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| 00:30.35 | klow | Anyone know a quicky to enable opus before building without entering the make menuselect? |
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| 01:18.21 | klow | So I'm succesfully calling an echo test with Bria for Mac, Asterisk 14.2 , opus codec, echo test works. But after 15 seconds or so, the call drops, and my Bria log says some stuff about certificate name mismatch, showing me the IP address of my asterisk server, rather than the hostname which the TLS cert actually contains. I'm guessing Bria is getting this IP address from the SDP during the call setup .. anyone have an idea about that? |
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| 05:59.20 | igcewieling | Can anyone recommend a left handed mouse? |
| 06:00.47 | Samot | That a serious question? |
| 06:01.32 | igcewieling | yes. |
| 06:02.03 | Samot | Well the last time I checked I just set the mouse to switch sides. |
| 06:02.09 | igcewieling | I searched on amazon and newegg but there are so many of them,. |
| 06:02.32 | igcewieling | Samot: many ergonomic mice are not symmetrical like that. |
| 06:02.40 | Samot | Oh. |
| 06:02.56 | igcewieling | Including the right handed mouse I moved to the left. 8-| |
| 06:03.13 | Samot | That was a descriptor that helped. |
| 06:08.50 | igcewieling | I'm getting old. 8-( |
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| 07:25.33 | klow | anyone in here use homer / res_hep (pjsip) ? |
| 07:26.03 | klow | Trying to get it hooked up, have hep.conf and the modules loaded, but doing some sip stuff and looking for udp packets i see nothing leaving asterisk or any logging regarding it |
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| 08:22.22 | klow | nevermind got that working |
| 08:24.06 | klow | next problem ;) I'm able to register and even place calls with TLS, no certificate warnings, until after I make the first call, something is prompting both my softphones (different software) to give me a certificate warning, showing the IP address of my asterisk server, rather than the hostname which the cert uses. I'm thinking this is some asterisk setting where its defaulting to the IP in the SDP, or a Contact Header, or something? |
| 08:24.28 | klow | Its a certificate name mismatch, that happens after the initial TLS connection - and I really believe the certs/hostnames to be fine |
| 08:24.45 | klow | asterisk 14.2 pjsip , Bria and Groundwire clients |
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| 10:37.28 | tparcina | Sometimes I am experiencing problems with attendant call transfer. |
| 10:38.39 | tparcina | One user receives the phone call, answers it, and after a while tries to transfer it to another user. |
| 10:40.26 | tparcina | Phone of another user rings, he talks to the first user and when the first user hangs up the original caller and second user can't hear each other. |
| 10:41.09 | tparcina | This doesn't happen on every transfer, just on cca 10% of transfers. |
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| 10:41.49 | tparcina | I have done SIP debug, but I can't really figure out what is wrong there. |
| 10:56.36 | pawiecki | Hi. I'm adding sip trunk to my *. So far it works, but responds with 404 to OPTIONS. How can I make it respond with 200 OK? |
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| 14:03.48 | xdije | hi |
| 14:04.14 | xdije | anyone has experience with ATCOM asterisk boxes |
| 14:13.44 | wyoung | i dont |
| 14:18.57 | Samot | Nor I. |
| 14:19.38 | WIMPy | Is it a historic or a current thing? |
| 14:28.19 | Samot | I think it's the same as the NeoGates. |
| 14:28.59 | Samot | And since it supports the Skype module, I'm going t say it's running =< 1.8 |
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| 14:38.14 | Samot | xdije: What issue are you having? |
| 14:40.30 | xdije | before atcom was open to do config from command line, and used that to connect my ATCOM IP02 to GOIP gsm pbx, now with the newer version of ATCOM firmware, is not possible to do config from command, and i couldnt connect to my GOIP |
| 14:40.47 | Samot | That's an ATCOM issue. |
| 14:40.52 | Samot | You'll need to contact them. |
| 14:41.22 | Samot | Places like that do not have latest versions of Asterisk installed on them. |
| 14:41.38 | xdije | i know i just asked maybe someone went through the same issue |
| 14:42.00 | Samot | Not in here. GUI based Asterisk releases are not supported here. |
| 14:42.09 | Samot | For very specific reasons, like this.. |
| 14:42.20 | Samot | Where they are using an old version/modified version of Asterisk |
| 14:42.42 | Samot | Or they use predesigned dialplan/configs that unless you use it, you won't know what they are doing |
| 14:43.18 | Samot | But generally, these boxes were designed and made 3-4 years ago. |
| 14:43.43 | Samot | That's the problem with the NeoGates. They run 1.6.2, which they modified. |
| 14:44.05 | Samot | Since I don't use them for their PBX functions, just as FXS gateways not a big deal. |
| 14:44.38 | Samot | But I've used their PBX systems and they are not great. |
| 14:45.12 | Samot | In fact, I had to give them a fix for their own PBX dialplan. |
| 14:45.15 | xdije | as you said , if you want to use them only for fxs gateways then they are fine |
| 14:48.21 | xdije | what do you recommend as an asterisk pbx with gsm function |
| 14:48.31 | Samot | I have yet to see an "Asterisk IP PBX" come out of China running anything that wasn't from 3-5 years ago for Asterisk. |
| 14:48.40 | Samot | I can't recommend anything. |
| 14:48.49 | Samot | I don't use GSM and never had to. |
| 14:53.16 | Samot | I'm kinda happy with that too. |
| 14:54.05 | xdije | thanks for the information, and happy new year |
| 14:54.07 | Samot | Just like I've never had to deal with PRI/TDM/FXO cards in Asterisk boxes. |
| 14:54.19 | xdije | i've to leave now |
| 14:54.26 | xdije | bye |
| 14:54.29 | Samot | Bye. |
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| 15:59.35 | chris1980 | hi, I was on the other day with tls issues, ive configured chan_sip with encryption as pjsip gave me service unavailable as sood as i set media_encryption, and the phone client (csipsimple) states that srtp is working but I havnt transferred any keys, is there a way I can confirm this is truly working and encrypted? |
| 16:00.56 | Samot | You'll see encryption in the SDP. |
| 16:03.37 | chris1980 | sorry, whats the sdp? |
| 16:06.19 | chris1980 | ok got it sorry |
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| 18:02.50 | dpenev | Hi All, I am developing a DAHDI channel driver for an Analog FXO port. The driver is sworking but I have hard time to make callerid working. It works from time to time if I increase the asterisk CLI verbosity. Which suggest some timing issue probably. Bellow I have included the signalling log for my FXO driver. Can someone confirm if this is what is expected to be done? |
| 18:03.08 | dpenev | 1. Driver detects an incoming ringing (first ring) based on the voltage pattern on the DAA RING/TIP pins 2. Driver calls dahdi_hooksig(channel, DAHDI_RXSIG_RING); 3. Driver detects incoming ringing has stopped (first ring stops) 4. Driver calls dahdi_hooksig(channel, DAHDI_RXSIG_OFFHOOK); 5. Driver receives ioctl DAHDI_TONEDETECT. Driver does noting 6. Driver detects an incoming ringing (second ring) 7. Driver calls dahdi_hooksig(cha |
| 18:03.15 | dpenev | 8. DAHDI calls the driver .hooksig function with argument DAHDI_TXSIG_OFFHOOK 9. Driver puts the DAA off-hook |
| 18:05.58 | Samot | #asterisk-dev I think is the place for that. |
| 18:07.29 | dpenev | Samot: thanks! |
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| 18:33.12 | chris1980 | when srtp is enabled will packets still use udp? ive captured the interface in wireshark, the tls shows as encrypted, but the actuall call packets just show data, does this mean srtp is not functional |
| 18:35.29 | [TK]D-Fender | srtp = udp |
| 18:35.33 | [TK]D-Fender | ALL voice = UDP |
| 18:37.54 | chris1980 | i knew voice is udp, im just trying to confirm srtp is working and i thought it might be different in some way |
| 18:38.48 | chris1980 | the call doesnt show up in wireshark sip voice menu either |
| 18:41.26 | [TK]D-Fender | "wireshark sip voice menu" doesn't say any specific protocol. |
| 18:41.29 | [TK]D-Fender | SIP != voice |
| 18:41.40 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 18:41.55 | [TK]D-Fender | I think you'd better get a proper breakdown of what that menu is supposed to actually represent |
| 18:46.20 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 18:46.55 | chris1980 | sorry, im no expert, its for an assignment on voip security, which I have no experience of, ive managed to get an asterisk server running after many headaches (Sagot helped with file permissions yesterday) and presumed (clearly wrong) that the call would show up. |
| 18:47.01 | igcewieling | Javascript! <rant duration=30m" /> 8-| |
| 18:47.24 | igcewieling | chris1980: srtp and tls are tough things to make work. |
| 18:48.51 | chris1980 | i understand that now! :) |
| 18:50.22 | chris1980 | my two devices state that tls and srtp is working but i havent uploaded the crt or pem as the tutorial states |
| 18:51.34 | chris1980 | at least they connect |
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| 19:26.15 | *** join/#asterisk kb3vgw (~kb3vgw@12.144.144.173) |
| 19:26.30 | kb3vgw | ok guys how goes |
| 19:26.47 | kb3vgw | anyone here using the asterisk 1.8 on openwrt |
| 19:26.54 | igcewieling | LOL! |
| 19:27.01 | kb3vgw | ? |
| 19:27.46 | kb3vgw | been along time since I touched asterisk but there is no freeswitch for openwrt so its asterisk or yate |
| 19:27.53 | igcewieling | 1.8 was released 2010 and support was stopped 2015 |
| 19:28.39 | kb3vgw | well seems no one has updated the pkgs for opemwrt 1.8 is the last set of pkgs |
| 19:28.52 | igcewieling | I prefer Asterisk 11 because of internal legacy issues, but if you are starting out try Asterisk 13 or Asterisk 14 |
| 19:29.14 | kb3vgw | asterisk 18 it might be |
| 19:29.33 | igcewieling | kb3vgw: using it is your choice, but you'll find that almost nobody will help. |
| 19:29.35 | kb3vgw | sorry they call it asterisk 18 in pkgs |
| 19:29.53 | igcewieling | it isn't Asterisk 18 |
| 19:29.58 | drmessano | lol |
| 19:30.20 | drmessano | So you're gonna run Asterisk on a router, for like 4 concurrent calls |
| 19:30.24 | drmessano | no transcoding |
| 19:30.29 | drmessano | Get a Pi |
| 19:31.09 | kb3vgw | this is no normal router |
| 19:31.26 | drmessano | lol ok |
| 19:31.26 | kb3vgw | its a bananapi r1 arm board dual core 1 ghz |
| 19:31.40 | kb3vgw | with a gig of ram |
| 19:31.52 | drmessano | So if its a bananapi, why are you running OpenWRT? |
| 19:31.55 | kb3vgw | its just a small office 5 people |
| 19:32.09 | kb3vgw | it a r1 router board |
| 19:32.17 | kb3vgw | its ment to be a router |
| 19:32.39 | kb3vgw | and hahahah on the joke |
| 19:33.00 | drmessano | The R1 is a open hardware router that can run on a variety of open source operating systems including OpenWrt, Android, and Bananian. |
| 19:33.15 | drmessano | So use a different distro that supports a modern asterisk release |
| 19:33.19 | drmessano | Like the Bananian |
| 19:33.30 | kb3vgw | yes but I used it toreplace a crappy router |
| 19:33.44 | [TK]D-Fender | You also should be running a PBX on yuour router |
| 19:33.49 | [TK]D-Fender | that's all your eggs in one basket |
| 19:34.08 | kb3vgw | its a small office and they wanted a all in one |
| 19:34.12 | kb3vgw | so I am doing it |
| 19:34.34 | [TK]D-Fender | People aks for a lot of stupid things |
| 19:34.41 | [TK]D-Fender | Apparently including using unsupported releases |
| 19:34.55 | kb3vgw | well its a small office and they dont want to put alot in it as its a startup |
| 19:35.07 | [TK]D-Fender | Buy a 2nd computer for < $100 |
| 19:35.21 | kb3vgw | well I have to look maybe the version I chose was wrong |
| 19:35.32 | kb3vgw | brb swapping the chip |
| 19:38.52 | *** join/#asterisk kb3vgw_ (~kb3vgw@12.144.144.173) |
| 19:39.29 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 19:42.38 | kb3vgw_ | looks like I should have used asterisk 13 |
| 19:42.46 | kb3vgw_ | I wikk have to update the pkgs |
| 19:43.21 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 19:50.51 | EnrgySmth | Crazy question. I have a couple old 900Mhz wireless phone systems, spare desktops and basic linux knowledge. If I wanted to set up a phone system for my grandkids that plays recordings when they connected to the pbx... is that reasonably doable with Asterik? |
| 19:53.29 | igcewieling | EnrgySmth: not really. |
| 19:53.55 | kb3vgw_ | why they could do a dial a ext that plays a wav |
| 19:54.06 | kb3vgw_ | donr it in the past |
| 19:54.10 | kb3vgw_ | done |
| 19:54.16 | kb3vgw_ | even on FS |
| 19:54.38 | igcewieling | either the 900Mhz phones are proprietary . which case, they won't work with Asterisk OR the phones are regular analog phones, which require analog cards for Asterisk |
| 19:55.19 | igcewieling | In either case, you'll be spending hundreds of dollars for the hardware. |
| 19:55.20 | kb3vgw_ | do there is a sip to analog interface |
| 19:55.27 | kb3vgw_ | I for get the name of them |
| 19:55.30 | kb3vgw_ | pap2 |
| 19:55.41 | igcewieling | they are called ATAs. |
| 19:55.47 | kb3vgw_ | yes ata's |
| 19:56.03 | kb3vgw_ | you can get them on ebay cheap |
| 19:56.12 | igcewieling | They don't support disconnect supervision to you'll have stuck calls all the time if you use the ATA as lines into a PBX's CO port. |
| 19:56.49 | igcewieling | they should work OK for standard analog phones without an external PBX. |
| 20:03.39 | klow | Hey guys.. looking for some suggestions with a slightly weird TLS problem I am having, posted to asterisk-users but no reply yet: http://lists.digium.com/pipermail/asterisk-users/2016-December/290615.html thanks ! |
| 20:05.26 | [TK]D-Fender | "no reply yet". You posted this TODAY and we're between Christmas & New Years |
| 20:05.40 | kb3vgw_ | hey TK long time |
| 20:05.47 | kb3vgw_ | < RIch Neese |
| 20:05.55 | kb3vgw_ | been a long time ma |
| 20:06.00 | kb3vgw_ | man |
| 20:17.52 | kb3vgw_ | ok getting 13 installe don install now |
| 20:17.57 | kb3vgw_ | this brings it up |
| 20:27.55 | kb3vgw_ | also having asterisk on a emergancy router for field deploment is not bad |
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| 20:28.17 | klow | [TK]D-Fender - not a complaint. Just trying to get this sorted. Unfortunately I'm under a crunch, that necessitated me switching to asterisk 14 and pjsip and less time for testing than I would like, no holiday downtime for me ;( sometimes asterisk-users posts sometimes get no replies, I posted before bed hoping i'd be lucky since lots of community seems to be in europe |
| 20:29.04 | [TK]D-Fender | You didn't post any debug, configs, or anything useful at all |
| 20:29.27 | [TK]D-Fender | That wasn't even qualified as an attempt at properly asking for help |
| 20:36.03 | *** join/#asterisk polysics (~polysics@75.144.233.245) |
| 20:36.42 | polysics | hey #asterisk? Still have a strict "no FreePBX questions" policy here? :D |
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| 20:47.50 | drmessano | polysics: There's an actual #freepbx channel |
| 20:48.02 | drmessano | Been there for 12 years |
| 20:48.35 | polysics | I know, but there is more people here :) |
| 20:48.57 | drmessano | polysics: 100% of the people in #FreePBX are there for FreePBX |
| 20:49.10 | drmessano | The percentage here is less |
| 20:49.21 | polysics | kay, you'll notice I did not even ask the actual question :) |
| 20:49.32 | drmessano | Did you even ask in #FreePBX? |
| 20:50.23 | polysics | yes, I did |
| 20:57.19 | [TK]D-Fender | And being answered there |
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| 21:46.45 | kb3vgw_ | [TK]D-Fender, my old nik is UnixDog back when I worked on astersk years ago |
| 21:47.03 | kb3vgw_ | but long time no see man hope things have been well for you |
| 22:07.45 | *** join/#asterisk EnrgySmth_ (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 22:13.51 | EnrgySmth_ | Does anyone what would be needed to interface an analog phone (900Mhz) to Rasberry PI Asterisk? |
| 22:14.28 | EnrgySmth_ | in a regular pc would just add a card |
| 22:14.43 | EnrgySmth_ | is there maybe a USB version of something like that? |
| 22:15.30 | EnrgySmth_ | just brainstorming on a way to tinker with Asterisk on stuff I mostly have laying around? |
| 22:15.36 | igcewieling | look at Xorcom. |
| 22:15.46 | EnrgySmth_ | thanks igcewieling |
| 22:15.51 | igcewieling | if you want analog interfaces, then it is going to cost you. |
| 22:16.38 | EnrgySmth_ | maybe cheaper to get wifi headset then |
| 22:19.33 | kb3vgw_ | he needs a ata for each device |
| 22:19.53 | kb3vgw_ | or there isa multi port 1 Just have to find it |
| 22:20.00 | drmessano | Buy a Cisco SPA112 |
| 22:20.11 | igcewieling | there are many ATAs with 2 analog FXS ports |
| 22:20.15 | drmessano | $40 ATA |
| 22:21.28 | *** join/#asterisk pawiecki (~pawiecki@user-94-254-167-54.play-internet.pl) |
| 22:24.04 | kb3vgw_ | http://www.ebay.com/itm/VOIP-Phone-Adapter-Gateway-2-Line-Lan-Wan-ATA-Compare-to-Linksys-SP2-With-VPN-/111927180257?hash=item1a0f613be1:g:7vEAAOSwyjBW2436 |
| 22:24.10 | kb3vgw_ | a device like this |
| 22:24.25 | EnrgySmth_ | wow that is perfect - thank you |
| 22:24.52 | EnrgySmth_ | i would only have a single handset in this case - just want to be able to interact with the pbx menus, etc |
| 22:26.40 | Samot | I've never even heard of Chima. |
| 22:27.43 | kb3vgw_ | there is one from InnoMedia for 10$ |
| 22:27.43 | Samot | Hah, maybe that's why. It comes with a 210 power unit. |
| 22:28.12 | Samot | 5:19:56 PMÂ <drmessano>Â Buy a Cisco SPA112 |
| 22:28.13 | EnrgySmth_ | have a model or something kb3vgw_ ? |
| 22:28.14 | Samot | ^ That. |
| 22:28.31 | Samot | They are inexpensive and will last through pretty much anything. |
| 22:28.37 | Samot | And widely supported. |
| 22:28.44 | EnrgySmth_ | ok thanks guys |
| 22:29.09 | drmessano | Stay away from random Chinese crap |
| 22:29.17 | drmessano | Unless YOU can support it |
| 22:29.19 | EnrgySmth_ | really appreciate all the advice. Cisco seems like a safe way to go |
| 22:29.20 | Samot | ^ ++++++ |
| 22:30.07 | drmessano | If you're not an expert, buy what others can help you with.. Cisco SPA112 is a great choice. |
| 22:31.03 | Samot | drmessano: You're not going to believe what happened today. |
| 22:32.23 | *** join/#asterisk kb3vgw (~kb3vgw@12.144.144.173) |
| 22:32.25 | Samot | A n00b in #kamailio actually came to the conclusion that it was out of his skill set and he should hire some one/company for the service instead of hacking his way through a production system. |
| 22:32.32 | Samot | All on his own, even. |
| 22:32.35 | kb3vgw | sorry power went off |
| 22:32.45 | drmessano | Must be Christmas, Samir |
| 22:32.49 | drmessano | Samot |
| 22:32.52 | kb3vgw | 'I need to get the battery for the r1 board |
| 22:33.09 | Samot | I complimented him. |
| 22:33.39 | EnrgySmth_ | nite guys - thanks again for the advice! |
| 22:34.07 | kb3vgw | http://www.ebay.com/itm/ATA-Phone-InnoMedia-VOIP-MTA-XT-MTA6328-2Re-/182309734592?hash=item2a7281c8c0:g:xYMAAOSwnLdWtJRL |
| 22:34.26 | Samot | Unlike the n00b in the mik room later arguing about VRRP when he was asking for help with it. |
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| 22:34.42 | kb3vgw | Grandstream makes a multi port unit also |
| 22:34.47 | drmessano | That's a great option, if HE can set it up. I doubt he'll find any documentation on it |
| 22:35.01 | drmessano | Better to recommend something SAFE |
| 22:35.26 | Samot | Grandstream, Cisco.. |
| 22:35.36 | drmessano | SPA112 is $40 |
| 22:35.40 | Samot | These are probably the best bet for an ATA. |
| 22:35.44 | Samot | Most used and supported. |
| 22:36.01 | kb3vgw | http://www.ebay.com/itm/InnoMedia-MTA6328-2Re/182227466317?_trksid=p2047675.c100005.m1851&_trkparms=aid%3D222007%26algo%3DSIC.MBE%26ao%3D2%26asc%3D38530%26meid%3D92d7e1ac670b400196f88db8d5294f91%26pid%3D100005%26rk%3D3%26rkt%3D6%26sd%3D182309734592 |
| 22:36.02 | Samot | I would go with Cisco first. But I personally don't like Grandstreams. |
| 22:36.24 | kb3vgw | cisco ata seem not to have alot on ebay |
| 22:36.32 | kb3vgw | and its cisco/linksys now |
| 22:36.39 | drmessano | No it's not |
| 22:36.47 | Samot | Cisco bought Linksys like a decade ago. |
| 22:37.04 | drmessano | Belkin bought Linksys |
| 22:37.15 | drmessano | The SPA 112 is branded Cisco |
| 22:37.17 | Samot | Well Cisco bought the SPA stuff. |
| 22:37.23 | Samot | Or something. |
| 22:37.31 | Samot | All the SPA lines were rebranded as Cisco. |
| 22:37.37 | Samot | IP phones and ATAs. |
| 22:37.45 | drmessano | There is no more Linksys phone stuff |
| 22:37.53 | Samot | Right. |
| 22:38.15 | drmessano | I wouldn't buy an SPA 112 off of eBay anyway.. anything on there's likely to be counterfeit |
| 22:38.21 | drmessano | Amazon |
| 22:38.22 | Samot | I think the only thing that was truly rebrand without the model changing was the SPA8000. |
| 22:38.24 | kb3vgw | http://www.ebay.com/itm/ATA-SIP-Cisco-Linksys-RTP300-NA-VoIP-Router-NEW-in-never-opened-box-/252660929943?hash=item3ad3c3d597:g:FPcAAOSwYIxX7L56 |
| 22:38.45 | Samot | EOL device. |
| 22:38.46 | drmessano | RTP300 is a waste. |
| 22:38.52 | Samot | If it has Linksys on it, it's dead. |
| 22:38.59 | drmessano | Exactly |
| 22:39.04 | kb3vgw | the go with this |
| 22:39.09 | Samot | SPA112 |
| 22:39.17 | Samot | We are giving you the exact model. |
| 22:40.05 | Samot | You can spend $15 on a non-widely used/supported ATA and then spend hours trying to make it work.. |
| 22:40.25 | drmessano | I don't understand why the fuck you would recommend EOL and unGoogle-able devices to a newb |
| 22:40.32 | Samot | Or you can spend $30-$40 on a SPA112 and have a plethora of support and information at your finger tips. |
| 22:40.38 | [TK]D-Fender | And end up with an inferior product when you find that model X has no standard dialplan syntaxt support, no T.38 as an option, etc |
| 22:40.50 | [TK]D-Fender | just more hassle than being a cheapskate is worth |
| 22:41.09 | [TK]D-Fender | Spend a few bucks on something that isn't outdated, or a piece of crap |
| 22:41.10 | drmessano | $40! |
| 22:41.13 | [TK]D-Fender | and your TIME |
| 22:41.14 | Samot | People never really think about mime = money... |
| 22:41.15 | drmessano | SPA112 |
| 22:41.19 | drmessano | AMAZON |
| 22:41.19 | Samot | Er..time.. |
| 22:41.21 | kb3vgw | http://www.ebay.com/itm/CISCO-SYSTEMS-ATA-186-188-ANALOG-TELEPHONE-ADAPTOR-/172460556029?hash=item2827733efd:g:yboAAOSwEzxYYXvI |
| 22:41.21 | drmessano | DONE |
| 22:41.26 | drmessano | HAHA |
| 22:41.29 | Samot | I was channeling Spinal Tap for a second. |
| 22:41.35 | Samot | Oh my gawd. |
| 22:41.40 | [TK]D-Fender | mime = money, but it's a tought field to break into. |
| 22:41.40 | drmessano | ATA 186 is 15 years old |
| 22:41.45 | Samot | Please stop pasting random Ebay links. |
| 22:41.47 | drmessano | WTAF |
| 22:41.55 | Samot | We're not a consumer buyers review board. |
| 22:42.03 | kb3vgw | just pointing him to low cost uts |
| 22:42.18 | drmessano | THAT HE WONT BE ABLE TO SET UP |
| 22:42.23 | [TK]D-Fender | 15 year old shit is NOT what you should be aiming for. |
| 22:42.31 | Samot | Stop recommending outdated and unsupported devices. |
| 22:42.35 | Samot | It's horrible advice. |
| 22:42.38 | drmessano | THAT NO ONE WILL HELP HIM SUPPORT |
| 22:42.42 | drmessano | JUST STOP |
| 22:42.44 | [TK]D-Fender | Apparently the years haven't taught you much... |
| 22:42.55 | drmessano | SRSLY |
| 22:43.13 | Samot | This is like asking a 25 yo computer tech to look at an old 8080.. |
| 22:43.17 | Samot | They'll just stare at it. |
| 22:43.26 | kb3vgw | we still use them we have 150 of them and they work for what we need for basic pickup phones |
| 22:43.43 | Samot | So you're going to support the random IRC stranger? |
| 22:43.48 | drmessano | Next he'll be suggesting he get a soldering iron and some sound cards |
| 22:43.58 | kb3vgw | no |
| 22:44.03 | Samot | No, tonsofpcs will do that. |
| 22:44.18 | Samot | Then why are you recommending things he won't be able to get support for? |
| 22:44.47 | Samot | "I know how to use this old, archaic device. Use it" |
| 22:44.48 | drmessano | If you want him to get some old shit off of eBay then give them your home phone number |
| 22:45.06 | drmessano | Because you were not planning on helping him set it up there's really no point in him getting something he can't work with |
| 22:45.21 | drmessano | And nobody else on IRC is going to help him with |
| 22:45.24 | kb3vgw | just found a digium a50 for 249 |
| 22:46.09 | Samot | 5:38:46 PM â EnrgySmth quit |
| 22:46.14 | Samot | ^ You can stop looking. |
| 22:46.19 | Samot | He's not even here anymore. |
| 22:46.51 | Samot | So not only are you making bad suggestions, you're making them to no one. |
| 22:47.10 | Samot | It's like watching Clint Eastwood at the RNC event. |
| 22:48.02 | kb3vgw | I am configuring *13 on this router setup for 7 phones now the gui looks like they used the digium gui and pulled it apart |
| 22:48.12 | kb3vgw | but its nice to have |
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| 23:49.28 | klow | If you select codec_opus in menuselect, compilke, make, make install, you should see the opus module loaded, have opus command in asterisk cli, and see something in 'core show translation' aye? |
| 23:52.15 | kb3vgw_ | you have to enable the module |
| 23:52.23 | kb3vgw_ | reload aterisk |
| 23:52.30 | kb3vgw_ | and it should show up |
| 23:57.30 | klow | hmm, ya I'm not ending up with a codec_opus shared lib after compile |
| 23:57.37 | klow | i do have it selected in menuselect .. |
| 23:57.41 | klow | i think . |
| 23:57.52 | klow | when i run "make menuselect" again, it is ticked on |
| 23:58.46 | [TK]D-Fender | So you SEE the magic .so compiled? |