:27.59 | Samot | But they also make models that are different in some way then the rest. |
| 14:28.14 | asteriskmonkey | dont know about that, seems cars have had 4 wheels for the longest time |
| 14:28.21 | Samot | Right |
| 14:28.25 | Samot | But they all don't have blue tooth |
| 14:28.27 | Samot | Or wifi |
| 14:28.31 | Samot | Or sat radio |
| 14:28.36 | Samot | Or power locks |
| 14:28.42 | asteriskmonkey | meh that becomes standerd in time like everything |
| 14:28.44 | Samot | etc, etc etc |
| 14:28.48 | Samot | Right |
| 14:28.49 | asteriskmonkey | this is the life of a software dev :/ |
| 14:28.52 | Samot | What I am saying is.... |
| 14:28.55 | Samot | Go look at the market |
| 14:29.01 | Samot | See what is standard... |
| 14:29.33 | Samot | Before you release a piece of software that no one is going to use because is lacks a feature or features that all other programs have. |
| 14:29.45 | asteriskmonkey | people over feature things |
| 14:30.04 | asteriskmonkey | im making simple tools for simple non-tech people |
| 14:30.25 | Samot | I understand that. |
| 14:30.29 | Samot | The interface looks good. |
| 14:31.16 | asteriskmonkey | thanks :) , ive done alot of research there are some really monsterous call center type softwares out there |
| 14:31.20 | Samot | Just don't shoot yourself in the foot. |
| 14:31.25 | asteriskmonkey | thanks :) |
| 14:31.34 | asteriskmonkey | i do that alot, im running out of toes, but im learning :) |
| 14:32.47 | Samot | Just remember that when you release it, you are responsible for it. |
| 14:34.23 | asteriskmonkey | hits release then disconnects internet and hides in closet |
| 14:35.04 | Samot | There is a reason I give AsterNIC/FOP2 and others $$$ |
| 14:35.19 | Samot | Or use their free versions. |
| 14:35.31 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 14:35.31 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:35.36 | Samot | 1) The free versions rarely require support or have issues. |
| 14:36.03 | Samot | 2) The paid versions rarely require support or have issues and when they do, quick turn around on resolution. |
| 14:39.43 | Samot | So I would bench mark this at about 100 calls or more. |
| 14:40.12 | Samot | Because that's probably going to be more the avg case use. |
| 14:40.40 | Samot | I'd go with a 2CPUx4GB RAM VM |
| 14:41.14 | Samot | That should be enough to handle that call volume and your event monitoring.. |
| 14:42.05 | Samot | Because the honest truth is no real call center or anyone doing outbound sales campaigns are going to be using a $5 VPS. |
| 14:42.25 | *** join/#asterisk cervajs2 (~cervajs2@178.148.broadband4.iol.cz) |
| 14:43.00 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
| 14:44.46 | cervajs2 | i want move ps_contacts from odbc back to astdb. whats the recommended configuration in sorcery.conf/extconfig.conf? |
| 14:45.32 | file | not defining it for contacts will use the default which it astdb |
| 14:45.34 | file | er is |
| 14:46.34 | cervajs2 | file: ill try. tnx |
| 15:01.19 | *** join/#asterisk kamyl (~user@unaffiliated/kamyl) |
| 15:04.51 | cervajs2 | this works. now i want change identify=realtime,ps_endpoint_id_ips to pjsip.conf file. i cant find example howto do this. tips? |
| 15:05.13 | file | the default is pjsip.conf, just don't define it |
| 15:06.05 | cervajs2 | i tried this. but pjsip show settings show endpoint_identifier_order : ip,username,anonymous after restart |
| 15:06.27 | cervajs2 | in pjsip.conf i have endpoint_identifier_order=username,ip,anonymous |
| 15:13.01 | cervajs2 | changed user_agent to some test string and its look like pjsip.conf global section is not parsed |
| 15:13.37 | cervajs2 | or pjsip show settings show bad things |
| 15:14.01 | cervajs2 | ill create pastebin of my config |
| 15:17.15 | *** join/#asterisk yuka_ (~elegast@5.79.86.129) |
| 15:17.53 | cervajs2 | https://pastebin.com/YVFwVsMD |
| 15:27.16 | *** join/#asterisk kamyl (~user@unaffiliated/kamyl) |
| 15:44.17 | *** join/#asterisk dunderproto (~dunderpro@122-117-44-11.HINET-IP.hinet.net) |
| 15:45.30 | *** join/#asterisk spicyramen (~Adium@207.198.105.19) |
| 15:50.50 | zaf | what's the softphone of choice these days for windows? |
| 15:57.49 | *** join/#asterisk elcontrastador (~textual@c-67-182-43-55.hsd1.ca.comcast.net) |
| 16:17.27 | *** join/#asterisk kunwon1 (~kunwon1@unaffiliated/kunwon1) |
| 16:24.34 | *** join/#asterisk spicyramen (~Adium@104.132.1.77) |
| 16:37.15 | *** join/#asterisk miralin (~Thunderbi@91.237.94.9) |
| 16:40.41 | igcewieling | My new Zenpad is so terrible I wrote my first review on Amazon. 8-| |
| 16:49.30 | *** join/#asterisk captain118 (uid167508@gateway/web/irccloud.com/x-qagdcsdbdebqpywc) |
| 16:51.37 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:28.46 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:45.25 | mub | test |
| 17:45.58 | *** join/#asterisk mub (~jub@static-173-53-12-18.rcmdva.fios.verizon.net) |
| 17:54.35 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:19.32 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 18:21.16 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 18:21.16 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 18:31.53 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 18:35.05 | *** join/#asterisk hereIam (~kid@fobos.ion.team) |
| 18:38.12 | hereIam | hi guys! let me introduce a small but probably interesting project https://github.com/roman-yerin/ion.phone |
| 18:41.23 | salviadud | hereIam, looks like a very simple interface |
| 18:41.55 | hereIam | salviadud: so - is it good or not? :) |
| 18:44.12 | salviadud | I just took a look at the demo image, hehe |
| 18:44.31 | salviadud | Don't know if I should compile this on my fedora |
| 18:45.01 | hereIam | you should not, it is running inside your browser |
| 18:45.38 | salviadud | takes a look at the documentation |
| 18:45.44 | salviadud | ohhhh, you're right |
| 18:47.21 | hereIam | I just hope it can save some life hours for somebody :) |
| 18:49.01 | elegast | Is anyone using asterisks with sippy b2bua? I like the fact that it's pure python, and thus easy to hack on/integrate/build services on top, or should I be using a full fledged sip server/router like kamaillio or opensips instead? |
| 18:49.36 | [TK]D-Fender | elegast, Depends what you want obviously |
| 18:49.44 | [TK]D-Fender | elegast, We can't answer that for you |
| 18:50.58 | hereIam | asterisk is a b2bua itself, isn't it? |
| 18:51.49 | elegast | Well, I'd like to do call routing policy logic and authentication in python, it seems light weight enough, but im unsure if this what b2bua's are for, or if I should use something like opensips |
| 18:52.52 | [TK]D-Fender | depends if you really don't want * choosing where your calls go or not. |
| 18:53.05 | hereIam | if the only one thing you need is authentication, use kamailio or opensips |
| 18:53.19 | [TK]D-Fender | What is the point of having this layer separate from * for you? |
| 18:54.41 | hereIam | b2bua is for topology hiding and for some PBX functions |
| 18:54.43 | elegast | Well, I don't like putting a pbx straight on the internet foremost, and I want control of how connections are routed, both at IP level and the sip signalling level, I'm thinking asterisk alone can't do those things |
| 18:56.02 | [TK]D-Fender | Explain your view on the limitation of * concerning "routing" |
| 18:56.06 | elegast | e.g. certain sip endpoints could be roaming devices, which might not work out of the box across networks, so i need something inbetween that can rewrite sip data and such |
| 18:56.34 | [TK]D-Fender | Can you show a case where * doesn't work? |
| 18:57.09 | hereIam | [TK]D-Fender: * is more than needed if he just need to change some sip packets |
| 18:57.27 | [TK]D-Fender | I don't know that yet... |
| 18:57.32 | [TK]D-Fender | Which is why I'm asking |
| 18:57.58 | [TK]D-Fender | I'm just trying to get a solid case for the effort of putting a middle-man in there |
| 18:59.05 | hereIam | it's not needed to say that kamailio and opensips are much more productive than * |
| 18:59.39 | hereIam | so if authentication and routing is everything he needs... |
| 19:00.07 | [TK]D-Fender | Again, lets just find out from him.... |
| 19:00.51 | [TK]D-Fender | Get the real problem specified and verified that something external actually adds value |
| 19:01.17 | hereIam | he just wrote that :) well, ok, he didn't said that's all he need :) |
| 19:02.05 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 19:02.24 | elegast | I don't technically need asterisk for its pbx logic, theres currently a fritzbox in an office somewhere, it's behind multiple isps. The fritzbox is currently performing the pbx function, but sip connections need to go trough a datacenter, where roadwarriors can connect to also, so that both roadwarriors and the fritzbox can use the same sip trunk |
| 19:03.11 | elegast | but the roadwarriors themselves must not connect to the office network directly |
| 19:03.32 | [TK]D-Fender | "must not" because? |
| 19:03.41 | [TK]D-Fender | Arbitrary mandate? |
| 19:06.18 | elegast | The office is a home office, and the connection at times unreliable (though multiple isp connections, its consumer lines) having remote users connect to the office first and then call out, introduces extra lag, and some networks don't work well with nat and so might need to be rtpproxied across |
| 19:07.42 | [TK]D-Fender | Sounds like this wholse setup is better off being hosted offiste then |
| 19:08.56 | elegast | I agree, but the client is uncomfortable moving the 'pbx' offsite .. |
| 19:09.39 | [TK]D-Fender | And uncomfortable being on the inside of a nested and unstable connection |
| 19:10.13 | [TK]D-Fender | I wouldn't want to leave the remote ones stranded because of this |
| 19:10.34 | elegast | Ideally, I'd like to replace the fritzbox behind the consumer network, with asterisk and move it inside the datacenter too. |
| 19:14.00 | elegast | My assumption is that this sort of thing can't be done with asterisk alone, is that correct? |
| 19:15.50 | hereIam | no, it can be done with asterisk. just need to make something for safety |
| 19:16.47 | [TK]D-Fender | In as much as you should do that for any given layer |
| 19:17.15 | [TK]D-Fender | if it just proxies into the interior the same general kind of vulnerabilities can exist on the outside layer leading to you getting attacked the same ways |
| 19:19.23 | elegast | Right, thats why I was looking towards sippy, it terminates both call legs and then re-establishes them, keeping track, allowing for enforcements of rfcs and such + codec negotiation |
| 19:24.51 | [TK]D-Fender | How is that more restrictive than what you can do with * itself? |
| 19:24.55 | [TK]D-Fender | or "better" |
| 19:28.37 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 19:28.43 | elegast | Well, I don't know *'s full capabilities yet, so I can't really answer that, my assumption is the fact that many exploits would be tailored to asterisks, having something inbetween that *only* does sip packets, and can ensure everything is proper, seems like added value |
| 19:29.22 | [TK]D-Fender | Why would you leave the non-SIP bits exposed anyway? |
| 19:29.47 | [TK]D-Fender | You don't seem ot understand * yet |
| 19:30.17 | [TK]D-Fender | So you should start by actually learning and using it first before you go trying to shove middlemen into the process and com0plicting your install before you have a justified reason to do so |
| 19:30.18 | elegast | You're right, I don't yet |
| 19:30.22 | elegast | Which is why I'm asking |
| 19:31.12 | [TK]D-Fender | You're imagining problems you don't have |
| 19:31.13 | elegast | Well, I don't need the asterisk, at this point in time anyway, I need the middle man parts of sippy |
| 19:31.58 | [TK]D-Fender | Well if sippy does everything you want ... and you were considering putting it between * and your other devices .... then is sounds like you didn't need * in the first place |
| 19:32.07 | [TK]D-Fender | So I'm not sure why you were thinking of this arrangement at all... |
| 19:33.00 | elegast | Well, originally the question was wether or not asterisk + sippy would be a good choice, thinking ahead, possibly replacing the fritzbox pbx with asterisk |
| 19:33.13 | elegast | at a future point in time |
| 19:35.49 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 19:35.49 | *** mode/#asterisk [+o ChanServ] by card.freenode.net |
| 19:41.18 | *** join/#asterisk jkroon (~jkroon@165.16.204.174) |
| 19:44.37 | *** join/#asterisk LunaLovegood (~alice@75.98.139.193) |
| 19:48.21 | LunaLovegood | I have a contact=... line for each [type=aor], but the ATAs also send REGISTER messages and then the contact entries end up duplicated (two contacts, but same IP/user), and INVITEs are sent twice when someone calls the phone. Is there a way to make Asterisk respond "SIP/2.0 200 OK" to REGISTER message, but otherwise ignore them? |
| 20:06.34 | [TK]D-Fender | Nope |
| 20:06.48 | [TK]D-Fender | When not just let them register alone? |
| 20:08.07 | *** join/#asterisk spicyramen (~Adium@104.132.1.77) |
| 20:11.33 | LunaLovegood | They're weird and buggy and they don't work if I try to use passwords. So I do it all with fixed IP addresses. |
| 20:11.57 | LunaLovegood | I have contact= and contact_permit= lines in my pjsip.conf |
| 20:12.44 | LunaLovegood | And I can't stop them from sending REGISTER requests even if they're redundant in this case. |
| 20:14.58 | [TK]D-Fender | Fix the clients |
| 20:15.05 | [TK]D-Fender | If they're that screwed up you can't trust them |
| 20:15.39 | sawgood | LunaLovegood: I know that feeling ... |
| 20:15.51 | sawgood | can't stop them from sending REGISTER requests |
| 20:15.57 | [TK]D-Fender | So accept them |
| 20:16.00 | [TK]D-Fender | and use them |
| 20:16.11 | LunaLovegood | can't change them since it would be expensive, they're modem/ata/router/wifi combos |
| 20:16.50 | sawgood | for me: the username/secret are not part of the context anymore (because) we changed to host=staticIP |
| 20:17.23 | sawgood | I took those fields out ... (but) I'm testing the account on another Asterisk box (to build as a testing box) |
| 20:17.58 | LunaLovegood | where is the host= option? can't find it in the pjsip help |
| 20:18.33 | sawgood | chan_pjsip ... I'm using chan_sip |
| 20:18.46 | LunaLovegood | oh i see. |
| 20:19.47 | LunaLovegood | I'm using it for connecting to the SIP trunk, but I couldnt find a set of option that could get it to work for some of my ATAs. So chan_sip on one side and res_pjsip on the other. |
| 20:21.10 | sawgood | what brand of ATA? |
| 20:21.17 | LunaLovegood | Arris TG1672G |
| 20:21.39 | sawgood | nice ... is that your box or the providers? |
| 20:22.34 | LunaLovegood | what do you mean? It's the one I give to my clients since it does all in one (Internet/WiFi/phone) and has a battery for power outages |
| 20:23.27 | sawgood | LunaLovegood: nice ATA ... its rare for me to see Arris ouside of a large providers hands (nice) to know you have your own ... for your own customers |
| 20:23.39 | sawgood | how many FXS ports does that box give you? |
| 20:24.14 | LunaLovegood | two. FXS are the one you plug phones in, right? |
| 20:24.43 | sawgood | yep ... only 2 huh? What other ports do you get? |
| 20:25.17 | LunaLovegood | It's got a 4 x GbE switch and 2.4 and 5.8 GHz WiFi. |
| 20:25.59 | LunaLovegood | The wifi is not AC though, only N. |
| 20:27.49 | LunaLovegood | Would specifying the ip address in my Dial call help? |
| 20:28.05 | LunaLovegood | Like Dial(PJSIP/Cx5ax7qN6W@172.19.4.227) or something? |
| 20:28.36 | sawgood | dual band AC is hot right now I hear |
| 20:29.11 | LunaLovegood | AC is only for 5 GHz tho |
| 20:29.13 | *** join/#asterisk libardi (~libardi@179.159.11.133) |
| 20:29.30 | sawgood | for those two FXS ports (are you) running phone cables from the Arris to standard phones (analog)? |
| 20:31.03 | LunaLovegood | yeah, tho since the customer owns the wire, you can often just unplug the PSTN in the basement (but leave all the wires still connected together) and connect the ATA in any of the existing wall outlets. |
| 20:32.15 | sawgood | nice ... when that actually works .. local cross-connect |
| 20:33.23 | sawgood | In your Arris: GUI: you have a SIP trunk (leaving out) to a proivder (or) to your own swith offsite? |
| 20:33.43 | sawgood | switch off-site ... |
| 20:34.16 | LunaLovegood | sip trunk privider in montreal |
| 20:34.45 | drmessano | LunaLovegood: Are you even using the Arris as a cable modem or just as a client router? |
| 20:34.45 | LunaLovegood | I wouldn't have the time to learn and setup all that myself. |
| 20:35.22 | LunaLovegood | Cable modem of course. I don't think you can set it up to send/receive the phone data over the ethernet ports. |
| 20:35.53 | drmessano | How do you get around the CableCo provisioning the box so that the customer can't access the admin for the phone side? |
| 20:36.02 | drmessano | and by customer, I mean, you or them |
| 20:36.37 | LunaLovegood | The cable company is mine. And the SIP provisioning can only be done from the RF port side. |
| 20:36.42 | LunaLovegood | AFAIK |
| 20:36.54 | LunaLovegood | I haven't tried hacking the firmware |
| 20:36.56 | drmessano | Oh so you are the cable provider |
| 20:36.59 | sawgood | he owns the Arris himself (the Arris) does not belong to the cable company |
| 20:37.12 | drmessano | sawgood: No, he is the cable company |
| 20:37.16 | sawgood | kind of cool if you ask me ... I do not see that too often |
| 20:37.56 | drmessano | So the answer is much more obvious |
| 20:38.58 | drmessano | I was wondering, if you weren't the one controlling the box, how you had access to provision the voice ports. heh |
| 20:39.37 | LunaLovegood | There's tons of peculiarities to each kind of modem so we (my cousin and I) have a lot of "fun" trying to get them to work. |
| 20:40.43 | drmessano | I'd love to find a simple ATA with a battery |
| 20:41.18 | sawgood | I know they are costly .... |
| 20:41.30 | LunaLovegood | You can get a 40$ Cisco one and a 50$ UPS? It doesn't use a lot of power so even the smallest UPS should last hours |
| 20:41.43 | LunaLovegood | probably cheaper in US dollars |
| 20:41.47 | drmessano | LunaLovegood: I could do a lot of things. I just wanted that specific config |
| 20:43.31 | drmessano | There is still sufficient loss in the DC <> AC <> Conversion that's an awful waste |
| 20:43.31 | LunaLovegood | How about a motorcycle battery and a 12 V brick? (or maybe 13-14 V so it actually charges the battery) Just plug them together with the ATA and it should work I think? |
| 20:43.48 | drmessano | Sure, but deploying that customers? |
| 20:44.01 | LunaLovegood | oh yeah sorry, I thought you meant for yourself |
| 20:44.18 | drmessano | I have a 12v roll running the length of my place |
| 20:44.22 | drmessano | I have a 12v rail running the length of my place |
| 20:44.40 | drmessano | Attached to batteries 3x the size of a car battery |
| 20:44.42 | *** join/#asterisk aness (~aness@cm-84.209.155.137.getinternet.no) |
| 20:44.55 | drmessano | *I* have plenty of backup power |
| 20:45.02 | LunaLovegood | that's nice, UPS's should be more efficient if they don't have to boost the voltage back to 120V |
| 20:45.35 | drmessano | Just would be nice to have an ATA with a battery to deploy |
| 20:46.51 | sawgood | I have one (four ports) with battery ... (cost 620 new though) |
| 20:46.51 | LunaLovegood | The ATA is useless tho if the modem/router goes offline during power outages though. So why not an ordinary UPS for both? |
| 20:47.25 | drmessano | Most ISP provided modems have batteries now |
| 20:47.37 | LunaLovegood | unless it's for an appartment building and you have ethernet everywhere? |
| 20:47.39 | LunaLovegood | oh. |
| 20:48.20 | LunaLovegood | I don't think they do unless the client also buys their phone service.? |
| 20:48.41 | drmessano | They're shipping out the same box either way |
| 20:49.27 | drmessano | It's cheaper that way, and they can upsell them on voice for years after |
| 20:49.45 | LunaLovegood | The batteries need to be changed every ~5-6 years, I don't think they'll bother for the customers who don't buy the phone service. |
| 20:50.08 | drmessano | Probably not |
| 20:50.11 | LunaLovegood | plus it gets annoying receiving all those SNMP traps about battery needing replacement. |
| 20:51.34 | sawgood | out here, Comcast uses the Arris all day / night (for) their "analog" service handoff (and) that box has a battery in it |
| 20:51.45 | sawgood | that would be cool if we could get that box on the cheap! |
| 20:52.08 | sawgood | actualy, that version is for coaxial, so forget that |
| 20:53.11 | drmessano | All of them are |
| 20:53.39 | sawgood | oh ... thats why then ... got it |
| 20:53.45 | LunaLovegood | If you use the same Arris (TG1672G) as me, be careful when handling them. The plastic stubs that hole the cpu heatsink in place are really weak. |
| 20:54.06 | drmessano | LunaLovegood: Im not a CableCo, so that doesn't work for me at all |
| 20:54.25 | drmessano | If I can't provision the voice ports, it's useless |
| 20:55.03 | drmessano | Probably goes for most of us :) |
| 20:55.23 | sawgood | Adtran is rather costly |
| 20:56.15 | drmessano | Nothing wrong with a Cisco ATA |
| 20:57.08 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 21:06.22 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:14.16 | sawgood | drmessano: do you use any Asterisk based ATAs? instead of a Cisco one? |
| 21:14.47 | [TK]D-Fender | no such thing.... |
| 21:15.31 | sawgood | well: re-wording: a ATA which the engine software is Asterisk under a Linux-type OS ... |
| 21:15.47 | drmessano | Ive never seen an Asterisk ATA |
| 21:15.56 | sawgood | I have one ... |
| 21:16.01 | [TK]D-Fender | Why would a dumb-shit ATA have a PBX CORE in it? |
| 21:16.08 | drmessano | Who the hell makes one? |
| 21:16.09 | [TK]D-Fender | that is a total waste of resources |
| 21:16.30 | drmessano | sawgood: made by who? |
| 21:16.44 | drmessano | Why would you even want that? |
| 21:17.06 | sawgood | 1 port ATA = Yeastar TA100 or 2 port ATA Yeastar TA200 |
| 21:17.11 | sawgood | they work rather nice |
| 21:17.15 | drmessano | ROFL |
| 21:17.18 | drmessano | No, just no |
| 21:17.23 | drmessano | Yeastar is total shit |
| 21:17.27 | sawgood | ha ha ha! |
| 21:17.41 | drmessano | Why would you even |
| 21:17.46 | sawgood | Also: Xorcom makes them too |
| 21:17.58 | drmessano | Huh? |
| 21:18.01 | sawgood | funny = you think Yeastar is shit! |
| 21:18.19 | drmessano | Xorcom makes what? |
| 21:18.50 | [TK]D-Fender | sawgood> 1 port ATA = Yeastar TA100 or 2 port ATA Yeastar TA200 <- Where do we see this? |
| 21:19.37 | drmessano | Im curious about this Xorcom claim |
| 21:19.51 | drmessano | What proof do you have of it, as well |
| 21:19.52 | [TK]D-Fender | Xorcom does have appliances with * built in |
| 21:19.55 | sawgood | drmessano: you like Xorcom, huh? |
| 21:20.10 | sawgood | Atoc and Zycoo have them too |
| 21:20.21 | [TK]D-Fender | But that isn't an "ATA running asterisk", that is a "Turnkey small PBX with a couple of PORTS built in" |
| 21:20.24 | drmessano | I asked you about a Xorcom ATA with Asterisk? |
| 21:20.38 | drmessano | [TK]D-Fender: Indeed |
| 21:20.40 | [TK]D-Fender | sawgood, Xorcom != ATA |
| 21:20.54 | sawgood | Fender: check this URL out: http://www.yeastar.com/analog-telephone-adapter/ |
| 21:20.57 | [TK]D-Fender | I didn't just buy a Sony stereo with a CAR built in |
| 21:21.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 21:21.02 | sawgood | TA100 and TA200 ... they work well (For me) |
| 21:21.24 | [TK]D-Fender | where do I see Asterisk built into that? |
| 21:21.32 | sawgood | They support SSH right to Asterisk 13! |
| 21:21.53 | sawgood | The GUI front end for setup is clean too |
| 21:22.31 | [TK]D-Fender | Show me this Yeastar ATA running *...... |
| 21:23.09 | sawgood | without giving you an IP to SSH (which I could) ... I don't know how to prove it to you |
| 21:23.30 | sawgood | I have them TA200 in many places registered with a SIP path (asterisk to asterisk) delivering phone calls |
| 21:23.48 | [TK]D-Fender | Go do that then |
| 21:23.52 | drmessano | Zycoo isn't an ATA |
| 21:23.53 | [TK]D-Fender | This i want to see... |
| 21:24.06 | sawgood | ha ... I'd rather give you one (and mail it to you) as a gift! |
| 21:24.19 | [TK]D-Fender | I'd accept getting into this one right now |
| 21:24.53 | sawgood | you don't think the TA200 could run Asterisk 13? |
| 21:25.08 | [TK]D-Fender | "think"? |
| 21:25.11 | [TK]D-Fender | SHOW ME |
| 21:25.16 | [TK]D-Fender | I don't do guessing games. |
| 21:25.29 | [TK]D-Fender | WHY would anyone bake * into a dumb ATA? |
| 21:25.48 | [TK]D-Fender | But letting that concept alone slide .. SHOW ME |
| 21:26.24 | sawgood | its not dumb for an ATA to use Asterisk (actually) it makes things smooth ... |
| 21:26.37 | drmessano | Not in the slightest |
| 21:26.41 | [TK]D-Fender | No other ATA needs a PBX CORE |
| 21:26.43 | drmessano | There's no advantage to it |
| 21:26.50 | [TK]D-Fender | there is nothing * offers to make anything "smooth" |
| 21:26.52 | sawgood | huge advantage! |
| 21:27.06 | drmessano | sawgood: What advantage? |
| 21:27.09 | drmessano | Explain |
| 21:27.10 | [TK]D-Fender | PM CONNECTION DETAILS |
| 21:27.14 | [TK]D-Fender | I want to see this thing |
| 21:27.29 | sawgood | you got it ... but give me a short bit ... I have a few things in front of this |
| 21:27.36 | drmessano | sawgood: What advantage? |
| 21:27.38 | drmessano | Explain |
| 21:27.42 | drmessano | I think thats fair |
| 21:27.48 | [TK]D-Fender | sawgood, Or TeamViewer your PC |
| 21:27.52 | [TK]D-Fender | Which would be easier |
| 21:28.02 | sawgood | one advantage is = you can see "both" sides of a circuit (and) troubleshoot between to known engines (with dual) SSH sessions ... |
| 21:28.06 | [TK]D-Fender | (assuming no OS issues) |
| 21:28.30 | sawgood | asterisk on one side and Asterisk on the other side (for) communcations that is sweet! |
| 21:28.56 | *** join/#asterisk nny (40149c0e@gateway/web/cgi-irc/kiwiirc.com/ip.64.20.156.14) |
| 21:29.04 | sawgood | Xorcom calls their devices "channel banks" but they are Asterisk based |
| 21:29.30 | sawgood | Xorcom Astribank! |
| 21:31.02 | drmessano | That's a channel bank.. not an ATA |
| 21:31.19 | drmessano | The context is important here |
| 21:32.34 | sawgood | yeah but its close and has FXS ports and runs asterisk: so close to an ATA (but costly) |
| 21:32.48 | sawgood | Yeastar has the TA200 (2 port) for like $39 dollars |
| 21:32.51 | sawgood | I have many on the shelf |
| 21:33.10 | drmessano | I can buy a SPA112 for the same price |
| 21:33.59 | sawgood | I like that unit too, but having SSH to your ATA and seeing what is happening from the CLI is "too cool" |
| 21:37.44 | sawgood | Although: probably long gone is that other "Aterisk based" 2 port ATA by Atcom .... |
| 21:38.06 | sawgood | I think thought from memory: those units were Asterisk 1.6 at best ... |
| 21:42.33 | [TK]D-Fender | is waiting on direct or TV guided tour of this thing... |
| 21:43.03 | sawgood | I have one: but funny: I cannot recall the password to it ... so I'll factory reset and get that to you |
| 21:43.25 | sawgood | I leave one setup for on the fly testing, but I have it down that I do not need it that much anymore |
| 22:13.30 | salviadud | wonders if that ATA is legal |
| 22:57.00 | KNERD | legaL? |
| 23:00.15 | salviadud | Well, I forget what gpl version * uses |
| 23:00.35 | file | 2. |
| 23:12.02 | *** join/#asterisk elegast (~elegast@5.79.86.129) |
| 23:16.48 | salviadud | Has anyone integrated skype for business server with asterisk? |
| 23:17.06 | KNERD | i thought that was depreciated by M$ |
| 23:17.24 | salviadud | really? I recently download an iso for 2015 version |
| 23:18.35 | salviadud | Anyway, I was just wondering if it's as easy as getting my agents a ringroup |
| 23:18.47 | *** part/#asterisk kharwell (kharwell@nat/digium/x-jlngcnoiuboszeeh) |
| 23:19.30 | salviadud | Say, a call comes in from the pstn, * answers, when I ask for their extension, I Dial(SIP/localextension&SIP/skypextension@siptrunkfromskype) |
| 23:20.04 | salviadud | If it's as easy as that, then, kudos for me. |
| 23:20.40 | salviadud | The weird part would be if a call comes in from native skype, to make it ring on * |
| 23:25.11 | salviadud | mentions skype and the chan goes silent. |
| 23:25.27 | salviadud | Please excuse my nonsense, skype will never be as cool as iax2 |
| 23:30.36 | Samot | Skype for Business is the latest MS voice venture. |
| 23:30.49 | Samot | salviadud: Does it do SIP trunks/peers? |
| 23:31.10 | salviadud | It's supposed to, yeah |
| 23:31.28 | Samot | OK well then that's how it would talk to Asterisk. |
| 23:31.31 | salviadud | I was about to install skype server but I found out my domain controller might not be replicating. |
| 23:31.47 | salviadud | And that's another story, somebody has to fix it, maybe it has to be me. |
| 23:32.18 | salviadud | sip addresses can be in the form of e-mail? |
| 23:32.39 | Samot | No. |
| 23:32.50 | Samot | SIP addresses are in the form of an URI |
| 23:32.54 | Samot | The same as email. |
| 23:33.03 | Samot | sip:user@domain:port |
| 23:33.06 | Samot | ^^ SIP URI |
| 23:33.19 | salviadud | That's what I meant Samot |
| 23:33.38 | salviadud | I didn't mean that sip addresses have subject and body |
| 23:34.08 | salviadud | Well, it should be pretty straightforward, as soon as I get it working I'll share my findings. |
| 23:34.15 | salviadud | or bumps. |
| 23:48.15 | drmessano | He's gonna install Skype on a DC? |