| 00:14.49 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 00:22.38 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:22.38 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 00:33.02 | *** join/#asterisk matrix1233 (~matrix123@197.2.184.182) |
| 01:05.01 | nickaugust | when I call someone it shows up as <extension>@<ipaddress> is there a setting in asterisk to make it show up as <extension>@<hostname> instead? also i'm having problems with presense now shwoing up in clients could this be the cause? |
| 01:05.14 | nickaugust | presence |
| 01:09.08 | Samot | nickaugust: You need to clarify yourself a bit. |
| 01:09.34 | Samot | How are you calling someone? It is another device on the PBX? Through a trunk? |
| 01:10.58 | *** join/#asterisk matrix1233 (~matrix123@197.2.184.182) |
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| 01:16.09 | *** join/#asterisk Y04NN (~y04nn@nayon.fr) |
| 01:17.08 | nickaugust | Samot: I'm calling another person on the same server yesd |
| 01:17.50 | Samot | Show a call to another device on the server. |
| 01:17.52 | nickaugust | actually presence works in jitsi but not in linphone so i'm wondering if its some hostname settings, but shouldnt the server set that? |
| 01:17.54 | Samot | asterisk -rvvvvvvvv |
| 01:17.58 | Samot | sip set debug on |
| 01:18.02 | nickaugust | ok one sec |
| 01:18.04 | Samot | ~pb |
| 01:18.04 | infobot | it has been said that pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:19.14 | nickaugust | someone is trying to brute force me right now so the logs are all clogged up, can I set sip debug just for one extension? |
| 01:20.30 | Samot | You should fix that first. |
| 01:20.43 | nickaugust | lol do you handle that in asterisk or your firewall? |
| 01:21.05 | Samot | I would handle it with iptables. |
| 01:21.16 | nickaugust | yeah i'll look at doing some pf rules |
| 01:21.28 | nickaugust | it never seems to stop though, just from new ips |
| 01:21.29 | Samot | And drop requests from the IPs that are attacking you |
| 01:22.01 | nickaugust | can I slow down the rate at which asterisk accepts registration requests? that would probably help |
| 01:22.44 | Samot | I would look at how to rate limit sip requests through iptables, there's a bit of documentation out there on it. |
| 01:28.27 | nickaugust | Samot: |
| 01:28.36 | nickaugust | whoops, one sec |
| 01:28.46 | nickaugust | http://pastebin.ca/3786605 |
| 01:29.00 | nickaugust | so this is 102 calling 111 |
| 01:29.20 | nickaugust | theres a couple other failed registrations in between but should give an idea whats going on |
| 01:30.21 | nickaugust | so even when I dial out on linphone it says i'm calling extension@ipaddress rather than extension@hostname which is how I have is saved in the contacts list |
| 01:30.35 | nickaugust | its making me wonder if thats whats causing the presence problems |
| 01:31.48 | Samot | So you're calling with the SIP URI? |
| 01:32.52 | Samot | Why are you dialing with a SIP URI? |
| 01:33.21 | nickaugust | should I just put the extension? |
| 01:33.51 | Samot | Yes. |
| 01:34.05 | Samot | Or not, that's up to you. |
| 01:34.28 | nickaugust | ive tried both ways, setting the contacts up as extensions or full uris |
| 01:34.51 | nickaugust | if I dial out just the extensions it shows me dialing extension@hostname but the receiving caller still sees extension@ip |
| 01:35.48 | Samot | You're confusing what the softphone is showing vs what is actually happening. |
| 01:35.55 | Samot | The softphone dials to Asterisk. |
| 01:36.01 | Samot | Asterisk then does stuff with it. |
| 01:36.22 | Samot | In this case you want it to bring up a sip channel and call another device. |
| 01:36.29 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:36.42 | Samot | How you send that call to the other device, that's on your dialplan. |
| 01:36.51 | Samot | And what information you set with it... |
| 01:37.16 | nickaugust | gotcha, so I could likely set the hostname to show from the dialplan |
| 01:37.28 | Samot | Why? |
| 01:37.34 | Samot | Why do you need the hostname in the callerid?! |
| 01:37.52 | nickaugust | I don't I'm trying to figure out the presence issues |
| 01:38.02 | nickaugust | look at the first two messages in the log |
| 01:38.08 | Samot | oK so that is completely different. |
| 01:38.10 | nickaugust | you see the invite and the it 401s |
| 01:38.23 | Samot | Presence doesn't have INVITEs |
| 01:38.57 | Samot | Also.. |
| 01:39.13 | Samot | ALL outbound calls from your phone are AUTH'd. |
| 01:39.33 | Samot | So you will see the INVITE and 401 challenge response and then the INVITE again with the auth details in the digest. |
| 01:39.40 | nickaugust | ah, gotcha |
| 01:40.09 | Samot | Registration is for _inbound_ calls to your device. |
| 01:40.15 | Samot | So Asterisk knows where to send them. |
| 01:40.28 | nickaugust | ok so actually in this log there is no presence messages, one sec i'll make a new one |
| 01:40.38 | Samot | Did you create hints? |
| 01:40.44 | nickaugust | yeah ive got the hints |
| 01:41.01 | nickaugust | and the state shows on the server |
| 01:41.02 | Samot | What is the issue with presence? |
| 01:41.17 | Samot | It's not updating? |
| 01:41.23 | nickaugust | it doesnt show in linphone, but id does in jitsi |
| 01:41.40 | Samot | Because you need to subscribe to presence. |
| 01:41.48 | Samot | Jitsi sends mass subscriptions out. |
| 01:42.17 | nickaugust | I think I did though when I click the box in the contact list |
| 01:42.21 | Samot | Again, you are mixing up with out an application is doing something vs how it is really done. |
| 01:42.23 | nickaugust | hold on you'll see in the logs it 401s as well |
| 01:42.35 | nickaugust | unless i'm mistaking that again |
| 01:42.59 | Samot | -with out +what |
| 01:43.16 | Samot | You need to tell Linphone to subscribe to the presence of the other users you want to watch |
| 01:43.24 | Samot | Just like if you want to BLF their devices. |
| 01:43.36 | Samot | But are you updating their presence when calls come in? |
| 01:44.19 | Samot | You understand what presence is for right? |
| 01:45.32 | nickaugust | http://pastebin.ca/3786611 |
| 01:45.40 | nickaugust | yeah I understand that |
| 01:45.49 | nickaugust | whats BLF? |
| 01:46.05 | nickaugust | I set the hints and I see the state change on the server |
| 01:46.16 | nickaugust | i just want the icon next to their name to show if they're on the phone |
| 01:46.24 | Samot | Does Linphone support presence? |
| 01:46.28 | Samot | OK |
| 01:46.34 | Samot | That's not presence. |
| 01:46.41 | Samot | That's BLF, Busy Lamp Field. |
| 01:46.47 | Samot | You want to watch the status of their device. |
| 01:46.47 | nickaugust | yeah ive... ah, ok |
| 01:46.53 | Samot | Presence = Uuser |
| 01:46.53 | nickaugust | like I dont need "eating lunch now" or whatever |
| 01:46.56 | Samot | Presence = User |
| 01:46.59 | nickaugust | i jsut want to see if they are on the phone or not |
| 01:47.04 | Samot | That's BLF |
| 01:47.14 | Samot | You need to create hints for the devices. |
| 01:47.20 | nickaugust | yeah ive got that done |
| 01:47.40 | Samot | SIP/2.0 489 Bad Event |
| 01:48.17 | nickaugust | yeah on SUBSCRIBE |
| 01:48.23 | *** join/#asterisk matrix1233 (~matrix123@197.2.184.182) |
| 01:49.03 | Samot | Right |
| 01:49.10 | Samot | Which is how you watch hints |
| 01:50.05 | nickaugust | failure: no Accept header: pvt: stateid: -1, laststate: 0, dialogver: 0, subscribeco |
| 01:50.20 | nickaugust | looks like the request is missing a header |
| 01:50.36 | Samot | core show hints |
| 01:50.39 | *** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net) |
| 01:50.42 | Samot | Show all the hints that are generated |
| 01:51.57 | nickaugust | http://pastebin.ca/3786614 |
| 01:53.02 | Samot | Event: presence |
| 01:53.08 | Samot | ^^ Just caught that.. |
| 01:53.18 | nickaugust | in the log? |
| 01:53.38 | Samot | It's the Event of your SUBSCRIBE |
| 01:53.49 | Samot | You're telling to to subscribe to presence.. |
| 01:53.54 | nickaugust | so should it not be asking for presence |
| 01:53.55 | Samot | Or Linphone is.. |
| 01:54.03 | Samot | Make sure you have Linphone configured right. |
| 01:54.03 | nickaugust | ok.. so if I take that out of the settings on the contact |
| 01:54.19 | nickaugust | but how do I tell it to do BLF? |
| 01:54.29 | Samot | You're going to have to read the docs. |
| 01:54.34 | Samot | Busy Lamp Field. |
| 01:54.55 | Samot | I can't recall if Linphone supports it or not. |
| 01:55.33 | nickaugust | ah ok i'll see what i can find |
| 01:55.40 | nickaugust | so I do NOT want presence |
| 01:56.36 | nickaugust | if linphone does not support BLF can I just turn on presence to get the state of the line? |
| 01:57.32 | nickaugust | side note just saw this "Note: the accept header is optional, but in some server you need this header to avoid 489 bad event error" |
| 02:04.21 | nickaugust | i'm starting to think linphone doesnt support BLF |
| 02:04.53 | nickaugust | is there a better softphone that supports BLF and runs on linux? |
| 02:05.19 | nickaugust | I guess I could migrate everyone to jitsi but it just has so much stuff in there I dont need |
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| 02:29.10 | [TK]D-Fender | BLF = presence |
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| 02:33.15 | Samot | In a sense, yes. It watches the state of the device. |
| 02:33.34 | Samot | But Presence is terms of having a Presence server, etc. Is for the status of the user, not the device. |
| 02:34.06 | Samot | Device state != User state |
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| 03:42.42 | nickaugust | [TK]D-Fender: did you see this last log? the presence events are returning 401 but the hints and state shows correctly on the server |
| 03:43.19 | nickaugust | http://pastebin.ca/3786614 |
| 03:43.37 | [TK]D-Fender | Sorry, an error has occurred. Reason: That is an invalid ID, or the post has expired. |
| 03:44.29 | [TK]D-Fender | <Samot> Device state != User state <- there's a semantic attached because * does not ACCEPT it from devices |
| 03:44.34 | [TK]D-Fender | it's a one-way relationship |
| 03:44.54 | Samot | Right. |
| 03:44.54 | [TK]D-Fender | so * shows based only on wthe channels it is maintaining |
| 03:45.00 | [TK]D-Fender | but it is still presence |
| 03:45.06 | Samot | Asterisk Presence is not done like a standard Presence server. |
| 03:45.12 | Samot | You have to update the hints with the dialplan. |
| 03:45.17 | Samot | Right... |
| 03:45.19 | [TK]D-Fender | just not "smart" in integrating with phones where you can tell it "I'm busy" |
| 03:46.10 | Samot | There are cases where the phone needs to stay in idle states even if the user is busy |
| 03:46.29 | Samot | Sometimes the phone cannot speak for the user and vice versa. |
| 03:46.39 | Samot | I can be on the phone but still chat. |
| 03:46.46 | Samot | Those a chat system. |
| 03:46.55 | [TK]D-Fender | "If he ain't talkin', he ain't busy." |
| 03:47.04 | Samot | Depends on the agent and their needs. |
| 03:47.16 | [TK]D-Fender | And alternate views on "reality" :) |
| 03:48.00 | Samot | When you have provided services to various arrays of companies and entities the use cases and requests are quite extensive. |
| 03:48.17 | Samot | U of M had over 500 dial plans |
| 03:48.25 | Samot | For internal calling. |
| 03:48.55 | Samot | I don't mean Asterisk dialplan but what digit patterns where accepted and what they did...based on various items. |
| 03:49.32 | [TK]D-Fender | Yup, some people get real adhoc about this stuff... beyond any practical purpose usually |
| 03:49.40 | Samot | Well.. |
| 03:49.44 | [TK]D-Fender | "They're the customer" |
| 03:49.49 | [TK]D-Fender | #check |
| 03:49.53 | Samot | In other platforms Presence is it's own thing. |
| 03:50.09 | [TK]D-Fender | yup |
| 03:50.32 | [TK]D-Fender | There's a reason * is used lightly as only an application server in larger deployments |
| 03:50.47 | Samot | I agree. |
| 03:51.00 | Samot | U of M was on BroadSoft |
| 03:51.42 | [TK]D-Fender | I've never gotten around to learning full SIP solutions |
| 03:52.03 | Samot | Yeah, well I've only done provider level stuff. |
| 03:52.10 | [TK]D-Fender | I'm aware of the pieces in general but should get some actual practical experience at some point |
| 03:52.12 | Samot | I got to deal with a lot of solutions. |
| 03:52.22 | [TK]D-Fender | My skillset is a bit light & general |
| 03:52.43 | [TK]D-Fender | just finished an introductory class to vmware today |
| 03:52.45 | Samot | If I decided to not be self employed.. |
| 03:52.55 | Samot | I would have to look at places like Flowroute, etc for jobs. |
| 03:53.50 | Samot | VoIP/LEC/Hosting/DC |
| 03:54.57 | Samot | However, my network-fu is light.. |
| 03:55.30 | Samot | At the small places at least one own was a network engineer on some level.. |
| 03:55.39 | Samot | At the LEC, there was a team. |
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| 10:40.08 | GeneralSpongebob | Hi, after 'sudo service asterisk stop' then start again I get "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" when trying to connect with asterisk -r what log can I watch to see why? |
| 10:40.48 | GeneralSpongebob | /var/log/asterisk/messages and queue_log keep chowning to root despite me setting them to 'asterisk' |
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| 13:38.38 | wasanzy | hello |
| 13:40.36 | wasanzy | for an incoming call, I see this in cli SIP/dwg-0000000b, I want to check which SIP channel the call comes from and base on that, send the call to another context |
| 13:40.40 | wasanzy | how possible is that? |
| 13:41.36 | [TK]D-Fender | based on it how? |
| 13:42.06 | [sID] | Hi |
| 13:42.13 | [sID] | Question, I have a queue and added number out, the connection goes out of queue and how do I get status after disconnect and receive? |
| 13:42.26 | [TK]D-Fender | What status? |
| 13:43.28 | [sID] | dialstatus |
| 13:43.48 | [TK]D-Fender | You need to be clearer about exactly what you're trying to do |
| 13:44.34 | wasanzy | [TK]D-Fender: if the call is coming from SIP channel a, send it to [context] |
| 13:44.55 | [TK]D-Fender | Thre is no such thing as "SIP channel a" |
| 13:45.03 | [TK]D-Fender | Every call is its own channel |
| 13:45.16 | [TK]D-Fender | ever call matching a PEER goes to the context that peer is set to |
| 13:45.37 | [sID] | [TK]D-Fender: Wants to get queued status when the client disconnects or answers a call and retrieves the talk time. |
| 13:46.17 | wasanzy | so I have a SIMBOX connected to my asterisk server, it has like 8 SIM cards in it. all of them goes to one context, so under this context I want to do something like GotoIF(...) |
| 13:46.42 | [TK]D-Fender | [sID], Where how? You're not benig clear about WHERE you are loking to see this information, and in what form |
| 13:46.46 | [TK]D-Fender | being* |
| 13:47.16 | [TK]D-Fender | wasanzy, Does each channel auth as the same peer? |
| 13:47.55 | wasanzy | [TK]D-Fender: sorry please come again. auth as the same peer how? |
| 13:48.43 | wasanzy | I did sip show peers |
| 13:49.15 | wasanzy | dwg/dwg 0.0.0.0 D Auto (No) No 5060 Unmonitored |
| 13:49.26 | [TK]D-Fender | you have 8 ***SIM *** cards. From your very vague description it's sounding like you want to know which ***SIM*** it arrived on. Your gateway sendss the calls to * via SIP, no? Do calls from all 8 land on the same PEER? |
| 13:50.05 | wasanzy | [TK]D-Fender: yes all comes from the same peer |
| 13:50.25 | wasanzy | Should I pastebin sample call? |
| 13:50.36 | [TK]D-Fender | wasanzy, Then go look at 2 calls and see what is identifiably different between them |
| 13:50.49 | [TK]D-Fender | You should not have to ask if you should be looking at the calls. |
| 13:50.59 | [TK]D-Fender | The answer is "always". |
| 13:51.13 | wasanzy | s@from-PSTN:1] Answer("SIP/dwg-0000000b", "") |
| 13:51.29 | wasanzy | s@from-PSTN:1] Answer("SIP/dwg-0000000c", "") |
| 13:51.30 | [TK]D-Fender | PASTEBIN |
| 13:51.35 | wasanzy | that is two different call |
| 13:51.42 | [TK]D-Fender | SIP DEBUG <--------------- |
| 13:51.51 | wasanzy | ok |
| 13:52.16 | [TK]D-Fender | You asked if there was something in the call to identify and you AREN'T even looking at tthe actual call debug |
| 13:52.21 | [TK]D-Fender | You've coming here for YEARS |
| 13:52.59 | [TK]D-Fender | You should have a clue by now what "looking" actually means, especially when you just asked specifically about "sip call" |
| 13:53.26 | [TK]D-Fender | That device shouldn't be sending calls to "s" unless * registers to it and you didn't pass it an exten to target |
| 13:53.35 | [TK]D-Fender | how are you registering to it? |
| 13:53.36 | [sID] | [TK]D-Fender: How can I get dialstatus from the queue where there was an external connection |
| 13:54.10 | [TK]D-Fender | [sID], "external connection" means nothing. What EXACTLY are you trying to do? |
| 13:54.16 | wasanzy | [TK]D-Fender: it is registered to asterisk |
| 13:54.51 | [TK]D-Fender | wasanzy, it shouldn't be sending calls to "s". * doesn't accept calls randomly to that exten, it was targeted explicittly. What would it end up there? |
| 13:55.25 | [sID] | [TK]D-Fender: Member => Local/0048123456789@Trunk,0 |
| 13:55.50 | [TK]D-Fender | [sID], I'm expecting a proper description of exactly what you are expecting to process, where, and to what end |
| 13:56.04 | [TK]D-Fender | [sID], Things you made don't tell me what you want to DO. |
| 13:56.25 | [TK]D-Fender | I built a table... I needed a CHAIR. |
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| 13:57.16 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:58.13 | wasanzy | call 1: https://paste.linux.community/view/b7d0e36b |
| 13:58.23 | wasanzy | call 2:https://paste.linux.community/view/06a39e66 |
| 13:59.17 | *** join/#asterisk syadnom_ (~syadnom@host-72-174-219-84.bln-mt.client.bresnan.net) |
| 13:59.51 | [sID] | [TK]D-Fender: Wants to get dialstatus from queue. I have a test queue and member in it as I wrote above that is an external connection and now I want to retrieve dialstatus of this connection from the queue. |
| 14:00.18 | [TK]D-Fender | WHERE? |
| 14:00.19 | [TK]D-Fender | HOW? |
| 14:00.20 | syadnom_ | hi all, looking for some help configuring OPUS. I've got it running but core show translation paths opus only shpws opus:48000 |
| 14:00.29 | [sID] | in dialplan |
| 14:01.13 | [TK]D-Fender | Your local channel is it's own thing and uses the same hangup handling as any other channel |
| 14:01.28 | [TK]D-Fender | using either a coded handler or the typical "h" |
| 14:02.00 | [TK]D-Fender | Because if it's a bridged call dialplan doesn't normally continue after dial() terminates |
| 14:02.04 | [sID] | When a connection disconnects from this queue, it does not get status |
| 14:02.19 | [TK]D-Fender | there is no "get status |
| 14:02.22 | [TK]D-Fender | is the channel ALIVE? |
| 14:02.32 | [TK]D-Fender | No, it is clearly done processing its dialplan |
| 14:02.43 | [TK]D-Fender | so do the notmal dialplan things for it to continue processing |
| 14:03.08 | [TK]D-Fender | On the assumption that dial got answered |
| 14:03.25 | [sID] | [TK]D-Fender: |
| 14:03.26 | [sID] | http://wklej.to/J6tbm |
| 14:03.29 | [TK]D-Fender | And if it did then it should end up in "h" |
| 14:04.01 | [TK]D-Fender | Queue doesn't return dialstatus |
| 14:04.07 | [TK]D-Fender | You are inventing things |
| 14:04.15 | [TK]D-Fender | Everything queue calls is HIDDEN. |
| 14:04.22 | [TK]D-Fender | Queue does the dirty work. |
| 14:04.37 | [TK]D-Fender | "core show application queue" <- |
| 14:05.42 | wasanzy | [TK]D-Fender: I have looked but didn't see anything so different from the calls |
| 14:06.39 | [TK]D-Fender | and you called 2 different SIMs for those calls? |
| 14:06.40 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 14:06.56 | wasanzy | [TK]D-Fender: yes |
| 14:07.23 | [TK]D-Fender | wasanzy, then go look at reconfiguring your gateway. It makes the calls the same way for both. It has to be told to do something different |
| 14:07.48 | wasanzy | I will look at that |
| 14:07.55 | [sID] | [TK]D-Fender: Can not get status? |
| 14:08.01 | *** part/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
| 14:08.31 | [TK]D-Fender | <[TK]D-Fender> "core show application queue" <- |
| 14:09.20 | [TK]D-Fender | QUEUE calls those member devices you defined. YOU are not calling Dial(). It is using a Local Channel to make your call. that is an entirely different channnel contained when Queue calls the member. it does NOT get passed backwards |
| 14:09.32 | [TK]D-Fender | Queue() returns what it says it returns. |
| 14:10.40 | [sID] | [TK]D-Fender: Ehh does not understand unfortunately :( |
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| 14:11.24 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:11.29 | [TK]D-Fender | that dial happens in a LOCAL channel |
| 14:11.33 | [TK]D-Fender | that does NOT get passed back |
| 14:12.42 | [TK]D-Fender | What exact status are you expecting to get? The queue will tell you if it ANSWERED or not already. it's in Queue's instructions |
| 14:12.53 | [sID] | And can you pass it on? |
| 14:13.08 | [TK]D-Fender | NO DIALSTATUS |
| 14:13.10 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 14:13.15 | [TK]D-Fender | **************NO************** |
| 14:13.28 | [TK]D-Fender | That is in the LOCAL CHANNEL.NOTHING comes back from that |
| 14:13.36 | wasanzy | [TK]D-Fender: I see the SIMBOX can actually register each port on different SIP user ID |
| 14:14.21 | [TK]D-Fender | wasanzy, So if you want them identifiably different ... you should probably do that |
| 14:14.34 | [sID] | [TK]D-Fender: I need to know whether it was received or not and to download the time |
| 14:14.38 | wasanzy | doing that..... |
| 14:15.04 | [TK]D-Fender | [sID], There is already a QUEUE LOG |
| 14:15.32 | [TK]D-Fender | [sID], And if you want to know if it answered.... read Queue()'s INSTRUCTIONS |
| 14:15.38 | [TK]D-Fender | [TK]D-Fender> <[TK]D-Fender> "core show application queue" <- |
| 14:21.05 | [sID] | [TK]D-Fender: This option c: Continue in the dialplan if the callee hangs up. ? |
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| 14:22.38 | syadnom_ | I can't figure out why me opus8 entry in codecs.conf isn't usable, or my opus48 etc. |
| 14:23.07 | syadnom_ | I have opus48 in allow, an entry in codecs.conf, and xlite (which supports opus 48k) and it won't accept the call |
| 14:23.17 | syadnom_ | 'no compatible codec' |
| 14:24.25 | [TK]D-Fender | [sID], sounds like HALF a solution. Have you tried it? |
| 14:25.05 | [sID] | Yes, but I do not want to act. |
| 14:25.43 | [sID] | Queue(61666,RhHc,,,15) |
| 14:26.17 | *** join/#asterisk hehol (~hehol@gatekeeper.loca.net) |
| 14:26.53 | [TK]D-Fender | <[sID]> Yes, but I do not want to act. < huh? |
| 14:28.34 | [sID] | My English is not good :) |
| 14:28.46 | [sID] | act = does not work |
| 14:29.34 | [TK]D-Fender | <[sID]> [TK]D-Fender: This option c: Continue in the dialplan if the callee hangs up. ? <- that is if the CALLEE hangs up. Says nothing about the CALLER |
| 14:33.55 | [sID] | [TK]D-Fender: big H: |
| 14:36.46 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 14:37.56 | [sID] | [TK]D-Fender: OK, I did, thanks for the help |
| 14:39.01 | [TK]D-Fender | <PROTECTED> |
| 14:39.10 | [TK]D-Fender | this doe not say "continue" in any way |
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| 15:00.09 | [TK]D-Fender | [TK]D-Fender> As of January 1, 2017 the patent terms of most Licensed Patents under the G.729 Consortium have expired, the remaining unexpired patents are usable on a royalty-free basis.[7] Thus, G.729 can be used free-of-charge |
| 15:00.09 | [TK]D-Fender | <[TK]D-Fender> Does look like it should be... but all I see on Digium's page is the same paid licensing option |
| 15:00.49 | [TK]D-Fender | file, ^ |
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| 15:19.07 | file | has no comments |
| 15:20.10 | [TK]D-Fender | RFC's |
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| 15:45.09 | igcewieling | Real Men use hardware transcoding. 8-| |
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| 15:48.33 | [TK]D-Fender | Real men know if it will make a difference and not waste money if it isn't sensible :) |
| 15:49.23 | drmessano | Real men don't use servers from 2004 that were underpowered and required hardware transcoding |
| 15:49.47 | drmessano | Overhead is insanely cheap now |
| 15:50.03 | drmessano | It's also really hard to stick a card in a virtual machine |
| 15:50.31 | [TK]D-Fender | That's why I insist on VT-d or IOMMU :) |
| 15:50.53 | [TK]D-Fender | needs to get a server system for home and a new NAS |
| 15:51.33 | drmessano | You should get the new FREENAS, it will turn your whole house into an enterprise |
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| 15:52.25 | [TK]D-Fender | Except I'm looking touse Proxmox for my server and a hardware NAS |
| 15:52.41 | [TK]D-Fender | so PC+toaster for smaller footprint,and cost |
| 15:53.14 | drmessano | Ah. Because FREENAS can host all your VMs now and everything |
| 15:53.29 | [TK]D-Fender | FreeNAS is something I've kept an eye on for years, but have been debating where I would actually deploy it |
| 15:53.31 | drmessano | It's all the letters in the alphabet |
| 15:54.02 | [TK]D-Fender | it also has a larger RAM overhead, etc |
| 15:54.19 | [TK]D-Fender | And ties my storage to myy virtualiztion system |
| 15:54.27 | drmessano | I was disappointed to see they turned it into an everything-in-a-box distro |
| 15:54.46 | [TK]D-Fender | Well mid/higher end NAS' these days also support VM's |
| 15:54.52 | drmessano | Right |
| 15:54.58 | [TK]D-Fender | so I count it as "keeping up with the Jones' |
| 15:55.26 | drmessano | FreeNAS was just FreeNAS and it was good that it hadn't kept up |
| 15:55.41 | drmessano | Now it's enterprisey-ish-ness |
| 15:56.27 | [TK]D-Fender | "Prosumer" |
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| 16:02.20 | Samot | I can't do that here. |
| 16:02.41 | Samot | I don't have the space for extra stuff like servers any more in the house. |
| 16:03.17 | Samot | I just have to use external drives. |
| 16:04.23 | Samot | I just use WD MyCloud drives for network based storage for the house. |
| 16:05.05 | drmessano | Hopefully they've improved |
| 16:05.19 | Samot | For what I need them to do, they are fine. |
| 16:05.20 | drmessano | We deployed myCloud drives for a while and they all died |
| 16:05.25 | Samot | Haven't had an issue. |
| 16:05.35 | Samot | Been running for over two years. |
| 16:06.00 | drmessano | This was 7 or 8 years ago. I had considered getting one |
| 16:06.04 | Samot | And that includes being pulled off low height shelves by dogs getting tangled in cords... |
| 16:06.32 | Samot | Being put in boxes and shipped around.. |
| 16:06.35 | drmessano | Home server backs up to a 320 GB drive that I've had for about nine years |
| 16:06.44 | drmessano | Probably time to replace it |
| 16:06.48 | Samot | So far, it's taken a beating.. |
| 16:06.54 | [TK]D-Fender | I've got a pretty shitty D-Link DNS-320 w/ 2x2TB WD Green drives |
| 16:07.02 | Samot | Been in places where dust and dog hair are rampant.. |
| 16:07.08 | Samot | Still works like a charm. |
| 16:07.13 | [TK]D-Fender | The NAS cripples perfmance to like 22MMB/s |
| 16:07.48 | drmessano | Ouch |
| 16:13.04 | [TK]D-Fender | Sustained data transfer rate 144Mb/s |
| 16:13.15 | [TK]D-Fender | http://www.storagereview.com/seagate_barracuda_green_2tb_review_st2000dl003 |
| 16:13.20 | [TK]D-Fender | Wereseagate, not WD |
| 16:14.08 | [TK]D-Fender | but either way the bottleneck was the NAS, not the LAN (SSD Win>Win share tested at 100MB/s) or the drives |
| 16:19.16 | wasanzy | getting this error: check_auth: username mismatch, have <dwg>, digest has <zenith> |
| 16:20.26 | wasanzy | sip show peers: zenith/dwg 192.168.10.91 D Auto (No) No 5060 Unmonitored |
| 16:22.12 | drmessano | Pretty self explanatory |
| 16:22.18 | drmessano | What is the device? |
| 16:22.33 | wasanzy | drmessano: Simbox |
| 16:23.26 | wasanzy | I have configured two different sip accounts on the device, so I wonder why it is giving me that |
| 16:23.52 | [TK]D-Fender | because you're using PEER and it matches on IP |
| 16:24.08 | [TK]D-Fender | and those different accounts are calling in from the same IP |
| 16:24.10 | wasanzy | I am using friend not peer |
| 16:25.27 | wasanzy | https://paste.linux.community/view/b05fcc56 |
| 16:25.40 | wasanzy | that is my sip.conf for that |
| 16:27.10 | [TK]D-Fender | show full debug |
| 16:28.27 | wasanzy | https://paste.linux.community/view/4eac49a8 |
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| 16:31.19 | [TK]D-Fender | I can't tell if those sectrets aredifferent or what you set in the gateway |
| 16:31.26 | [TK]D-Fender | the FRON isnot shoing a forced username either |
| 16:31.49 | wasanzy | [TK]D-Fender: the secretes are different |
| 16:32.12 | wasanzy | forced username? how do I force it? |
| 16:32.28 | *** part/#asterisk igcewieling (~ewieling@ip98-170-215-112.pn.at.cox.net) |
| 16:33.46 | [TK]D-Fender | This is your gateway... |
| 16:33.53 | [TK]D-Fender | Time to pay more attenntionto it |
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| 17:36.59 | igcewieling | Verizon's idiocy is mindboggling. |
| 17:38.46 | igcewieling | 'Again, we are very sorry to hear that you did not receive any notification of the maintenance. As per our internal group, per the database for maintenance notification, the company name for circuit ID: U199928, only shows "UNKNOWN."Â ' We've had the circuit for at least 8 years and run 2/3 of our voice traffic over it. |
| 17:39.01 | igcewieling | fscking morons. |
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| 20:19.55 | K0HAX | Is AES_CM_128_HMAC_SHA1_80 not supported for SRTP anymore? |
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| 20:29.40 | drmessano | K0HAX: What leads to you ask that question? |
| 20:29.50 | drmessano | K0HAX: What leads you to ask that question? |
| 20:30.17 | K0HAX | drmessano: I'm trying to use SRTP with a Polycom VVX400 and that's the only algorithm that's trying to negotiate |
| 20:30.28 | K0HAX | I get this error: [2017-03-27 15:06:39] WARNING[30280]: res_srtp.c:734 crypto_activate: Could not set SRTP policies |
| 20:30.57 | Samot | How did you set this up? |
| 20:31.09 | Samot | Because I have no issues with VVXs and SRTP. |
| 20:31.26 | K0HAX | I'm using pjsip |
| 20:31.34 | Samot | OK. |
| 20:31.48 | Samot | So you're doing this over TLS |
| 20:31.50 | K0HAX | I compiled pjproject 2.6 and asterisk 14.3.0 on CentOS 7 following the guide |
| 20:31.52 | K0HAX | yes |
| 20:31.59 | Samot | And have a proper cert? |
| 20:31.59 | K0HAX | I'm using Homer to see the SIP packets |
| 20:32.20 | K0HAX | I have a self signed cert that's trusted by the phone, I am able to place calls if I disable SRTP on the phone side |
| 20:32.51 | Samot | Show call that fails with SRTP |
| 20:32.54 | K0HAX | I'm using media_encryption_optimistic = yes and media_encryption = sdes in my endpoint definition |
| 20:32.54 | Samot | asterisk -rvvvvvvvvv |
| 20:32.58 | Samot | pjsip set logger on |
| 20:33.00 | Samot | ~pb |
| 20:33.00 | infobot | somebody said pastebin was a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:34.38 | K0HAX | https://paste.k0hax.com/index.php?id=58d9775387c85 |
| 20:35.25 | Samot | OK, show the pjsip peer for this device. |
| 20:36.43 | K0HAX | https://paste.k0hax.com/index.php?id=58d977b445a58 |
| 20:36.47 | K0HAX | it does work with Zoiper btw |
| 20:37.59 | Samot | Sorry, should have specified more. |
| 20:38.03 | Samot | Show the peer config... |
| 20:38.07 | K0HAX | oh! |
| 20:38.47 | K0HAX | https://paste.k0hax.com/index.php?id=58d9785197e24 |
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| 20:41.18 | Samot | Show the Zoiper call, let's see the differences. |
| 20:41.22 | K0HAX | ok |
| 20:42.51 | K0HAX | https://paste.k0hax.com/index.php?id=58d97948afff3 |
| 20:45.49 | Samot | [2017-03-27 15:41:37] WARNING[30280]: res_srtp.c:734 crypto_activate: Could not set SRTP policies |
| 20:46.00 | Samot | I'm seeing the same error in the call with zoiper. |
| 20:46.11 | Samot | Show the transports section for pjsip. |
| 20:46.21 | Samot | That has the SSL stuff in it.. |
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| 20:47.36 | K0HAX | https://paste.k0hax.com/index.php?id=58d97a64320fe |
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| 21:12.40 | K0HAX | I feel like I should just blame the OpenSSL project for disabling random cipher suites... |
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| 21:14.46 | *** join/#asterisk ghoti (~paul@hq.experiencepoint.com) |
| 21:22.29 | K0HAX | It was the polycom config.. |
| 21:22.39 | K0HAX | I don't know what I changed, but it works now! lol |
| 21:24.57 | K0HAX | oh great, now it's intermittent |
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| 21:35.06 | K0HAX | I figured it out.. it's nat. :( |
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