| 00:53.10 | *** join/#asterisk Bhakimi (~textual@rrcs-69-75-121-202.west.biz.rr.com) |
| 01:19.59 | Samot | 65 polycoms, even those models, for $200 means that they either didnt know the value..or something like perhaps they are fscked |
| 01:20.17 | Samot | Hopefully it is the forementioned. |
| 02:46.33 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 03:47.34 | *** join/#asterisk roukoswarf (znc@rouk.org) |
| 04:06.35 | *** join/#asterisk puzzled (~puzzled@2001:982:1097:1::1:3) |
| 04:24.23 | *** join/#asterisk kannan (~chatzilla@182.65.90.227) |
| 05:02.29 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
| 05:02.34 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 05:02.53 | *** join/#asterisk smoke (~smoke@smoke.rokyta.cz) |
| 05:03.34 | *** join/#asterisk DanQuinney (sid18169@gateway/web/irccloud.com/x-wgmydrqqyrxonzyx) |
| 05:03.49 | *** join/#asterisk craysiii (sid205503@gateway/web/irccloud.com/x-qklcldrdelsivwof) |
| 05:05.07 | *** join/#asterisk mrhelpmann (~mrhelpman@i.am.mrhelpmann.xyz) |
| 05:05.29 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 05:10.22 | *** join/#asterisk Haris (~haris@unaffiliated/haris) |
| 05:10.24 | Haris | hello all |
| 05:10.27 | Haris | guys, anyone around ? |
| 05:10.30 | Haris | at this hour |
| 05:11.36 | Samot | Of course. |
| 05:13.01 | Haris | I have a telco, that's connected to me via registry, rather than peer |
| 05:13.13 | Haris | in sip show channels output, I can't see cli under user/ANR field |
| 05:13.25 | Haris | for inbound calls. but cli is showing at agent end |
| 05:13.48 | Haris | Is this my config/fault or the telco isn't sending me cli for incoming calls ? |
| 05:14.19 | Haris | another telco is connected via peer method. incoming calls coming from them are showing cli under user/ANR column in sip show channels output |
| 05:14.42 | Samot | Show an actual debug. |
| 05:15.33 | Samot | The device does't magically show CLID. |
| 05:15.44 | Samot | So it would seem they are sending those details. |
| 05:16.26 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 05:16.52 | Haris | that's why I'm confused. for one telco, its showing. for the other, its not |
| 05:17.33 | Samot | OK. |
| 05:17.39 | Samot | So this is only in sip show channels? |
| 05:17.56 | Samot | The CDRs, the CELs, they all show the CLID? |
| 05:19.20 | Samot | If the CLID is showing up at the agent's device but not in the User/ANR field in sip show channels, then the telco is passing CLID. |
| 05:19.35 | Samot | It just doesn't magically show up on the device without it being sent to the device. |
| 05:24.10 | Haris | ok. so why is it not showing in sip show channels output ? |
| 05:24.21 | Samot | Will you show a debug?! |
| 05:24.27 | Haris | making it ready for paste |
| 05:24.28 | Samot | And answer the other questions? |
| 05:24.35 | Samot | Don't f'ing EDIT IT |
| 05:24.37 | Haris | removing the unnecessary part |
| 05:24.44 | Haris | its 2500 lines |
| 05:24.51 | Haris | cutting it down to under 1000 |
| 05:25.05 | Samot | How can a single call be 2500 lines? |
| 05:25.41 | Haris | its not a single call. its normal "muliple calls going on" "activity" during business hours |
| 05:29.48 | Haris | ok. its 1500 lines |
| 05:30.13 | Haris | https://pastebin.ca/3858107 |
| 05:30.53 | Haris | the call from 125.209.93.196 |
| 05:31.18 | Haris | it shows the username for my sip registry in the user/ANR field, rather than the caller number |
| 05:31.29 | Haris | caller's+ no |
| 05:31.45 | Haris | the incoming call from 03400007510 |
| 05:31.53 | Samot | Wait.. |
| 05:32.01 | Samot | So it's not that nothing isn't there. |
| 05:32.09 | Haris | ? |
| 05:32.13 | Samot | It's just the sip register username instead of the number? |
| 05:32.14 | Haris | ofcourse |
| 05:32.18 | Haris | yes |
| 05:32.22 | Samot | Of course it's going to be the username |
| 05:32.28 | Haris | hmm |
| 05:32.31 | Samot | That's where the call is actually being sent. |
| 05:32.39 | Haris | its an inbound call |
| 05:32.44 | Samot | Right. |
| 05:32.52 | Samot | Inbound calls point to the your SIP register |
| 05:32.59 | Samot | That's the point of registering. |
| 05:33.08 | Haris | hmm |
| 05:33.08 | Samot | To tell the other side "This is my location to send stuff to" |
| 05:33.25 | Samot | Why does it matter if it is in this location? |
| 05:33.26 | Haris | is this different in peer connectivity ? |
| 05:33.34 | Samot | The caller Id is correct in the CDRs and CELS, right? |
| 05:33.38 | Haris | yes |
| 05:33.42 | Samot | OK |
| 05:33.45 | Samot | So what's the issue? |
| 05:33.48 | Samot | And yes, it is. |
| 05:33.52 | Samot | Because there's no user |
| 05:34.03 | Samot | They are routing the number directly to you |
| 05:34.17 | Samot | That is what is put in the user part of the request/to URIs |
| 05:34.31 | Haris | I can't see no for inbound calls, on cli, in ship show channels output |
| 05:34.41 | Samot | So? |
| 05:34.42 | Haris | %s/ship/sip/ |
| 05:34.45 | Samot | Why are you looking there? |
| 05:35.00 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 05:35.08 | Haris | why not |
| 05:35.10 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:35.18 | Samot | It's more a debug tool |
| 05:35.25 | Samot | Then a reporting tool. |
| 05:35.29 | Samot | It shows SIP dialogs. |
| 05:35.31 | Samot | ALL of them |
| 05:35.33 | Samot | Not just invites. |
| 05:35.43 | Samot | Notify, ACK, REGISTER, etc. |
| 05:36.46 | Samot | It also holds the LAST message.. |
| 05:36.50 | Samot | Not active messages. |
| 05:36.55 | Haris | for the other telco, which is connected to me, via peer method, for them, I can no for in/out calls in sip show channels output |
| 05:37.07 | Samot | I just explained why |
| 05:37.24 | Haris | I thought, that was for peer via registry method |
| 05:37.27 | Samot | Because the DIDs are routed to the IP of the PBX directly |
| 05:37.28 | Haris | peers+ |
| 05:37.47 | Haris | ok, so if they changed method of connectivity, this would also be that way ? |
| 05:37.52 | Samot | Yes. |
| 05:38.02 | Samot | Because that's how SIP Registers work. |
| 05:38.06 | Haris | this is the behaviour for registry vs peer method of connectivity |
| 05:38.11 | [TK]D-Fender | huh? |
| 05:38.25 | Samot | Telco A = He registers |
| 05:38.31 | Samot | Telco B = IP auth |
| 05:38.34 | [TK]D-Fender | You need a peer regardless |
| 05:38.40 | Samot | Not that. |
| 05:38.52 | Haris | That is what I'v been telling them for a good 4 days now |
| 05:38.55 | Samot | He's complaining that in "sip show channels" |
| 05:38.58 | Haris | change my connectivity from registry to peer |
| 05:39.01 | [TK]D-Fender | whether they need to to register doesn't change needing a peer to match the calls |
| 05:39.14 | Samot | With the REGISTERED trunk the User/ANR is showing the SIP user..not the DID |
| 05:39.30 | Samot | This has nothing to do with the actual registration. |
| 05:39.37 | Haris | hmm |
| 05:39.44 | Samot | It's how "sip show channels" is displaying the information for him. |
| 05:39.57 | Samot | Because they are sending the calls to his SIP user registeration.. |
| 05:39.59 | [TK]D-Fender | I don't recalls seeing actual debug |
| 05:40.10 | Samot | He pasted it |
| 05:40.13 | *** join/#asterisk boris_t (~boris_t@82.112.41.207) |
| 05:40.17 | Haris | https://pastebin.ca/3858107 |
| 05:40.41 | Samot | And it's pointless. |
| 05:40.54 | Samot | Because it has nothing to do with what he is asking about. |
| 05:40.57 | Samot | At all. |
| 05:41.12 | Haris | I'm still confused |
| 05:41.19 | Samot | Peer User/ANR Call ID Format Hold Last Message Expiry Peer |
| 05:41.19 | Samot | 204.132.107.230 (None) fdbaa869-3e2c0f (nothing) No Rx: REGISTER <guest> |
| 05:41.28 | Samot | ^^^ THIS |
| 05:41.46 | Samot | You want to know what the User/ANR is showing as the SIP user for your Teclo Trunk that uses REGISTRATION |
| 05:41.54 | Samot | That's because the calls are routed to the USER |
| 05:42.07 | Samot | The other Telco in which the calls are routing to the PBX directly... |
| 05:42.11 | Samot | They will show the NUMBER |
| 05:42.18 | Samot | Because that's where the calls are SENT |
| 05:42.44 | Samot | The User/ANR is pulled from the Request-URI |
| 05:43.06 | Haris | https://pastebin.ca/3858111 |
| 05:43.07 | Samot | When a call is routed to a SIP account, the Request-URI is <username>@<location>:<port> |
| 05:43.16 | Haris | ah |
| 05:43.44 | Samot | When a call is routed to a IP the Request-URI is <number>@<location>:<port> |
| 05:43.44 | Haris | so your saying that's how it works, and then saying its not how that works |
| 05:44.18 | Samot | The User/ANR value in "sip show channels" is the USER part of the Request-URI |
| 05:44.22 | Haris | see the username in the user/anr field. lol |
| 05:44.27 | Samot | Whatever that USER is, is what is there. |
| 05:44.52 | Haris | so, it actually does matter if its peer or registry based connectivity |
| 05:44.56 | Samot | Because the Teclo that you REGISTER to is setting the USER of the Request-URI to your USERNAME |
| 05:45.07 | Haris | to get asterisk to get that number, I need to change that method of connectivy from isp/telco |
| 05:45.17 | Samot | Why?! |
| 05:45.21 | Samot | Why are you using this? |
| 05:45.25 | Samot | It's a debug tool |
| 05:45.25 | Haris | ? |
| 05:45.28 | Samot | Not a reporting tool |
| 05:45.34 | Samot | "sip show channels" |
| 05:45.46 | Samot | You're making something out of completely nothing. |
| 05:45.48 | Haris | because I'm sitting at the server. I can't go to each agent's seat and see what calls are going on |
| 05:46.18 | Haris | I need to see what calls are going on at server level |
| 05:46.18 | Samot | "sip show channels" shows ALL SIP dialogs. |
| 05:46.23 | Samot | Not just calls. |
| 05:46.29 | Samot | So using it to monitor calls is dumb. |
| 05:47.19 | Haris | well .. it does show active calls, at any given time, not withstanding the excess in a recurring way |
| 05:47.38 | Samot | Dude, it shows the last message. |
| 05:47.49 | Haris | hmm |
| 05:47.51 | Samot | An ACK is either an INVITE being ACK'd. |
| 05:47.55 | Samot | Or it's a CANCEL being ACK'd. |
| 05:48.00 | Haris | let me see the web plugin I have with fpbx. perhaps it will also get it |
| 05:48.00 | Samot | Which one? Don't know. |
| 05:48.07 | Samot | It's a cache. |
| 05:48.20 | Samot | Dude, install FOP2 |
| 05:48.21 | Haris | hmm |
| 05:48.28 | Samot | It's exactly what it is made for. |
| 05:48.34 | Samot | You can monitor and manage calls |
| 05:48.37 | Samot | See all the extensions |
| 05:48.39 | Samot | Queues |
| 05:48.41 | Samot | Parking Lot |
| 05:49.02 | Haris | g00gl3s fop2 |
| 05:50.47 | Haris | fop 2.31 or fop mgr ? |
| 05:50.49 | Haris | or both ? |
| 05:50.57 | Samot | It's all the same. |
| 05:51.01 | Samot | As of 2.x |
| 05:51.33 | Haris | ok. getting the main FOP Version 2.31.13 |
| 05:52.31 | Samot | You can add a monitor button for the SIP trunks. |
| 05:52.39 | Samot | And it will show all the calls on those trunks.. |
| 05:53.03 | Samot | s/button/widget/ |
| 05:55.26 | *** join/#asterisk KValchev (~KValchev@ns.atsoftconsult-bg.com) |
| 05:55.40 | *** join/#asterisk jkroon (~jkroon@uls-154-73-35-201.wall.uls.co.za) |
| 06:16.10 | *** join/#asterisk [\\\] (~triplesla@unaffiliated/imsaguy) |
| 06:18.08 | Haris | is it safe to have fop2 change extensions_override_freepbx.conf to extensions_override_fop2.conf ? |
| 06:21.52 | *** join/#asterisk DanB (~DanB@clt-195.192.201.141.ip-anschluss.net) |
| 06:27.27 | kannan | in asterisk 11.20-vici (a version for Vicidial, an asterisk configurator), i want to react to security events and ban certain remote IPs. Fail2ban isn't working well though i tried it. I am thinking of parsing the security-filename (setup from logger.conf) but don't know how to optimally move the file pointer when parsing it. I am therefore considering patching logger.c to write the... |
| 06:27.29 | kannan | ...offending IP to a file when the logged line is in buffer, which can then be handled at leisure by a cron job. Is there any other better or obvious way that I am ignorant about? |
| 06:27.31 | drmessano | Haris: Safe? What about the ramifications of changing the file to something that's not #included anywhere |
| 06:28.19 | Samot | Sigh. |
| 06:28.26 | Samot | How about reading the documents. |
| 06:28.43 | Samot | extensions_override_fop2.conf is written out to /etc/asterisk |
| 06:29.01 | Samot | And then it is included in extensions_override_freepbx.conf |
| 06:29.06 | Samot | It doesn't "overwrite" it |
| 06:29.11 | Haris | *whew* ok |
| 06:29.11 | drmessano | lol |
| 06:29.15 | Samot | It appends an include statement. |
| 06:29.16 | Samot | FFS. |
| 06:29.18 | drmessano | Problem solved |
| 06:29.19 | Haris | I did read the README |
| 06:29.19 | Samot | READ DOCUMENTS |
| 06:29.33 | Samot | There's an entire install wiki |
| 06:29.59 | Samot | The guys over at AsterNIC know what they are doing. |
| 06:30.35 | Haris | I already have asternic |
| 06:30.41 | Haris | client side of it that is |
| 06:30.49 | Samot | AsterNIC is a company. |
| 06:30.53 | drmessano | lol |
| 06:30.54 | Haris | I know that |
| 06:30.56 | Samot | They have software products. |
| 06:30.57 | Haris | I mean its plugin |
| 06:31.04 | drmessano | No |
| 06:31.06 | Samot | ? |
| 06:31.10 | Samot | No. |
| 06:31.27 | Haris | I have the realtime tab opened for asternic's plugin |
| 06:31.28 | drmessano | stahp |
| 06:31.42 | Samot | AsterNIC CDRs is a third party module for FreePBX |
| 06:31.47 | Haris | but it doesn't show the number for which call is going on |
| 06:31.50 | Haris | hmm |
| 06:32.09 | Samot | What are you talking about? |
| 06:32.11 | drmessano | AsterNIC is a company |
| 06:32.14 | drmessano | they have applications |
| 06:32.24 | Samot | FOP2 is one of those applications. |
| 06:32.26 | drmessano | You don't have "an AsterNIC" |
| 06:32.27 | Haris | queue stats plugin |
| 06:32.30 | drmessano | FFS |
| 06:32.34 | Samot | That's an application |
| 06:32.48 | Samot | Queue Stats is designed for multiple PBX systems. |
| 06:32.59 | Haris | I know that part |
| 06:33.00 | Samot | It just happens to include plain Asterisk and FreePBX |
| 06:33.23 | Samot | These are third party applications you are installing. |
| 06:33.27 | Samot | That have FreePBX support. |
| 06:33.59 | drmessano | He clearly knows all this |
| 06:34.00 | Haris | asternic's queue stats is already installed |
| 06:34.07 | drmessano | +/- All |
| 06:34.09 | Samot | OK |
| 06:34.11 | Haris | I'm checking out fop2 on another box |
| 06:34.17 | Samot | OK |
| 06:34.26 | drmessano | I like Tacos |
| 06:35.25 | Samot | I like long slow deep wet kisses that last three days |
| 06:35.34 | Samot | BOOM |
| 06:35.38 | Samot | Bull Durham quote. |
| 06:36.29 | drmessano | I love a girl that can appreciate a mouth full of warm nuts |
| 06:36.45 | drmessano | Speaking of peanuts at the ballpark, of course |
| 06:37.04 | drmessano | There is no better treat |
| 06:37.16 | Samot | You guys. You lollygag the ball around the infield. You lollygag your way down to first. You lollygag in and out of the dugout. You know what that makes you? Larry! |
| 06:37.45 | Samot | "Lollygaggers!" |
| 06:37.50 | Samot | Lollygaggers |
| 06:38.11 | drmessano | I actually had refused to watch that movie for 20+ years |
| 06:38.37 | drmessano | I was really big into sports cards in the mid-80s to mid-90s |
| 06:38.40 | Samot | It's so damn good. |
| 06:38.53 | Samot | "Well, Nuke's scared because his eyelids are jammed and his old man's here. We need a live... is it a live rooster?" |
| 06:38.56 | drmessano | and one of the companies made some Bull Durham cards |
| 06:39.07 | drmessano | and included them as Limited Edition in random packs |
| 06:39.35 | drmessano | and I kinda felt like "What the fuck is a Bull Durham and why is this marketing shit being shoved down my throat" |
| 06:39.40 | drmessano | and I hated the movie |
| 06:39.42 | drmessano | Unseen |
| 06:40.03 | Samot | And when you finally watched it? |
| 06:40.40 | drmessano | It's not bad. But now it's kinda like.. |
| 06:40.42 | Haris | for some reason the web front end can't connect or use the backend fop2 service |
| 06:40.50 | drmessano | I love Ghostbusters.. because i've been watching it all my life |
| 06:41.07 | drmessano | But had I first watched it 2014.. Maybe not so much |
| 06:41.31 | Haris | worthless pathetic crap ghostbusters |
| 06:41.38 | drmessano | lol |
| 06:41.42 | drmessano | Yeah righto |
| 06:41.45 | Samot | Did you actually start the fop2 service? |
| 06:41.58 | drmessano | Ghostbusters is worthless pathetic crap? |
| 06:42.02 | Samot | Like it's not that hard to install. |
| 06:42.10 | drmessano | You have to be fucking kidding me |
| 06:42.18 | Samot | You do have to edit the configuration |
| 06:42.21 | Haris | many times |
| 06:42.24 | Samot | Give it an AMI user account. |
| 06:42.25 | Haris | does it need to be enabled ? |
| 06:42.29 | Haris | ah |
| 06:42.32 | Haris | my bad |
| 06:42.39 | Samot | Well you need to go and read the fscking install documents. |
| 06:42.44 | drmessano | Yeah it needs to be turned on |
| 06:42.46 | drmessano | JFC |
| 06:42.47 | Samot | That tell you what files to edit |
| 06:42.54 | Samot | What you need to do in FreePBX |
| 06:42.56 | drmessano | They actually have an install doc |
| 06:43.02 | Samot | And how to start the damn thing. |
| 06:43.10 | drmessano | I know that's against your religion to read shit, but still |
| 06:43.18 | drmessano | Might try it sometime |
| 06:43.40 | Samot | drmessano: You have Netflix? |
| 06:43.46 | drmessano | I do |
| 06:44.01 | Samot | Is "The Norsemen" an option for you? |
| 06:45.09 | drmessano | Im checking now |
| 06:45.15 | Samot | Sorry, just "Norsemen" |
| 06:46.26 | Samot | https://www.youtube.com/watch?v=6hk_rdfSCS0 |
| 06:46.39 | drmessano | Yes it is |
| 06:46.50 | Samot | It is well worth the watch. |
| 06:47.34 | drmessano | I'll add it to my list |
| 06:47.39 | drmessano | Looks good |
| 06:49.21 | Samot | Dead pan delivery.. |
| 06:52.20 | Haris | its not against my religion to read docs |
| 06:52.36 | Haris | its IN my religion to read them |
| 06:52.46 | Samot | Then you should have. |
| 06:52.48 | Haris | we gave 'boobs' to the world |
| 06:52.53 | Haris | books+ |
| 06:52.59 | Samot | You would have seen the steps needed to make this work. |
| 06:53.02 | Haris | paper+ books+ |
| 06:53.03 | Samot | Yeah. |
| 06:53.09 | Haris | my bad |
| 06:53.11 | Samot | You also gave us the Holocaust. |
| 06:53.17 | Samot | So.. |
| 06:53.33 | Haris | that holocaust debate needs to be revisited |
| 06:53.40 | Samot | So you invited books |
| 06:53.48 | Samot | Then decided they were bad and burned them. |
| 06:53.51 | Haris | invented+ |
| 06:53.53 | Samot | Full circle. |
| 06:53.59 | Haris | nope. that wasn't us |
| 06:54.21 | Haris | that were human beings who did not believe in our religion. but this is a very lengthy educational discussion, not for this time |
| 06:54.28 | Haris | %s/that/those/ |
| 06:55.14 | Haris | hitler was not part of us. neither our ally or someone we had any agreement(s) with |
| 06:55.38 | drmessano | How did we go from you not reading the FOP2 docs to hitler? |
| 06:55.42 | drmessano | Deflect much? |
| 06:55.42 | Haris | the holocaust as I see it was mostly because germans of the europe at the time, saw the cut throat worthless interest based economic system that jews of the time had instigated |
| 06:55.50 | Haris | my bad |
| 06:55.57 | Haris | going back to docs |
| 06:57.49 | Haris | I'm surprised. for intelligent, educated, docs reading human beings, your knowledge of other religion(s) come from cartoon(s), movies, rather than reading their actual docs/scripts |
| 06:57.56 | Haris | see that alot |
| 06:58.11 | Samot | I have no idea what you are talking about. |
| 06:58.23 | Samot | You actually haven't specified a religion. |
| 06:58.30 | drmessano | How do you know exactly what *I* know? |
| 06:59.00 | Samot | We have no clue what you mean by "our religion" |
| 06:59.03 | Samot | Who's? |
| 06:59.54 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:00.51 | Haris | well, I'd like to defer this, in favour of the work I'm doing. we can have that one later |
| 07:01.00 | Haris | for right now |
| 07:01.02 | drmessano | In other words |
| 07:01.06 | Haris | found the config file |
| 07:01.12 | drmessano | You dont want to piss anyone off so you can continue to get help |
| 07:01.20 | drmessano | What a fucking asshole |
| 07:01.29 | Haris | I'm a Muslim |
| 07:01.32 | Haris | my religion is Islam |
| 07:01.44 | Haris | I'm not sure which one you were talking about. actually one was never specified |
| 07:01.48 | Haris | but since it was pointed at me .. |
| 07:01.52 | Haris | I .. improvised |
| 07:02.07 | Samot | Well you claimed we knew nothing of "other religions" |
| 07:02.09 | Haris | the world is not you alone. the world wouldn't stop functioning without .. you |
| 07:02.17 | Samot | Except there was no reference point to what you meant by "other" |
| 07:02.31 | Haris | other = encompases all |
| 07:02.35 | Haris | * wildcard |
| 07:02.37 | Samot | No. |
| 07:03.06 | drmessano | You realize that "against your religion" was a *figure of speech*, right? |
| 07:03.09 | Haris | ok. I don't know how to set the AM MGR credentials |
| 07:03.21 | Haris | I may not have |
| 07:03.22 | drmessano | Since you're so intelligent and all, I would have figured that you would have guessed that |
| 07:03.24 | Samot | That would be in the FreePBX Wiki. |
| 07:03.28 | Haris | Its hard to grasp things |
| 07:03.31 | Haris | in first go |
| 07:03.34 | Samot | We can tell. |
| 07:03.35 | drmessano | For you, yes |
| 07:03.44 | *** join/#asterisk tsia (~tsia@port-4794.pppoe.wtnet.de) |
| 07:03.52 | drmessano | You've spent 2 years making 3 months progress |
| 07:04.07 | Samot | Not really. |
| 07:04.12 | Samot | He gave up on WebRTC. |
| 07:04.21 | drmessano | Ok, so 2 months |
| 07:04.24 | Haris | we don't have to go back to that one, right now |
| 07:04.32 | drmessano | Of course not |
| 07:04.40 | drmessano | Because again, you need help |
| 07:04.47 | drmessano | and don't want to piss anyone off |
| 07:04.49 | drmessano | Totally get it |
| 07:05.02 | Samot | I'm not going to provide FOP2 support. |
| 07:05.13 | drmessano | So instead we get "Well, I would say something, but" |
| 07:05.18 | Samot | I suggested FOP2 because using "sip show channels" to monitor calls was dumb. |
| 07:05.21 | Haris | I didn't ask for fop2 support. I asked for AM MGR credentials part |
| 07:05.24 | drmessano | Nice micro-aggression there |
| 07:05.27 | Samot | FreePBX WIKI |
| 07:05.32 | Samot | I already said that |
| 07:05.33 | Haris | going there |
| 07:05.46 | Haris | I don't need to piss anyone off |
| 07:05.55 | drmessano | No, of course not |
| 07:05.57 | Samot | "How to do stuff in FreePBX GUI? Where should I look?" |
| 07:05.58 | Haris | unless your really into having a discussion on that topic |
| 07:06.00 | drmessano | Because you require constant help |
| 07:06.01 | Samot | ^^ The WIKI |
| 07:06.01 | Haris | which most .. aren't |
| 07:06.13 | Haris | and since this is also not the place, I'd be kicked for being off-topic |
| 07:06.16 | drmessano | Don't want to shit where you eat |
| 07:06.18 | drmessano | I get it |
| 07:06.27 | Samot | Dude, let's not talk about "off-topic" |
| 07:06.35 | Haris | tell drmessano that |
| 07:06.35 | Samot | You bring FreePBX issues in here all the time. |
| 07:06.43 | drmessano | Samot: Hitler would talk about it |
| 07:06.48 | Haris | correction: asterisk |
| 07:06.49 | Samot | You literally are off-topic all the time. |
| 07:06.50 | Samot | No |
| 07:06.56 | Samot | FreePBX is a distro that uses Asterisk |
| 07:07.01 | Samot | There is no GUI support here. |
| 07:07.07 | Samot | Because FreePBX does things in a certain way |
| 07:07.10 | Haris | not asking gui Qs here |
| 07:07.14 | drmessano | Haris: Also, I don't need to be "told" anything |
| 07:07.16 | Samot | Sure you are. |
| 07:07.18 | Haris | AM MGR is not the gui part |
| 07:07.21 | drmessano | SO keep your fucking hateful comments to yourself |
| 07:07.22 | Haris | I'm going to wiki |
| 07:07.24 | Samot | "How do I setup AMI Manager" |
| 07:07.27 | Samot | 100% GUI |
| 07:07.28 | *** join/#asterisk pchero_work (~pchero@109.70.54.56) |
| 07:07.29 | drmessano | I dont need your virtue signalling BS |
| 07:07.31 | Haris | there's no hate in my comments .. yet |
| 07:07.35 | Haris | ah ok |
| 07:07.42 | Haris | didn't see am mgr is part of fpbx, not asterisk |
| 07:07.45 | Samot | There's a frigging section for it. |
| 07:08.06 | Haris | I never asked to give you that. neither did I start the conversation about religion |
| 07:08.10 | Samot | "Asterisk Manager Users" |
| 07:08.21 | drmessano | Haris: There was no conversation about religion |
| 07:08.29 | Samot | "against your religion" is a figure of speech. |
| 07:08.36 | Samot | as in "Are you adverse to doing X" |
| 07:08.41 | Haris | there was |
| 07:08.52 | Samot | You treated it like a conversation. |
| 07:08.53 | drmessano | You took a figure of speech and started a fucking battle over it and decided to start hurling micro-aggressions |
| 07:08.55 | drmessano | Go figure |
| 07:08.56 | Haris | hold this one for another time please. its work time rightn wo |
| 07:08.59 | Haris | right now+ |
| 07:09.00 | drmessano | But we wont get into that |
| 07:09.17 | Samot | Yes. |
| 07:09.18 | drmessano | Samot: Please stop arguing.. he needs help with his PBX |
| 07:09.23 | Samot | Let's not fight |
| 07:09.24 | drmessano | Shut up now please and just help |
| 07:09.28 | Samot | We need to support him for free. |
| 07:09.35 | Samot | While he gets paid to be told what to do. |
| 07:10.04 | drmessano | Lets focus on the real issue here |
| 07:10.08 | Haris | no Samot. you get far more than me and still complain |
| 07:10.10 | drmessano | Which is helping Haris 24/7 |
| 07:10.22 | Haris | I just don't have the time to explain to you how you get far more than me right now |
| 07:10.38 | Haris | you don't help me 24/7. stop whining please |
| 07:10.40 | Samot | I have no idea what that statement means. |
| 07:10.41 | drmessano | It's because i'm Jewish |
| 07:10.52 | Haris | I respect you if your jewish |
| 07:10.54 | drmessano | This is basically a hate crime |
| 07:10.57 | Haris | makes us cousins of sort(s) |
| 07:11.14 | Haris | but work now, discussion later please |
| 07:11.26 | Samot | We're not stopping you from working |
| 07:11.27 | Haris | I know its night time, your tired. you need to rest |
| 07:11.28 | drmessano | Yes, please dont distract us from helping you |
| 07:11.50 | drmessano | Im wide awake |
| 07:11.54 | drmessano | Whats your excuse? |
| 07:11.56 | Samot | Well he has to work. I don't want to bother him while he's working. |
| 07:12.24 | drmessano | Samot: Save that for another time, please |
| 07:12.31 | drmessano | Right now, lets go back to my issues.. all of them |
| 07:12.31 | Samot | Sorry. |
| 07:12.40 | Samot | That's a long list. |
| 07:12.49 | Samot | I'm not certified for most of the items on it. |
| 07:12.57 | Samot | But I'll give it a ago. |
| 07:13.01 | Samot | gets out the dolls |
| 07:13.14 | drmessano | points to a spot on the bear |
| 07:13.28 | Samot | No. |
| 07:13.34 | Samot | The animal ones are just for show. |
| 07:13.47 | drmessano | Show on us on the Bear where he touched you |
| 07:13.53 | drmessano | Show us on the Bear where he touched you |
| 07:14.46 | drmessano | Sorry, that was very Kekish of me |
| 08:04.54 | *** join/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 08:06.07 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 08:13.20 | *** join/#asterisk pagios (~pagios@ip183.ip-5-196-177.eu) |
| 08:18.14 | *** join/#asterisk shanth_ (~shanth@wsip-98-182-126-226.ph.ph.cox.net) |
| 08:22.51 | *** join/#asterisk sekil (~sekil@nat-73.net011.net) |
| 08:27.36 | *** join/#asterisk troyt (~troyt@2601:681:4102:9a51:44dd:acff:fe85:9c8e) |
| 08:27.43 | *** join/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 08:28.09 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 08:44.43 | *** join/#asterisk boris_t (~boris_t@82.112.41.207) |
| 08:49.04 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 08:49.12 | *** join/#asterisk daemonwrangler (~daemonwra@15.179.154.104.bc.googleusercontent.com) |
| 08:52.35 | *** join/#asterisk petris (sid19918@gateway/web/irccloud.com/x-atyzyarciiroqsle) |
| 09:00.20 | *** join/#asterisk Bordr (~Bordr@c-24-9-55-138.hsd1.co.comcast.net) |
| 09:02.09 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 09:12.05 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 09:16.27 | *** join/#asterisk pagios (~pagios@ip183.ip-5-196-177.eu) |
| 09:39.03 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 09:41.40 | *** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net) |
| 09:45.15 | wyoung | Hai drmessano! |
| 09:53.55 | *** join/#asterisk j-fish (uid178161@gateway/web/irccloud.com/x-qqvbjqiltxelypnu) |
| 10:10.01 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 10:20.21 | *** join/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 10:29.40 | *** join/#asterisk smlb (~smlb@unaffiliated/smlb) |
| 11:12.04 | pagios | mih |
| 11:20.38 | wyoung | and hi to you too pagios ! |
| 11:43.52 | *** join/#asterisk karelk (~karel@31.10.154.117) |
| 12:14.47 | *** join/#asterisk petris (sid19918@gateway/web/irccloud.com/x-fkkvgoiogjgifarr) |
| 12:14.49 | *** join/#asterisk craysiii (sid205503@gateway/web/irccloud.com/x-yfjqujkqeqnnhuny) |
| 12:23.56 | *** join/#asterisk zopsi (~zopsi@dir.ac) |
| 12:36.50 | *** join/#asterisk eric_hill (~eric_hill@wsip-184-180-163-60.ks.ks.cox.net) |
| 12:56.59 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 13:06.15 | *** join/#asterisk sekil (~sekil@cable-89-216-221-181.dynamic.sbb.rs) |
| 13:22.23 | *** part/#asterisk Haris (~haris@unaffiliated/haris) |
| 13:25.00 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 13:59.53 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:59.53 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 14:09.48 | *** join/#asterisk bford (d8cff501@gateway/web/freenode/ip.216.207.245.1) |
| 14:09.49 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:12.04 | *** join/#asterisk kharwell (kharwell@nat/digium/x-yyahurtivznspyah) |
| 14:12.04 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:23.06 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-wnlktkozkcmlahyv) |
| 14:23.07 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:23.10 | *** join/#asterisk newtonr (newtonr@nat/digium/x-vbaacgmbvviorukp) |
| 14:23.10 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:46.02 | *** join/#asterisk rwb (~Thunderbi@74.85.159.204) |
| 14:55.58 | darkunderlord | hey, if I've been given a PJSIP trunk password that has a percent and equal sign, do I just wrap it in quotes? |
| 14:56.09 | darkunderlord | or escape it somehow? |
| 14:56.20 | Samot | why? |
| 14:56.38 | darkunderlord | because I can't change it and I'm getting a 403 from them |
| 14:56.56 | darkunderlord | lol your name is almost my dads. his is Tom Ryan |
| 14:56.57 | Samot | You can't escape it |
| 14:57.10 | Samot | It's plain text |
| 14:57.18 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 14:57.47 | darkunderlord | the web said a little about quotes, but still no dice, and of course we're going live, so this isn't a test that I can take my time on :D |
| 14:58.12 | Samot | Did you ask the provider to change it? |
| 14:58.19 | Samot | To something without special characters? |
| 14:58.50 | darkunderlord | they are pretty dumb, so not sure I want to. It's backwoods country telecom lol |
| 14:58.59 | Samot | OK |
| 14:59.00 | Samot | So.. |
| 14:59.10 | Samot | The easiest and simplest solution is that. |
| 14:59.23 | Samot | And if they are dumb, then they shouldn't be your provider. |
| 14:59.27 | darkunderlord | yeah, just want to know that wrapping in quotes works. |
| 14:59.32 | Samot | No. |
| 14:59.35 | Samot | It's plain text. |
| 14:59.35 | darkunderlord | hahaha, no choice. It's the country |
| 14:59.42 | Samot | No. |
| 14:59.44 | darkunderlord | middle of nowhere |
| 14:59.44 | Samot | It's VoIP |
| 14:59.54 | Samot | All you need is Internet. |
| 14:59.58 | darkunderlord | they are the internet provider too, so I have to use them |
| 15:00.03 | darkunderlord | for at least something :D |
| 15:00.04 | Samot | OK |
| 15:00.11 | Samot | You do not have to use them for voice. |
| 15:00.27 | Samot | You're not stuck with them for voice. |
| 15:00.31 | Samot | You're not using a PRI |
| 15:00.35 | Samot | Or a T1 for voice only |
| 15:01.00 | Samot | Tell them that you can't have special characters in the password. |
| 15:01.12 | Samot | I've worked for ITSPs and Telcos, it is not an uncommon thing. |
| 15:14.25 | darkunderlord | I got him to change the pw. thanks. I'm still trying quotes |
| 15:14.36 | Samot | Why? |
| 15:14.45 | Samot | What part of "plain text" is not clear? |
| 15:15.20 | Samot | gH%v12 is not going to work just because you did "gH%v12" or gH\%v12 |
| 15:15.32 | Samot | One will include two sets of quotes |
| 15:15.41 | Samot | One will include the backslash |
| 15:15.46 | Samot | All part of the password. |
| 15:18.16 | *** join/#asterisk AnonRecluse13 (~AnonReclu@91.210.190.121) |
| 15:19.27 | sekil | asterisk can use separate presence server nowadays? |
| 15:19.43 | Samot | What do you mean? |
| 15:20.41 | sekil | there were some talk about doing it the proxy way...to send a publish to presence server |
| 15:21.21 | Samot | Yes. |
| 15:21.26 | Samot | I'm sure that has been. |
| 15:21.41 | Samot | Asterisk cares not about the existence of a presence server. |
| 15:21.57 | Samot | Asterisk does Presence/Subscriptions with HINTS. |
| 15:22.34 | file | https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State |
| 15:24.41 | sekil | file: thanks...event state compositor can be a proxy? |
| 15:24.49 | sekil | file: not asterisk only |
| 15:24.52 | Samot | Did you not read it? |
| 15:25.06 | Samot | Because if you did, you wouldn't be asking that question. |
| 15:25.23 | sekil | Samot: I did |
| 15:25.37 | Samot | So the part where it keeps say "Other entities" |
| 15:25.48 | Samot | And then says "You can use Kamailio" |
| 15:25.52 | Samot | Which is a proxy |
| 15:26.20 | Samot | It expressly says and gives examples of a third party even state compositor. |
| 15:26.37 | sekil | Samot: that I did not read.. |
| 15:26.44 | sekil | Samot: thank you |
| 15:28.25 | sekil | very good then |
| 15:28.57 | file | ultimately what you can and can't use is dependent on the other entity |
| 15:29.01 | file | if they support it then you can |
| 15:29.03 | file | if they don't you can |
| 15:29.08 | file | er can't |
| 15:30.21 | Samot | Well and the actual endpoint needs to support it |
| 15:30.31 | *** join/#asterisk jkroon_ (~jkroon@165.16.204.43) |
| 15:30.48 | file | to the endpoint it's a normal SUBSCRIBE |
| 15:30.49 | *** part/#asterisk AnonRecluse13 (~AnonReclu@91.210.190.121) |
| 15:30.55 | Samot | Right |
| 15:31.16 | Samot | But they generally that the $PROXY field from the SIP Registration settings |
| 15:31.30 | Samot | So the subscriber is user@$PROXY |
| 15:32.03 | Samot | Same with Presence. The endpoint needs to support using non-$PROXY hosts for these. |
| 15:32.29 | Samot | Like with Polycom's. You can set the resource server to be different then the proxy |
| 15:32.39 | Samot | So your BLF/Presence comes from that server. |
| 15:34.22 | file | depends on deployment model |
| 15:34.33 | Samot | Yes |
| 15:34.54 | Samot | But just as you say it's up the to the other entity to support this for Asterisk |
| 15:35.15 | Samot | It's also up to the endpoint to be able to support a secondary server for this stuff too. |
| 15:35.27 | sekil | yeah you can set different presence server on many phones now |
| 15:35.36 | Samot | Right |
| 15:35.38 | Samot | On "many" |
| 15:35.40 | Samot | Not "all" |
| 15:35.50 | Samot | So again, the endpoint also needs to support this setup. |
| 15:35.53 | file | if you maintain completely separate services yes, if you run a full proxy setup then it doesn't have to be different |
| 15:36.10 | sekil | afaik cisco and panasonic |
| 15:36.15 | Samot | As you said, it depends on how it is deployed. |
| 15:36.18 | sekil | I only have conf poly phones |
| 15:36.27 | file | yup, we just provide the functionality - how people deploy it is up to them |
| 15:36.48 | Samot | I'm just pointing out the things that can be pitfalls in such a deployment |
| 15:37.18 | sekil | my use would be to publish some different states asterisk can provide to interested phones |
| 15:37.26 | Samot | Really would suck spending all this time setting something like that up |
| 15:37.36 | Samot | and then trying to use Cisco 79XX phones. |
| 15:37.49 | sekil | I meant Cisco spa5xx |
| 15:37.54 | Samot | I understand that |
| 15:38.03 | Samot | I'm making a point. |
| 15:38.51 | Samot | This is like laying out the perfect buffet line.. |
| 15:39.06 | Samot | And then not having the proper serving utensils. |
| 15:39.13 | sekil | anyhow I was inquirying |
| 15:39.16 | *** join/#asterisk jkroon_ (~jkroon@154.73.35.201) |
| 15:39.48 | sekil | I switched to having proxy for regs and presence..but it's good to know new possibilities |
| 15:43.16 | sekil | also I guess a NOTIFY from proxy to asterisk would be relayed to a phone |
| 15:43.40 | Samot | The proxy would never send a NOTIFY to Asterisk |
| 15:43.57 | sekil | it would if configured |
| 15:43.57 | Samot | Not for this. |
| 15:43.58 | file | and Asterisk would never forward it |
| 15:44.04 | Samot | ^^^ |
| 15:44.24 | Samot | Asterisk <--> Proxy <--> Users |
| 15:44.28 | sekil | file: ok |
| 15:44.33 | sekil | Samot: no |
| 15:44.42 | sekil | Samot: users -> asterisk -> proxy presence server |
| 15:44.51 | sekil | Samot: that's what I'm talking about |
| 15:44.52 | Samot | Why? |
| 15:45.00 | sekil | Samot: that's beside the point |
| 15:45.18 | sekil | Samot: It would be possible to send the notify |
| 15:45.29 | sekil | Samot: but file said it would not be forwarded |
| 15:45.36 | sekil | Samot: so it's pointless |
| 15:46.47 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 15:48.32 | Samot | The proxy should be between the users and Asterisk |
| 15:48.34 | Samot | That's the point of it. |
| 15:48.46 | Samot | Otherwise it's not really a proxy |
| 15:48.52 | Samot | It's just a presence server. |
| 15:49.27 | Samot | If your device goes to INUSE on Asterisk. |
| 15:49.42 | Samot | Asterisk needs to send a NOTIFY to the subscribers watching that device's status. |
| 15:49.50 | sekil | Samot: there can be a case where you have other people's pbxes or your own hosted ones...and just want to have a separate presence server |
| 15:49.51 | Samot | So Asterisk would need to send it to the Presence server.. |
| 15:49.55 | Samot | That would need to send it to the devices. |
| 15:50.05 | sekil | Samot: I'm not saying if it's a good case... |
| 15:50.18 | Samot | sekil: I've worked for ITSPs and CLECs for over a decade. |
| 15:50.25 | Samot | I am well aware of how to do this. |
| 15:50.35 | sekil | Samot: I actually have a proxy and pbx cluster now....proxy does regs and presence |
| 15:50.40 | Samot | So when I say what you are doing is not right, I'm talking from experience. |
| 15:50.49 | Samot | But OK. |
| 15:51.12 | sekil | Samot: I'm NOT doing it.. |
| 15:51.20 | sekil | Samot: I'm discussing it.. |
| 15:51.30 | sekil | Samot: I'm doing the opposite as I said three times now.. |
| 15:51.31 | Samot | OK |
| 15:51.33 | Samot | Fine. |
| 15:51.37 | sekil | Samot: and I like your input |
| 15:51.52 | sekil | Samot: you asked why.. |
| 15:51.55 | sekil | Samot: I replied.. |
| 15:52.36 | Samot | So you have Asterisk <--> Proxy <---> Users now? |
| 15:52.42 | sekil | Samot: nope |
| 15:52.55 | Samot | You have nothing not? |
| 15:52.57 | Samot | You have nothing now? |
| 15:52.59 | sekil | Samot: I have users -> OpenSIPS -> FreeSWITCH pbxes |
| 15:53.05 | Samot | OK |
| 15:53.34 | sekil | Samot: OS handles REGs..BLFs...group and directed pickup...pretty much all signalling possible.. |
| 15:53.39 | Samot | Are you going to be using Asterisk PBXes? |
| 15:53.52 | sekil | Samot: no..but other people are |
| 15:54.05 | Samot | Uhm.. |
| 15:54.11 | Samot | So you're not going to be using Asterisk.. |
| 15:54.18 | Samot | But you want to know how to do it on Asterisk? |
| 15:54.30 | sekil | Samot: to do what on asterisk? |
| 15:54.39 | Samot | Well.. |
| 15:54.46 | Samot | You asked about Presence on Asterisk |
| 15:54.49 | Samot | Using proxies |
| 15:55.02 | Samot | But now you're saying you're not going to be using Asterisk. |
| 15:55.04 | sekil | Samot: no..I was asking if Asterisk can use a different presence server |
| 15:55.10 | sekil | Samot: and now I know |
| 15:55.45 | Samot | 11:19:25 AM S<sekil> asterisk can use separate presence server nowadays? <-- Poorly worded question |
| 15:56.06 | Samot | That implies Asterisk can USE a separate presence server |
| 15:56.08 | sekil | Samot: you do have much free time if you ask me |
| 15:56.17 | Samot | Not "Can Asterisk be a Presence server?" |
| 15:56.24 | Samot | Yes, apparently. |
| 15:56.57 | Samot | So much free time. |
| 15:57.11 | Samot | That I've taken three calls and done a few other things during this conversation. |
| 15:57.14 | sekil | Samot: in a manner of what we talked about |
| 15:57.24 | sekil | Samot: Asterisk would be a presence user agent |
| 15:57.31 | Samot | That is a horrible idea. |
| 15:57.43 | Samot | First, Asterisk doesn't do Presence in the standard way |
| 15:57.53 | sekil | Samot: it's not an idea...it's a discussion as I said.. |
| 15:57.53 | Samot | Second, it is not designed for that. |
| 15:58.06 | sekil | Samot: I do not have an idea.. |
| 15:58.30 | sekil | Samot: and thank you for your input...but I was just asking if Asterisk can use separate presence server.. |
| 15:58.41 | Samot | You did it again |
| 15:58.53 | Samot | Yes. Asterisk can use a separate presence server. |
| 15:59.07 | Samot | No. Asterisk should not be used AS A presence server. |
| 15:59.11 | sekil | Samot: an idea would be to maybe offload other people's pbxes or something...didn't really thought about it much |
| 15:59.29 | sekil | Samot: I'd like to know first what could be done..then think about how to do it.. |
| 16:00.42 | Samot | Asterisk <--> Proxy <--> Users |
| 16:00.49 | Samot | Use the Presence abilities of the proxy |
| 16:01.17 | *** join/#asterisk miralin (~Thunderbi@91.237.94.14) |
| 16:01.21 | Samot | Users subscriber to the proxy for NOTIFYs. It's where their PUBLISH and SUBSCRIBE requests will go. |
| 16:01.42 | sekil | Samot: uhh...say you have a pbx |
| 16:01.51 | Samot | Asterisk = PBX |
| 16:02.04 | sekil | Samot: and for some reason..you would like to offload its presence to a different server |
| 16:02.27 | Samot | You can with Chan_PJSIP |
| 16:02.32 | Samot | As file showed. |
| 16:02.33 | sekil | Samot: you would say to a pbx owner to send all subscribes via it's publishing to your proxy |
| 16:02.37 | Samot | You can't with Chan_SIP |
| 16:02.44 | Samot | It has to go to the proxy server with the users. |
| 16:02.46 | sekil | Samot: so that's the idea.. |
| 16:03.05 | Samot | sekil: The users have to exist on the presence server. |
| 16:03.07 | sekil | Samot: and proxy would eventualy returned all notifyes |
| 16:03.16 | Samot | You just can't randomly let people use it. |
| 16:03.18 | sekil | Samot: no they don't Samot |
| 16:03.30 | Samot | They have to SUBSCRIBE |
| 16:03.39 | Samot | In order to SUBSCRIBE you have to auth |
| 16:04.06 | sekil | Samot: if you have users -> something -> proxy |
| 16:04.29 | sekil | Samot: or users -> something -> something -> somethting -> proxy presence |
| 16:04.36 | Samot | If I want to use Server A for presence/MWI/BLF/etc.. |
| 16:04.44 | Samot | I need to be able to SUBSCRIBE to Server A |
| 16:04.52 | Samot | Which requires a user account |
| 16:04.54 | sekil | Samot: if you can get somehting somehting something to send proper data...proxy would relay notifies back |
| 16:05.23 | Samot | You're making this over complicated. |
| 16:05.53 | sekil | Samot: that's debatable.. |
| 16:05.58 | sekil | Samot: but not inaccurate |
| 16:06.17 | sekil | Samot: anyhow..I gtg..thanks man..always nice to talk to you.. |
| 16:12.13 | *** join/#asterisk Bordr_ (~Bordr@c-24-9-55-138.hsd1.co.comcast.net) |
| 16:18.59 | *** join/#asterisk u0m3 (~u0m3@86.127.157.203) |
| 16:19.56 | *** join/#asterisk anonymouz666 (aafe5147@gateway/web/freenode/ip.170.254.81.71) |
| 16:21.07 | anonymouz666 | hi everyone. anyone have any clue on how to recrod a video call on format h264, h263? not transcoding, not conference, just record a pass-through call. |
| 16:21.53 | anonymouz666 | it looks like a very simple task but not from asterisk point of view. |
| 16:23.31 | file | there is no ability to do that currently |
| 16:24.36 | anonymouz666 | @file: do you think could make any sense do dial normaly using but using calling Record as Macro parameter? |
| 16:24.45 | file | that will not record a call. |
| 16:25.39 | anonymouz666 | @file: so there's no way to do that... not even in a 'creative' way? |
| 16:25.43 | file | nope. |
| 16:25.58 | anonymouz666 | using monitor? moving to a confbridge or meetme? |
| 16:26.05 | anonymouz666 | none of these would work? |
| 16:26.09 | file | not for video. |
| 16:26.11 | Samot | Asking different ways, isn't going to change it. |
| 16:26.17 | anonymouz666 | alright |
| 16:26.46 | salviadud | can you wireshark it? |
| 16:26.46 | anonymouz666 | Samot: sometimes could lead to another path to reach the goal |
| 16:27.05 | Samot | If it was you and I having the talk, sure. |
| 16:27.05 | anonymouz666 | I can't, need to do that at server side. |
| 16:27.15 | Samot | But when the guy who designs and builds it says "No there isn't" |
| 16:27.29 | Samot | That's a pretty clear way. |
| 16:27.42 | Samot | Well I guess there is one "creative" way |
| 16:27.43 | anonymouz666 | yes, sure. I understand. |
| 16:27.57 | Samot | You can actually re-code the core Asterisk code |
| 16:28.07 | anonymouz666 | Samot: you can always do that. |
| 16:28.07 | Samot | And then rebuild it with this support. |
| 16:30.01 | anonymouz666 | the point is this is not good for the project. video today is a must have in most of big companies. |
| 16:30.23 | anonymouz666 | that's not me saying that, it is the market, the companies. |
| 16:31.45 | *** join/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 16:33.02 | Samot | Sure. |
| 16:33.05 | *** part/#asterisk woddy (~woddy@unaffiliated/woddy) |
| 16:33.15 | Samot | But the actual recording and archiving of the video isn't. |
| 16:33.23 | Samot | They mainly want the audio. |
| 16:33.27 | Samot | Primarly. |
| 16:33.34 | Samot | Primarily. |
| 16:34.13 | Samot | The recording of a video call is only as good as the recording of the audio part of the call. |
| 16:34.41 | Samot | Otherwise you have a silent movie or worse, a poorly dubbed kung-fu movie |
| 16:35.31 | Samot | Not to mention there is not video over the PSTN. |
| 16:38.53 | file | video isn't something you can do well in a month |
| 16:39.00 | file | it's a recent focus, so it'll improve over time |
| 16:39.07 | Samot | Right |
| 16:39.42 | Samot | 4 or 5 years ago we spent like 90-120 days testing video |
| 16:40.03 | file | take for example the recording - none of the core parts were designed to record video |
| 16:40.11 | file | er |
| 16:40.15 | file | for two party calls |
| 16:40.19 | Samot | Cameras, phone configs, quality, etc. |
| 16:40.26 | Samot | And got _zero return_ on it. |
| 16:41.51 | Samot | But yeah, now that there is an actual focus on it.. |
| 16:41.59 | Samot | Give it a few years, things will happen. |
| 16:42.43 | *** join/#asterisk karelk (~karel@31.10.154.117) |
| 16:42.45 | Samot | Video will be like SIP and DSL. |
| 16:43.05 | Samot | A bunch of little guys that want to be cutting edge will pour money and time into it. |
| 16:43.08 | Samot | Make it a thing.. |
| 16:43.56 | Samot | Then ATT, Comcast and others will get behind it and then shit will roll. |
| 16:44.07 | Samot | And the little guys will get undercut. |
| 16:45.16 | Samot | Then you'll be able to do PiP video calls on your Comcast X1000 system. |
| 16:45.36 | Samot | So you can video call your friends while binge watching Game of Thrones. |
| 16:48.54 | Samot | I have no doubt that ATT/Comcast/others have resources already focused on video solutions. |
| 16:49.39 | Samot | And resources focused on watching companies that are already providing video solutions. |
| 17:04.09 | *** join/#asterisk RovingWriter (~rubble@unaffiliated/rovingwriter) |
| 17:17.25 | *** join/#asterisk hexhaxtron (~hexhaxtro@thunix.org) |
| 17:17.53 | hexhaxtron | I got about 150 users on my system. Can I create an account for them all using PAM auth? |
| 17:18.39 | Samot | You mean for their devices? |
| 17:18.43 | hexhaxtron | Can I create a SIP for them all using PAM auth? |
| 17:18.47 | Samot | No. |
| 17:21.17 | RovingWriter | 150 users on your system... is this a thin client system? |
| 17:21.17 | Samot | It's either done with the .conf files or the RealTime database. |
| 17:21.46 | RovingWriter | if the users are in a LDAP, u can do it |
| 17:22.56 | Samot | Doesn't RT support LDAP |
| 17:22.58 | Samot | ? |
| 17:23.00 | Samot | I can't remember. |
| 17:24.09 | RovingWriter | it does |
| 17:24.30 | RovingWriter | i was just elaborating on your answer slightly |
| 17:24.44 | Samot | I figured. I know RT supports a few databases. |
| 17:24.52 | Samot | I just couldn't remember if LDAP was in the list. |
| 17:24.57 | RovingWriter | sure is |
| 17:25.19 | Samot | I don't LDAP. |
| 17:26.30 | Samot | Nor do I Dap. |
| 17:26.44 | Samot | I find them both silly. |
| 17:26.46 | Samot | But that's me. |
| 17:33.06 | [TK]D-Fender | Yes you can use LDAP |
| 17:53.37 | *** join/#asterisk j-fish (uid178161@gateway/web/irccloud.com/x-aevfwbtapbvdqyfc) |
| 18:05.59 | *** join/#asterisk Iamnach0 (~Iamnacho@ip70-171-163-5.om.om.cox.net) |
| 18:19.13 | *** join/#asterisk robinak (~quassel@unaffilated/robink) |
| 18:43.23 | *** join/#asterisk sekil (~sekil@cable-89-216-221-181.dynamic.sbb.rs) |
| 18:46.07 | *** join/#asterisk MechWar (b3b2a018@gateway/web/freenode/ip.179.178.160.24) |
| 19:09.28 | *** join/#asterisk Typhon (~Typhon@dslb-088-065-219-190.088.065.pools.vodafone-ip.de) |
| 19:39.29 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 19:42.02 | *** join/#asterisk voipmonk (~voipmonk@138.68.2.1) |
| 19:47.20 | *** join/#asterisk jab416171 (~jab416171@c-67-172-234-125.hsd1.ut.comcast.net) |
| 20:13.38 | *** join/#asterisk Georbe (~George@athedsl-4434195.home.otenet.gr) |
| 20:19.13 | Georbe | Hi. I have a B410P ISDN card. I have configured the 1st port in TE mode, and I have connected an NT ISDN unit on it (provided by my TelCo). The light turns Green. The port is working fine, although I am getting the line "Primary D-Channel on span 5 up" and immidiatly the line "Primary D-Channel on span 5 down" all the time. |
| 20:20.34 | Georbe | After 1 or 2 hours, the ISDN port stops working. It is still Green, but no calls can be made through this port. Also the messages about the D-channel stops coming up on my logs. |
| 20:21.49 | Georbe | The only thing I can do to fix it, is to run "dahdi restart". But after 1 or 2 hours, it stops working again. Can anyone help me on this issue please? Thanks. |
| 20:25.15 | RovingWriter | so, do most * people use just the normal config files, extensions.conf, sip.conf, and go from there? most people don't use realtime? and most people write their dialplan in extensions.conf, as opposed to doing stuff dynamically? |
| 20:28.53 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 20:29.34 | MechWar | RovingWriter: Many people don't have knowledge to develop efficiently a dynamic dialplan |
| 20:31.48 | cresl1n | Georbe: is this setup using ptmp signaling or ptp signaling? |
| 20:31.55 | Georbe | ptp |
| 20:32.28 | cresl1n | if you get Primary D-channel on span x down very quickly, that sounds a lot more like ptmp signaling |
| 20:32.54 | cresl1n | with ptmp, usually the d-channel isn't always active, is brought down quickly after it is used |
| 20:32.59 | Georbe | I get the warning message every EXACTLY 16 seconds |
| 20:32.59 | *** join/#asterisk elguero (~miguel323@74-95-21-41-Connecticut.hfc.comcastbusiness.net) |
| 20:33.27 | Georbe | The error message is the following: "WARNING[23478]: sig_pri.c:1211 pri_find_dchan: Span 5: D-channel is down!" |
| 20:33.54 | Georbe | So your recomendation is to switch it to ptmp? |
| 20:34.02 | cresl1n | that might be what it's trying to do |
| 20:34.08 | Georbe | I will do it in a few momments |
| 20:34.17 | Georbe | *few minutes |
| 20:41.05 | Georbe | It seems that that was the issue... |
| 20:41.28 | Georbe | the warning messages (about the D-Channel) are gone |
| 20:41.33 | Georbe | Thank you very much. |
| 20:43.34 | Georbe | FunFact: the same TelCo gives me some ISDN Lines that they are working as PTP, and some ISDN Lines that they are working as PTMP!!! Isn't that weird? |
| 20:49.45 | Georbe | is there any way to find out if the D-Channel on a specific port is up or down? (using CLI) |
| 20:58.37 | [TK]D-Fender | pri show spans |
| 20:58.41 | [TK]D-Fender | pri show span X |
| 20:58.44 | [TK]D-Fender | take your pick |
| 21:02.20 | Georbe | what's the difference between those two outputs: 1) "B4XXP (PCI) Card 1 Span 1 OK" and 2) "PRI span 5/0: Up, Active" |
| 21:03.07 | Georbe | Alarm OK on the first output, is equal to Green light, and it is also equal with the "Up" on the second command output above? |
| 21:03.28 | Georbe | and the "Active" is the status of the D-Channel? |
| 21:06.36 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 21:11.47 | cresl1n | Ugh |
| 21:11.58 | cresl1n | I always forget what the difference between up and active means |
| 21:12.09 | cresl1n | I usually would heuristically deduce the line state |
| 21:12.14 | cresl1n | so look at Q.921 debug |
| 21:12.43 | cresl1n | see if multiframe mode is established |
| 21:13.00 | cresl1n | (i.e. sabme and ua are being exchanged by the two ends at least once every 10 seconds or so) |
| 21:13.21 | cresl1n | if that's not happening, and it's a ptmp line, that's ok, because the d-channel could be down |
| 21:13.34 | cresl1n | if it's a ptp line, that's bad, because it means the d-channel's down but it should be up |
| 21:13.47 | cresl1n | <â maintained/rewrote big chunks of libpri |
| 21:20.55 | *** join/#asterisk Georbe (~George@athedsl-4434195.home.otenet.gr) |
| 21:21.03 | Georbe | thanks cresl1n |
| 21:21.21 | Georbe | I will test the Q.921 debug right now |
| 21:26.46 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:27.18 | *** join/#asterisk tuxd00d (~tuxd00d@ip68-106-11-138.ph.ph.cox.net) |
| 21:32.57 | *** join/#asterisk Casper (~ThePhanto@unaffiliated/thephantom) |
| 21:34.14 | Casper | Hi there, voip phone. I was looking at wifi phones and couln't find any good looking inexpensive ones. All I find is one with an ethernet base and non-wifi wireless... Is that because wifi have issues? or I don't look hard enought? |
| 21:35.16 | RovingWriter | what do you consider inexpensive? |
| 21:35.36 | RovingWriter | right off the top of my head, the Yealink T29G is is wifi compatible, and sells for $120 |
| 21:36.12 | RovingWriter | also, your mobile phone is a wifi-enabled sip phone. :) |
| 21:36.45 | RovingWriter | might need to be more specific because good looking is subjective, and inexpensive is also relative |
| 21:38.19 | *** join/#asterisk d1gital (~d1gital@fsf/member/d1gital) |
| 21:41.48 | Casper | 220$ is expensive, and ebay cheap is inadequate usually :D |
| 21:42.38 | Casper | really, looking for something good that don't cost too much, and trying to figure out why they most use a base station (vendor lockin or a different purpose) |
| 21:44.27 | RovingWriter | well i have you an example of a phone that is wifi, and is $120 |
| 21:50.15 | DanQuinney | Yealink are very decent for their price |
| 21:53.06 | tuxd00d | You can pick up referburished yealinks next to free. |
| 21:54.18 | tuxd00d | You can use a WiFi Ethernet adapter to have any phone use WiFi. |
| 21:54.29 | RovingWriter | true ^ |
| 21:55.04 | RovingWriter | so, i assome many of you have at least tried twilio's stuff - what keeps u going with asterisk? seems like twilio has more stuff to it? |
| 21:55.51 | tuxd00d | I havenât had any problems with that method. Some brands will provide 99% update, other 100%.. Some adapters seem to âpower saveâ or something which doesnât allow for packets to route and return before the timeout. |
| 21:56.10 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 21:56.38 | tuxd00d | RovingWriter: Twillo has software? |
| 21:57.05 | Casper | will look for yealinks |
| 22:01.57 | DanQuinney | I can't imagine many itsp's use twilio |
| 22:03.49 | *** join/#asterisk moose55 (4a65f8de@gateway/web/freenode/ip.74.101.248.222) |
| 22:04.06 | moose55 | can anyone help me with vlan for polycom phones and daisy chained PCs? |
| 22:04.25 | moose55 | all my straight runs to phones work flawlessly |
| 22:04.51 | moose55 | once I introduce the daisy chained connections ( all of which are on their own switch ) - my voice traffic is horrendous |
| 22:04.58 | moose55 | I am using HP 1920 switches |
| 22:05.34 | moose55 | I know this is not exactly asterisk related but I would suspect you guys have set up environments before |
| 22:06.05 | Georbe | Can I record an incoming call as soon as it lands into a queue? Currently I have "Call recording" of the queue to yes, but it only records the call as soon as the extension answers the call. |
| 22:09.35 | Georbe | moose55: you have two VLANs. One for the PCs, and one for the Voice. On each port of the switch, you have to set up the Voice VLAN as tagged and the PCs VLAN as untagged. |
| 22:10.07 | moose55 | what if I dont know the vlan ID for the PCs? |
| 22:10.56 | Georbe | don't you have access to the configuration of the switch? |
| 22:11.23 | moose55 | I will tomorrow...their IT is no longer in |
| 22:11.37 | moose55 | We moved the daisy PCs to our switch |
| 22:11.42 | RovingWriter | tuxd00d, twilio has a full API, where you just host your dialplan with them, or they can hit webhooks and get info from you |
| 22:12.34 | RovingWriter | they dont necessarily have software, but they provide the same end result as a * box can, mostly. |
| 22:12.47 | tuxd00d | RovingWriter: Oh, thatâs what you meant. |
| 22:13.07 | RovingWriter | * can do some things twilio can't, and twilio can do things * doesn't without a lot of customization |
| 22:13.22 | *** join/#asterisk d1gital (~d1gital@fsf/member/d1gital) |
| 22:13.40 | Georbe | moose55: the command "sho vlan" will show you all the VLANs your network have, and the VLAN configured for each port of the switch |
| 22:14.43 | moose55 | I have to tag the vlan on the polycom config first correct |
| 22:14.52 | Georbe | then, for each port you will connect an IP Phone and a PC, you have to do this configuration: (let's say Voice VLAN is 10 and PCs VLAN is 50) |
| 22:15.14 | Georbe | vlan 10 |
| 22:15.28 | Georbe | tagged <port name> |
| 22:15.31 | Georbe | vlan 50 |
| 22:15.37 | Georbe | untagged <port name> |
| 22:15.48 | Georbe | that's it |
| 22:16.06 | *** join/#asterisk Dovid (~dovid@ool-4573a525.dyn.optonline.net) |
| 22:16.11 | moose55 | but I have to set the VLAN ID in the Polycom to be 10? |
| 22:16.21 | Georbe | you can also specify a port range (where ever I say <port name> |
| 22:16.33 | *** join/#asterisk Bhakimi (~textual@rrcs-69-75-121-202.west.biz.rr.com) |
| 22:16.54 | Bhakimi | does anyone in here know someone who has build a ringless voicemail drop application using asterisk ? |
| 22:17.26 | Georbe | do you use DCCHP? |
| 22:17.32 | Georbe | *DHCP |
| 22:17.37 | moose55 | static set all the phones |
| 22:20.55 | Georbe | If you enable LLDP on the HP switch, there is nothing to do on the phones. |
| 22:21.39 | Georbe | LLDP is a protocol just like Cisco's CDP |
| 22:22.28 | Georbe | polycom and HP switch will exchange data using LLDP and then polycom will know what the Voice VLAN is |
| 22:22.39 | tuxd00d | Bhakimi: It is not clear what you are asking for. |
| 22:23.02 | moose55 | so the polycom has LLDP enabled, I can leave it alone |
| 22:23.11 | Bhakimi | there are providers out there that call mobile numbers in a way to jam the call which then lets them leave a message |
| 22:23.26 | Bhakimi | they call it ringless voicemail drops or voicemail drops |
| 22:23.29 | moose55 | in the switch ( default ), I go to VLAN and create VLAN ID 122 |
| 22:25.30 | Georbe | yes |
| 22:25.47 | moose55 | Then go to Voice VLAN and add all the ports to VLAN 122 and set mode to (manual?) |
| 22:26.13 | Georbe | switchport mode access |
| 22:26.42 | Georbe | switchport access vlan 122 (if the DATA Vlan is 122) |
| 22:27.11 | tuxd00d | Bhakimi: Do they use a backdoor to the VoiceMail system of each mobile carrier? |
| 22:30.16 | moose55 | voice is 122 |
| 22:32.50 | *** join/#asterisk rwb (~Thunderbi@65.183.151.121) |
| 22:32.59 | moose55 | I dont have CLI access to the switch |
| 22:36.44 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 22:37.30 | Georbe | I don't know how you can do it from the Web interface. Actually the first time I have tried to do it through the Web interface, I didn't managed to do it, and that'a why I switched to CLI. |
| 22:38.56 | Georbe | but maybe YOU will figure it out. Give it a try. What you are trying to do is to give *tagged* access to Voice VLAN and *untagged* access to DATA VLAN. |
| 22:38.58 | moose55 | so the uplink to my router should be tagged vlan 122 though correct and with link type access |
| 22:39.31 | Georbe | the uplink with the router has to be in trunk mode |
| 22:39.53 | moose55 | but does the router need anything configured on it too? |
| 22:40.19 | Georbe | it is allready in trunk mode. If it is not, then you need another solution |
| 22:40.35 | Georbe | maybe make it a flat network without VLANs |
| 22:40.52 | moose55 | i tried...I have a separate subnet for the phones |
| 22:41.08 | moose55 | once I introduce the daisy switch, the voice goes coppy instantly |
| 22:41.16 | Georbe | the subnet is configured in the router also... isn't it? |
| 22:41.16 | moose55 | then as soon as I unplug, its fine |
| 22:41.20 | moose55 | yeah |
| 22:41.41 | Georbe | in what interface? is it a VLAN interface? |
| 22:41.48 | moose55 | no |
| 22:41.58 | moose55 | it is a separate router |
| 22:42.02 | moose55 | with its own static IP |
| 22:42.14 | moose55 | i was starting with a fresh HP1920 |
| 22:43.10 | Georbe | what about the DATA subnet... where is the router of the DATA subnet? |
| 22:43.21 | Georbe | and how is connected to HP1920? |
| 22:43.33 | moose55 | right |
| 22:43.34 | moose55 | so |
| 22:44.08 | moose55 | I have a router on a Pubic IP from ISP with DHCP off and its own LAN subnet |
| 22:44.19 | moose55 | that is uplinked to two switches ( A and B ) |
| 22:44.20 | Georbe | *and how is the router connected to HP1920? (I am talking about the DATA network) |
| 22:44.24 | Georbe | ok |
| 22:44.37 | Georbe | and the HP1920 where is connected? |
| 22:44.48 | moose55 | all pheons are on their own cable, no daisy, to switches A and B |
| 22:45.15 | moose55 | I then have a switch with PC and Phones daisied |
| 22:45.30 | moose55 | when I uplink that switch to the router, voice traffic still works |
| 22:45.41 | moose55 | because the PCs dont know my static DHCP off network |
| 22:46.10 | moose55 | its when I plug the uplink to the data network ( which is another switch ), the voice network gets choppy |
| 22:46.55 | Georbe | there is something wrong with your setup |
| 22:47.11 | Georbe | tell me more about the switched A and B |
| 22:47.23 | Georbe | do they support VLANs? |
| 22:47.25 | moose55 | nothing more, they are all HP 1920s ( for the voice network ) |
| 22:47.27 | moose55 | yeah |
| 22:47.39 | moose55 | they are all home run connections to phones only |
| 22:47.43 | moose55 | uplinked to a router |
| 22:47.47 | Georbe | so you have 3 HP1920 switches? |
| 22:47.58 | moose55 | yeah for my voice network |
| 22:48.08 | moose55 | theres 80 phones |
| 22:48.23 | Georbe | and how many switches do you have for your DATA network? |
| 22:48.54 | moose55 | not sure, I think like 5 which is why I need to stop my subnet from accessing the data uplink port |
| 22:49.15 | moose55 | or 3 switches as well, they have 2-3 as backplane connected |
| 22:50.47 | Georbe | The right configuration is the following: |
| 22:52.38 | Georbe | (Internet) --- (router)(ether1) ---trunk--- (switch A) ---trunk--- (switch B) ---trunk--- (switch C) |
| 22:53.03 | Georbe | ether1 on router has to be configured with vlans |
| 22:53.24 | Georbe | let's say VLAN 10 (for Voice) and Vlan 50 (for DATA) |
| 22:53.43 | *** part/#asterisk Casper (~ThePhanto@unaffiliated/thephantom) |
| 22:54.17 | Georbe | all ports from all three switches (that connects each switch to another) has to be in trunk mode |
| 22:54.44 | moose55 | does the router port need to be tagged, untagged, or excluded |
| 22:55.29 | Georbe | ther is no such thing on the router (tagged or untagged) |
| 22:55.59 | Georbe | if you configure vlans in a specific interface on a router, all VLANs are tagged |
| 22:56.31 | Georbe | that's why you configure the ports on all the switches as trunk. |
| 22:57.15 | Georbe | and wherever you connect a PC or a Phone (or both), you configure it in access mode |
| 22:57.29 | moose55 | im using a low end router...i can add the port to be a member of a vlan |
| 22:57.51 | Georbe | what router do you use? |
| 22:58.25 | moose55 | I am currently using an RV130 since I wasnt expecting this....I can bring an 1840 tomorrow but I am not sure of the current IOS |
| 22:59.09 | moose55 | I tagged ports 1,3,4 with a membership to vlan10 |
| 22:59.28 | Georbe | In Cisco routers, you configure subinterfaces (with dot1q) |
| 22:59.49 | Georbe | That's how you configure VLANs on Cisco routers. |
| 23:00.18 | moose55 | yeah - I dont think this will do that |
| 23:01.23 | Georbe | RV130 uses the tagged/untagged thing |
| 23:01.28 | moose55 | yeah |
| 23:02.02 | *** part/#asterisk kharwell (kharwell@nat/digium/x-yyahurtivznspyah) |
| 23:02.06 | Georbe | Let;s say that port2 is connected to your switch |
| 23:02.11 | moose55 | yeah |
| 23:02.43 | Georbe | then you have to set up all the VLANs you have as tagged on this port |
| 23:03.04 | Georbe | except the first VLAN, whitch will be untagged. (it is called management VLAN) |
| 23:03.10 | moose55 | yeah |
| 23:03.44 | moose55 | so I set port 2 to be tagged vlan10 and port 3 to be tagged vlan10...goes to switch A and Switch B |
| 23:03.59 | Georbe | on all other ports you have to configure all the vlans as excluded |
| 23:04.10 | moose55 | the uplink port on switch A and Switch B need to be tagged vlan10 correct |
| 23:05.01 | Georbe | where do you use port3? |
| 23:05.35 | moose55 | switch B |
| 23:06.26 | Georbe | and port2 is connected to switch A ? |
| 23:06.31 | moose55 | yes |
| 23:07.13 | Georbe | is there any cable connecting Switch A with Switch B? |
| 23:07.19 | moose55 | no |
| 23:07.29 | Georbe | this is wrong |
| 23:07.49 | moose55 | but the RV utilizes the 4 ports on the back as a switch as well |
| 23:08.40 | Georbe | you have to connect Switch A with Switch B, so you don't bottlenecks |
| 23:09.49 | Georbe | Sorry... RV130 has Gigabit ports, so there is no problem |
| 23:09.55 | moose55 | ok good |
| 23:10.28 | moose55 | so do I tag the uplink ports to the RV as vlan10 |
| 23:11.10 | Georbe | whatever you do on router, that's what you will do on the uplink ports |
| 23:11.44 | moose55 | ok |
| 23:12.09 | moose55 | so how does the polycom/pc know to use that port? |
| 23:17.57 | moose55 | so how does the polycom/pc know to use that port? |
| 23:19.22 | Georbe | polycom (not polycom/pc), uses LLDP and gets the information from the switch. It knows then what is the Voice VLAN and what is the DATA VLAN |
| 23:19.53 | Georbe | and then "uses" the Voice VLAN for itself, and "gives" the data VLAN to the PC. |
| 23:20.03 | Georbe | the PC doesn't know anything |
| 23:20.28 | Georbe | polycom it's just like a switch with two ports |
| 23:21.38 | Georbe | on the first port (the one connected to the switch) gets the two VLANs, and on the second port (the one connected to the PC) "gives" the DATA VLAN untagged. |
| 23:23.15 | Georbe | whenever (on the same port of the switch as mentioned on the above line) you connect the PC only (and not o phone), the PC "understands" only the untagged VLAN, and "ignores" the tagged VLANs |
| 23:27.33 | moose55 | for some reason this isnt working for me |
| 23:27.39 | moose55 | not sure if I am doing something wrong |
| 23:28.54 | moose55 | I tag the ports on the RV as members to vlan10, port 2 goes to switch C on port 24. I tag port 24 as vlan10. |
| 23:31.51 | moose55 | in voice vlan, i can add it to be vlan10. then voice vlan port mode has options of No Change, Auto, Manual and voice vlan port state has options No Change, Enabled, Disabled |
| 23:32.28 | moose55 | and then add all ports into the voice vlan10 |
| 23:39.53 | *** join/#asterisk moose55_ (4a65f8de@gateway/web/freenode/ip.74.101.248.222) |
| 23:44.06 | *** join/#asterisk moose55 (4a65f8de@gateway/web/freenode/ip.74.101.248.222) |