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01:56.29 | wonderworld | is Opus for Asterisk 13 available yet? http://blogs.digium.com/2016/09/30/opus-in-asterisk/ |
01:58.15 | file | in latest version, yes |
02:01.36 | wonderworld | i run 13.13.1. is it a seperate download or should it be available already after building from source? |
02:02.22 | file | you have to select it in "make menuselect" under Codecs to have it download, and that option requires the "xmlstarlet" utility be installed to use |
02:02.39 | wonderworld | tnx let me check |
02:04.37 | wonderworld | ok i didn't compile it XXX codec_opus |
02:05.52 | wonderworld | what dazzeled me was that asterisk announces to be able to handle OPUS via SDP/webrtc |
02:06.13 | wonderworld | but it didn't work. asterisk always fell back to ulaw |
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02:14.49 | wonderworld | where would i get xmlstartlet? |
02:37.58 | wonderworld | i manualy installed opus. now asterisk crashes in the middle of loading the module. https://ghostbin.com/paste/66kqz |
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03:35.03 | wyoung | @file is a perl thingy isn't it? |
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04:27.14 | Sean-Der | Does anyone know how to turn off Loop Detection, for some reason certain incoming calls from a SIP trunk are getting a 482 (Loop Detected) |
04:27.21 | Sean-Der | I can anonymize the inbound SIP if it helps |
04:27.55 | Sean-Der | but I can't tear down and debug my running Asterisk instance, because I can't reliabily reproduce this just sometimes I can get the box in a broken state |
04:28.18 | Sean-Der | I am running 13.13.1 |
04:28.50 | Samot | So they are sending you a call? |
04:28.57 | Samot | Or are you sending them a call? |
04:29.09 | Sean-Der | They are sending me a call, and I respond to the INVITE with a 482 |
04:29.51 | Sean-Der | The one thing that is broken on the other end is they are sending me the same call twice (on the backup trunk that is only supposed to be used in failure) |
04:29.59 | Sean-Der | like I am not responding to the first INVITE fast enough |
04:30.47 | Samot | Why are you responding with a 482? |
04:30.54 | Samot | That shouldn't be a reply to an INVITE |
04:30.59 | Samot | It should be a 1xx reply. |
04:31.04 | Samot | At least. |
04:32.05 | Sean-Der | I have no idea, Asterisk is doing it for me. I have no logic in my dialplan (or sip config) relating to this |
04:32.05 | Samot | Show one of these calls.. |
04:32.13 | Samot | asterisk -r |
04:32.17 | Samot | sip set debug on |
04:32.23 | Samot | or pjsip set logger on |
04:32.27 | Samot | Depending on your tech.. |
04:32.29 | Samot | Make a call |
04:32.31 | Samot | ~pb |
04:32.31 | infobot | well, 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. |
04:32.38 | Samot | Show the SIP debug.. |
04:32.47 | Samot | Want to see the INVITE and the 482 reply to it. |
04:32.56 | Sean-Der | Yep! |
04:35.54 | Sean-Der | Samot: mind if I PM you the paste (has some numbers and IPs) not a big deal if not |
04:35.59 | Sean-Der | and we can keep chat in here |
04:36.36 | Samot | ok |
04:38.06 | Sean-Der | I don't know what Asterisk uses to throw a Loop Detected, but gonna grep around and see if I can figure it out |
04:38.30 | Samot | Why are they sending you calls on a 10.10.x.x network? |
04:38.38 | Samot | Do you have a private LAN connection with them? |
04:39.23 | Sean-Der | It is a DMZ, the calls the majority of the time |
04:40.31 | drmessano | O.o |
04:40.43 | Sean-Der | maybe not the right term, but the interface is a 10.10, but it is a world routable box |
04:40.54 | Sean-Der | sorry the calls work the majority of the time |
04:41.04 | Sean-Der | and this call would work, if Asterisk wouldn't throw the 482 |
04:42.02 | Samot | 10.10 is not world routable. |
04:42.28 | Samot | 10.x.x.x is a completely private /8 |
04:42.40 | Sean-Der | 10.10 is the local interface, but it is setup so that all traffic to a world routable address forwards to it |
04:42.48 | Samot | OK. |
04:42.55 | Samot | So it's NAT'd. |
04:44.55 | Sean-Der | yes |
04:45.23 | Samot | Is this just coming straight into your Asterisk box? |
04:46.03 | Samot | Provider --> Asterisk |
04:46.10 | Sean-Der | yep |
04:46.19 | Samot | Which line in the debug is your Asterisk IP |
04:46.27 | Samot | Tell me the first line it appears. |
04:47.36 | drmessano | Its a secret |
04:47.40 | Samot | And not the INVITE with the 10.10. |
04:47.46 | Samot | No, he sent me the debug. |
04:48.05 | Samot | I have no issue with him not wanting information logged in botbot.. |
04:48.38 | Samot | And it's not sanitised, so yay! |
04:49.10 | Samot | I think I can see why Asterisk is replying with a 482 but need to confirm. |
04:49.23 | Sean-Der | I actually don't see it, I checked a couple times |
04:49.34 | Samot | It has to be there. |
04:49.43 | Samot | The WAN IP. |
04:49.48 | Samot | The external IP for it. |
04:49.51 | Samot | Not the local. |
04:50.06 | Sean-Der | That is the WAN IP |
04:50.07 | Samot | K |
04:50.32 | Sean-Der | I guess I don't understand how this was ever working, if I restart Asterisk it works for a while |
04:50.42 | Sean-Der | but eventually gets into a bad state |
04:51.42 | Samot | What is the IP in the TO header? |
04:51.47 | Samot | Is that yours? |
04:51.52 | Samot | Because if it's not... |
04:52.03 | Samot | This is a poorly routed SIP message. |
04:52.13 | Samot | And that's why Asterisk is seeing a loop. |
04:52.47 | Samot | Max-Forwards: 67 <-- You see this? |
04:53.03 | Sean-Der | yep! default is 70, only thing that rings a bell |
04:53.09 | Samot | OK |
04:53.21 | Samot | It goes down by one for every route in the path. |
04:53.31 | Samot | So it it was a straight route.. |
04:53.36 | Samot | It would always be 69 |
04:53.38 | Samot | DUDE |
04:53.51 | Samot | This took two extra routes. |
04:54.36 | Samot | And the TO URI is not your IP |
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04:54.46 | Samot | It's one of theirs in the Via headers. |
04:54.57 | Sean-Der | So this is out of my control? |
04:55.25 | Sean-Der | Looking through calls that worked it has always been this way |
04:55.32 | Sean-Der | So somehow this was working in Asterisk 1.8 |
04:55.41 | Samot | Well.. |
04:55.43 | Sean-Der | but I am taking 1.8 -> 13 boxes, or trying at least |
04:55.47 | Samot | That would have been a bug |
04:55.49 | Samot | Or.. |
04:55.57 | Samot | Things have changed in the last 5 years. |
04:56.14 | Samot | Think of it this way... |
04:56.16 | Sean-Der | I really should have upgrade sooner, thank you very much for your help Samot |
04:57.02 | Samot | Originates at PSTN -> Routes to A -> A routes to B -> B routes to C -> C routes to A |
04:57.14 | Samot | Loopish. |
04:57.19 | Sean-Der | I am gonna file a ticket with the SIP provider tomorrow, they don't have much in the way of debug and all I really do is write C/Dialplan |
04:57.32 | Samot | Your RR headers and Via headers look loopish.. |
04:57.53 | Samot | It may not pick it up right away... |
04:58.11 | Sean-Der | So I am pretty lost when it comes to anything besides the basics |
04:58.35 | Samot | SIP in generally has improved a lot over the years... |
04:58.49 | Samot | So things that worked 10 years ago... |
04:59.06 | Samot | Might not work now because of improvements and better ways. |
04:59.27 | Sean-Der | ok so I restarted the server, and it is working |
04:59.37 | Sean-Der | Do you mind if I dump pcap again, and compare maybe? |
04:59.37 | Samot | Show another call. |
04:59.41 | Sean-Der | ACK |
04:59.48 | Samot | I want to see the difference. |
05:08.38 | Sean-Der | Samot: so it looks like it still is malformed, but this call worked |
05:08.54 | Sean-Der | sorry for reason ZNC keeps timing out, I think my VPS is having issues |
05:08.57 | Samot | One sec. |
05:09.53 | Samot | Yes but a completely different set of routes completely. |
05:10.31 | Samot | And handled differently. |
05:11.16 | Samot | In the one the failed, the first record-route has the standard lr tag. |
05:11.21 | Samot | Which can be ignored.. |
05:11.55 | Samot | The working call, not only does it come from a different route. It has the lr=on tag which forces lr to be respected. |
05:12.36 | Samot | As well the second record-route on the working call, while it has just a normal lr tag (like the non-working one)... |
05:12.48 | Samot | It has a ftag attached... |
05:13.37 | Samot | This routing path is more sane and has better transaction and dialog management it looks like. |
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05:14.18 | Samot | I'm speculating on dialog management since I can't see the dialog happening in this. |
05:14.37 | Samot | But the transaction is being handled better. |
05:16.16 | Sean-Der | I am sorta stuck, because that is stuff I can't change |
05:16.37 | Sean-Der | I can make a ticket and send data to them |
05:17.06 | Samot | Yes. |
05:17.12 | Samot | Show them both. |
05:17.28 | Sean-Der | will do, and will try my best to relay your knowledge |
05:17.36 | Samot | They can make it so your route isn't load balanced or whatever they are doing. |
05:17.39 | Sean-Der | through the sieve of my cluelessnes |
05:17.54 | Samot | You don't need to say all I said.. |
05:18.12 | Samot | Just say "This way fails, this way doesn't. Please always route us to the way that doesn't." |
05:18.22 | Samot | If you want, make a few calls in a row. |
05:18.34 | Samot | Give them, here's X that failed and Y that didn't. |
05:18.41 | Samot | Notice X is always the same route. |
05:18.51 | Samot | Y is always the same route. Please always make Y |
05:18.54 | Sean-Der | will do, thank you very much for the help again that was super helpful |
05:19.22 | Samot | These are Sonus boxes too.. |
05:19.45 | Sean-Der | I saw people complaining about them online |
05:19.47 | Sean-Der | generally a pain? |
05:19.59 | Samot | They are expensive. |
05:20.30 | Samot | Who's your provider, if you don't mind.. |
05:20.57 | Sean-Der | don't mind! not closely tied to them https://www.nexvortex.com/ |
05:21.26 | Sean-Der | I didn't chose them and honestly has worked pretty well (up until now) |
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05:21.41 | Sean-Der | been with them since 2011 and haven't had issue 1.6 and 1.8 boxes |
05:21.44 | Samot | There are two types of companies that use Sonus SBCs... |
05:22.37 | Samot | Real carriers/providers and those that think they can be one because they had the money to spend on a really shiny box. |
05:23.25 | Samot | It don't matter you driving a Porsche when all you do is grind the clutch.... |
05:24.03 | Sean-Der | heh |
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05:24.47 | Sean-Der | well let's hope that tomorrow I send them these captures and they can just fix it |
05:25.04 | Samot | Or you can be like drmessano, who keeps it 100, by driving around in a stolen Cadillac playing some Skynard 8-tracks. |
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05:26.03 | drmessano | 4 on the floor too |
05:26.05 | Samot | He'll tell Neil straight up, no southern man want him around anyhow. |
05:26.43 | Samot | And a GUN RACK |
05:26.59 | drmessano | Ah its definitely a new day |
05:27.12 | Samot | "I don't even have A gun, let alone many that would facilitate a gun rack" |
05:27.21 | drmessano | Around 23:00 I said I had my fill of idiots |
05:27.39 | drmessano | So a buddy of mine posts asking for a recommendation for battery repair |
05:27.44 | drmessano | for an iphone 5s |
05:28.04 | drmessano | Some says |
05:28.13 | drmessano | "Try T-Mobile or Apple or online" |
05:28.25 | drmessano | Try shutting the fsck up |
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05:30.39 | drmessano | For $60 I may get my 5s done |
05:34.27 | wyoung | Samot: Doesn't sound like a real southerner then |
05:35.37 | Samot | No. |
05:35.48 | Samot | But I was quoting a classic movie. |
05:35.52 | wyoung | right |
05:35.57 | wyoung | Never heard of it :P |
05:36.23 | wyoung | Do they use asterisk in the movie? |
05:37.35 | Samot | No. |
05:37.47 | Samot | Because, 1992. |
05:56.51 | drmessano | So back when Asterisk was in beta |
05:59.59 | Samot | I though ti was like 98/99 when it was first released... |
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06:02.50 | Samot | When did Kelly Ann go to the Obama pronunciation of ISIS? |
06:05.22 | drmessano | It was late 99, I think |
06:05.57 | drmessano | KAC is such a train wreck |
06:06.11 | drmessano | Worst part is she's not even as amusing as Spicer |
06:06.50 | drmessano | He should definitely be invited to Astricon |
06:08.22 | Samot | At least he took SNL in stride. |
06:20.09 | drmessano | Yeah |
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07:43.16 | tparcina | Good morning. |
07:43.48 | tparcina | Samot: What makes you think it is an network issue? |
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08:42.56 | RomanDcoz | Hello Guys, when enable TCP protocol i am getting below error, Please suggest me. |
08:42.57 | RomanDcoz | http://pastebin.com/h1SbyuHa |
08:44.08 | RomanDcoz | error like : [Feb 10 13:24:30] ERROR[20366]: tcptls.c:908 ast_tcptls_client_start: Unable to connect SIP socket to 178.Provider.IP:5060: Connection refused |
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09:03.19 | juned | Hi all, is it possible to add IP range for peer ? I want receive incoming calls from IP range instead of single IP. |
09:04.06 | juned | [provider_1_trunk] |
09:04.06 | juned | type=102.254.233.23/24 |
09:04.06 | juned | context=provider_1_incoming |
09:04.15 | juned | something like this |
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09:21.01 | bounceman | Howdy, I use the AGI application after hangup that does some calculatings then update the cdr row in mysql based on the uniqueid. The thing is, my AGI is performed before the actual CDR is written to the database meaning that no uniqueid exists. Is it possible to force asterisk to write the CDR in the dialplan? |
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09:35.00 | bounceman | Alternative is that I call my python file that in turn calls another python file with a sleep, which for me is a bad solution |
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10:05.37 | dokma | My MOH musci sounds hideous. Apparently it depends also on the format of the files so I was wondering which format is best for * ?? |
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11:42.27 | Samot | tparcina: Because no audio/one way audio issues area always network issues. |
11:43.50 | Samot | juned: Not with Chan_SIP |
11:44.02 | tparcina | Samot: I have look at captured network traffic, and I have managed to narrow down the problem. |
11:44.31 | tparcina | Samot: A is behind my VoIP provider. |
11:44.56 | Samot | Right. |
11:45.03 | juned | Hi Samot, can we do that using pjsip ? |
11:45.05 | Samot | RomanDcoz: Via: SIP/2.0/TCP 202.MY.asterisk.IP:5060;branch=z9hG4bK238009e6;rport |
11:45.21 | tparcina | And the problem is with RTP stream that * is sending to my ISP (to A), after the transfer. |
11:45.24 | Samot | ^^^ Does your provider support TCP transport. |
11:45.33 | tparcina | Samot: Then new RTP stram starts. |
11:45.35 | Samot | juned: yes. |
11:45.43 | juned | Thanks Samot... |
11:45.52 | Samot | Yes. |
11:45.58 | Samot | Of course, it's going to change. |
11:46.01 | tparcina | Samot: It goes from same address:port to the same address:port as previous RTP stream. |
11:46.11 | Samot | Uh? |
11:46.12 | Samot | No. |
11:46.17 | RomanDcoz | Samot: dont know, i think need to ask provider for this |
11:46.23 | Samot | The RTP ports between the provider and the PBX will NOT change. |
11:46.29 | Samot | That channel is never changed. |
11:46.32 | Samot | It's always the same. |
11:47.10 | Samot | Provider to PBX = 1 channel |
11:47.11 | tparcina | Samot: Just I have noticed on this one case that when there was the problem the used RTP codec has changed. |
11:47.14 | Samot | That's it. |
11:47.26 | Samot | That channel is all that exists between Asterisk and the Provider.. |
11:47.39 | Samot | Those RTP ports are established when the call comes in. |
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11:47.49 | tparcina | Samot: Now I'm trying to find more failed transfer examples to see do they all have codec change. |
11:48.14 | tparcina | Samot: I don't know, we are using UDP with our provider. |
11:48.24 | Samot | That TCP comment wasn't for you. |
11:49.06 | tparcina | ok :) |
11:53.24 | RomanDcoz | is this correct way to set cdrs value using socket fputs($oSocket, "SetCDRUserField: 123456789\r\n"); ? |
11:56.40 | RomanDcoz | I need to set value like custom in any cdrs field , I am using cdr_mysql module to insert value directly |
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12:06.49 | MaliutaLap | dokma: the format that matches the codec you're using |
12:07.29 | MaliutaLap | dokma: also you can't just use any music, most codecs have limited frequency ranges - so choose wisely |
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13:16.59 | apb1963 | http://pastebin.com/sYQrueAD My softphone appears to register, but I'm getting an authentic failure/invalid password when I try to make a call. That led me to check peers and it tells me status on the ext is "unreachable" - even though it seems to register. No idea what's going on so there's extra debug in the paste. |
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13:51.31 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
13:55.37 | Samot | INVITE sip:1-703-376-3246@192.168.0.102 SIP/2.0 <-- WTF? |
13:56.37 | apb1963 | ? |
14:01.46 | Samot | Why are there dashing in the INVITE |
14:02.02 | apb1963 | why not? |
14:02.22 | Samot | And how are you matching or stripping them? |
14:02.34 | apb1963 | I'm not |
14:02.45 | apb1963 | I'm not doing anything special |
14:02.56 | [TK]D-Fender | I'm not seeing an answer to the 401 challenges |
14:04.48 | apb1963 | Anybody got a D-Fender to English translator handy? |
14:05.12 | [TK]D-Fender | you send the invite, * sends a 401 CHALLENGING them, and Jitsi isn't coming back with papers |
14:05.22 | apb1963 | ok so it's a jitsi issue |
14:05.39 | apb1963 | but.. not configuration right? Bug yes? |
14:06.43 | Samot | Yeah, that's what I'm seeing to. |
14:07.05 | Samot | I'm also seeing a bad INVITE structure.. |
14:07.32 | Samot | All a million codecs and other crap that really don't need to be involved but hey.. |
14:07.44 | Samot | INVITE sip:1-703-376-3246@192.168.0.102 SIP/2.0 <-- This will NOT work. |
14:08.09 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-veoptmlcioumhovw) |
14:08.09 | *** mode/#asterisk [+o newtonr] by ChanServ |
14:08.09 | apb1963 | Why not? |
14:08.13 | Samot | 1) You have no way to match that and strip the - |
14:08.19 | Samot | 2) No carrier accepts that format. |
14:08.38 | Samot | The dashes are human readable things. |
14:08.55 | Samot | They are not part of any routing of a DID. |
14:10.16 | *** join/#asterisk GreenCult (b541d67c@gateway/web/freenode/ip.181.65.214.124) |
14:11.00 | apb1963 | well, it's never been a problem prior to TLS |
14:12.26 | Samot | So sending a call like that from Jitsi works on plain ole UDP? |
14:12.30 | Samot | Non-TLS? |
14:13.50 | [TK]D-Fender | <apb1963> but.. not configuration right? Bug yes? <- configuration could be an issue |
14:14.58 | apb1963 | well, actually there is a bug.. TLS doesn't work at all unless I set the proxy to the same machine; otherwise it doesn't pick up 5061 as the port. So.. yeah, there's that. |
14:15.06 | apb1963 | Samot, yes |
14:16.01 | apb1963 | [TK]D-Fender, and by setting the proxy to the same machine I should also point out that I don't normally use a proxy. |
14:16.43 | Samot | So you have a pattern match of _1-NXX-NXX-XXXX? |
14:17.42 | apb1963 | Samot, Looks somewhat familiar what's automatically set for me in the gooey. |
14:17.52 | apb1963 | ^to |
14:19.29 | Samot | And you're sending it like that to the provider?! |
14:19.35 | apb1963 | yummy yummy gooey gooey rich and chewy inside out |
14:19.50 | apb1963 | whatever you see in the log |
14:20.00 | Samot | I see a failed call. |
14:20.05 | Samot | I see a call that never makes it to dialplan. |
14:20.19 | Samot | So no, I can't see how it was accepted by the provider in that format. |
14:20.32 | Samot | Nor do I know of ANY provider that would accept that format. |
14:21.07 | apb1963 | I"m assuming at this point that it doesn't make it to the dialplan because it doesn't answer the 401 challenge. |
14:21.13 | Samot | Yes. |
14:21.48 | Samot | So why don't you show us a non-TLS call that works.. |
14:21.50 | apb1963 | so.. once jitsi fixes that issue, i'll revisit that issue |
14:22.14 | Samot | That's kind of a show stopper bug. |
14:22.21 | Samot | Not responding to 401 challenges. |
14:22.31 | Samot | Make registering mighty hard. |
14:23.19 | [TK]D-Fender | <apb1963> so.. once jitsi fixes that issue, i'll revisit that issue <- so far nothing said it was a BUG on their part |
14:23.30 | [TK]D-Fender | <[TK]D-Fender> <apb1963> but.. not configuration right? Bug yes? <- configuration could be an issue |
14:26.06 | wonderworld | which libsrtp version is recommended for use with asterisk? |
14:26.16 | Samot | I want to see proof that backs up the claim that it works non-TLS and that call actually makes it to carrier. |
14:26.18 | apb1963 | Well, I can't make it work at all unless I set the proxy to the same machine, it simply ignores the port setting and uses 5061 regardless. So, that's a bug. |
14:26.34 | apb1963 | s/5061/5060 |
14:26.44 | Samot | No, that's a cofiguration issue. |
14:26.49 | apb1963 | i.e. the wrong port. |
14:26.51 | apb1963 | No, that's a bug. |
14:27.00 | Samot | Where do you put the port in the host name? |
14:27.13 | apb1963 | ? |
14:27.27 | Samot | Well there's generally two options for hosts. |
14:27.33 | apb1963 | jitsi has a configuration window. It allows me to specify a port. |
14:27.34 | Samot | The Registration/SIP Proxy host |
14:27.41 | Samot | Or the Outbound Proxy Host |
14:27.57 | apb1963 | Both |
14:28.15 | Samot | OB proxy is never required. |
14:28.18 | Samot | Or shouldn't be. |
14:28.25 | apb1963 | With UDP, it's not. |
14:28.59 | Samot | With TLS it should be either. |
14:29.05 | apb1963 | Hence the bug |
14:29.35 | apb1963 | Using the proxy settings forces it to use the right port. |
14:29.57 | apb1963 | It's a workaround.. that doesn't really work because jitsi doesn't respond properly we now know. |
14:30.10 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
14:30.16 | Samot | Do you have the right port in the main SIP proxy? |
14:30.21 | Samot | :5061? |
14:30.23 | apb1963 | Yes |
14:32.47 | [TK]D-Fender | Just show all of it |
14:32.58 | apb1963 | http://picpaste.com/pics/Screenshot_from_2017-02-10_06-30-52-WsA4kBzA.1486737153.png |
14:33.01 | [TK]D-Fender | We shouldn't be wasting time doing this blind |
14:33.28 | [TK]D-Fender | And ditch that keepalive |
14:33.36 | [TK]D-Fender | it's flooding pointlessly |
14:34.10 | apb1963 | None, Options, Register, CRLF are my options |
14:34.21 | [TK]D-Fender | DITCH <- |
14:34.25 | apb1963 | None |
14:34.42 | apb1963 | I was wondering about that. Thanks |
14:35.40 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
14:35.53 | *** part/#asterisk juned (~juned@202.131.119.122) |
14:37.03 | apb1963 | summary window: http://picpaste.com/pics/Screenshot_from_2017-02-10_06-35-13-1k0yakSn.1486737387.png |
14:38.09 | wonderworld | whats the thing with Opus support? is it ready to be used in a project or might the patents render it useless soon? The "data collection feature" made me feel not too good about it, sounds marvelous though... |
14:38.13 | *** join/#asterisk jjrh (~weechat12@2607:f0b0:8:8035:796f:592e:3a21:a6eb) |
14:38.42 | EmleyMoor | I have discovered an undesireable behaviour when someone repeatedly calls my fax number, which is routed through Asterisk, so as to hit a "busy" modem. The call is rejected as "busy" which causes my ITSP to route it to my mobile. I want to be able to make the call either hold or at least be processed for having ID (and not a definitely invalid one) before passing it out of the dialplan. Is there any ea |
14:38.48 | EmleyMoor | sy way I can implement this? (I have ... |
14:38.51 | EmleyMoor | ... processing for presence/validity of caller ID on my dialplan already, just not passed through for fax. |
14:38.54 | EmleyMoor | ) |
14:39.51 | Samot | Where does the fax number go? |
14:40.07 | Samot | Does it route to an endpoint like an ATA? |
14:40.17 | [TK]D-Fender | IIRC you should not be looking at ZRTP if you were looking for SRTP |
14:40.22 | WIMPy | EmleyMoor: Experiment with different cause codes in Hangup(). |
14:40.56 | EmleyMoor | Samot: To an iaxwodem |
14:40.59 | EmleyMoor | modem* |
14:41.04 | Samot | OK |
14:41.12 | Samot | So they are repeatedly calling? |
14:41.16 | EmleyMoor | WIMPy: Ah... good point... |
14:41.24 | Samot | Like one right after the other? |
14:41.38 | WIMPy | EmleyMoor: Why is a software modem busy? |
14:41.40 | EmleyMoor | Samot: In this case, that is what happened (fake Danish number, tech support scam call) |
14:41.53 | Samot | But they are making multiple calls at once? |
14:41.57 | EmleyMoor | WIMPy: Because it's already dealing with the first call |
14:42.08 | Samot | Are they making enough calls to trigger the modem to throw a busy?! |
14:42.08 | EmleyMoor | Samot: Yes |
14:42.12 | WIMPy | Why do you only have one running? |
14:42.34 | Samot | So everything is acting as it should. |
14:42.58 | Samot | 1) you cannnot take a fax call and put it in the queue |
14:43.15 | EmleyMoor | I could add more, perhaps... but I actually very rarely receive calls genuinely for it anyway |
14:43.24 | WIMPy | Why not? |
14:43.36 | Samot | Put a fax call in the queue? |
14:43.48 | WIMPy | yes |
14:43.55 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
14:43.59 | Samot | Uhm. |
14:44.11 | Samot | Negotiation... |
14:44.15 | Samot | When answered. |
14:44.29 | Samot | A sending fax machine is expecting tones.. |
14:44.34 | WIMPy | You don't have to answer to use Queue. |
14:44.39 | Samot | OK. |
14:44.43 | Samot | Sure. |
14:44.52 | Samot | OR you can just handle your calls correctly. |
14:44.58 | Samot | I'm a BIG fan of that. |
14:45.14 | WIMPy | Why wouldn't that be correctly? |
14:45.44 | Samot | Well he has a failover to his cell phone triggered when the FAX line is busy. |
14:45.53 | Samot | I'm not sure that's the best way to handle a fax call failover. |
14:45.55 | WIMPy | Maybe it doesn't give yu enough time, but it surely increases the chances to receive that fax earlier. |
14:46.04 | Samot | Unless he's going to interrupt the tones and shit out a fax |
14:46.18 | [TK]D-Fender | <EmleyMoor> WIMPy: Because it's already dealing with the first call <- make multiple iaxmodems |
14:46.34 | WIMPy | No, but that's a question of what kind of configuration options the ITSP offers. |
14:46.35 | EmleyMoor | Hmmm... good point - may be able to disentangle this at the ITSP end |
14:46.35 | [TK]D-Fender | EmleyMoor, Of just answer the call and ditch if you don't want it to remain unanswered |
14:47.38 | Samot | 1) Why are you sending busy FAX calls to a cell phone as the failover/CFWD Busy option??! |
14:47.59 | Samot | 2) Why don't you block the call that keeps flooding you? |
14:48.23 | Samot | 3) If you need to answer more faxes, do what TK said. Make more modems. |
14:48.58 | EmleyMoor | Samot: 1) It's a "last resort" option - and the number is/was grouped under the same group as my voice numbers 2) Because until it does I can't tell it's going to 3) If a genuine need arose I would |
14:49.04 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
14:49.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:49.50 | Samot | So you never get flooded from the same "fake" source twice? |
14:53.38 | EmleyMoor | Samot: No. |
14:53.51 | *** join/#asterisk u0m3 (~u0m3@188.25.22.193) |
14:54.07 | EmleyMoor | Anyway, I've regrouped my numbers... so that should resolve it |
15:00.07 | apb1963 | http://pastebin.com/xQKynYdV Fresh call after turning off OPTIONS. There are some very.. odd things going on. Search for "odd". |
15:01.53 | *** join/#asterisk Rini (uid196547@gateway/web/irccloud.com/x-vqhogijukwcdnpde) |
15:02.09 | *** join/#asterisk kharwell (kharwell@nat/digium/x-alxmgtuwqujppomb) |
15:02.09 | *** mode/#asterisk [+o kharwell] by ChanServ |
15:03.34 | *** join/#asterisk samwiere_ (~samwierem@095-097-255-066.static.chello.nl) |
15:03.41 | Samot | <--- Reliably Transmitting (NAT) to 192.168.0.11:58840 ---> <-- ARe you really behind NAT on the same network? |
15:04.50 | Samot | Call-ID: bb93e88c37069908664f2ab4677e8939@0:0:0:0:0:0:0:0 <--- I would say that's a bad Call-ID |
15:06.32 | [TK]D-Fender | SIP/2.0 488 Not acceptable here |
15:06.58 | [TK]D-Fender | [2017-02-10 06:53:21] DEBUG[25969][C-00000039] chan_sip.c: No compatible codecs for this SIP call. |
15:08.52 | Samot | That is.....impressive. |
15:09.14 | Samot | Consider the fact he offers just about every codec known to man from Jitsi. |
15:09.17 | apb1963 | Samot, I'm not sure if I'll be moving the extension around or not. For one thing, I was testing, so I set it to NAT to go outside.. I was under the impresion it shouldn't matter if I'm on the same LAN. |
15:09.39 | Samot | No, it will determine if there is NAT or not. |
15:09.46 | apb1963 | Samot, I was also under the impression that the caller id was irrelevant. |
15:09.47 | Samot | But if there are issues with network traffic.. |
15:09.57 | Samot | Perhaps just set nat to NO to skip the detection. |
15:10.02 | Samot | ?? |
15:10.10 | Samot | Call-ID |
15:10.13 | Samot | Not caller id. |
15:10.48 | Samot | Call-ID is kinda one of the things that are used to track calls. |
15:10.51 | Samot | In transactions. |
15:10.54 | apb1963 | ok... no idea what I'm supposed to do about it. Neither that nor the compatible codec issue, etc. |
15:11.19 | apb1963 | Samot, oh, well then that's not important at the moment since right now I'm not doing any tracking that I'm aware of. |
15:11.20 | Samot | Well the codecs issue is, fix what codecs you allow on Asterisk. |
15:11.25 | Samot | Not YOU |
15:11.34 | Samot | The actual SIP TRANSACTION |
15:11.48 | Samot | It should have an IP |
15:12.16 | Samot | Call-ID should be in a valid URI format. |
15:12.49 | Samot | Honestly, go get a softclient that doesn't suck. |
15:14.18 | Samot | Jitsi is like the PFSense of softphones. |
15:14.24 | apb1963 | and what makes you think it sucks? |
15:14.34 | Samot | Well.. |
15:14.52 | Samot | Every time someone with Jitsi has an issue... |
15:14.58 | Samot | OK, not every time |
15:15.09 | Samot | But like 98% of the time, it's due to Jitsi just sucking. |
15:15.13 | apb1963 | jitsi, like asterisk... has many configurable options. |
15:15.51 | Samot | You're sending a poorly formed Call-ID |
15:15.55 | apb1963 | Well, so far I'm hearing to change asterisk config to presumably allow more codecs, but why that would be I don't know. |
15:16.06 | Samot | Your Asterisk side doesn't have codecs enabled proberly... |
15:16.23 | Samot | What codecs do you have enable on Asterisk? |
15:16.38 | apb1963 | I've got ulaw, alaw, and... I think it's GSM but will have to doublecheck |
15:16.42 | Samot | OK |
15:17.08 | Samot | Look at that mess of a SDP section you have. |
15:17.22 | Samot | None of the codecs Asterisk supports... |
15:17.25 | Samot | Is offered first. |
15:17.38 | Samot | They are like the 8th or 9th offering... |
15:17.42 | Samot | It never makes it to them. |
15:17.59 | apb1963 | oh and g726 |
15:18.02 | Samot | OK |
15:18.12 | Samot | So make it so Jitsi is ONLY using those codecs. |
15:18.19 | Samot | Why offer 25 codecs? |
15:18.26 | apb1963 | I don't know which is the best one to use |
15:18.26 | Samot | In random order... |
15:18.37 | Samot | THEY HAVE TO MATCH |
15:18.42 | apb1963 | that much i know |
15:18.45 | Samot | OK |
15:18.51 | igcewieling | when testing use ulaw or alaw |
15:18.55 | Samot | So if you are using ulaw, alaw, g726 and gsm.. |
15:19.03 | Samot | That's ALL Jitsi should have ENABLED |
15:19.22 | Samot | Yes, ulaw should always be first |
15:19.34 | Samot | or alaw, depending on your country of origin. |
15:20.07 | Samot | g711 is pretty much the standard. |
15:20.30 | *** join/#asterisk EnrgySmth (d8eba101@gateway/web/freenode/ip.216.235.161.1) |
15:22.07 | [TK]D-Fender | ULAW / G711u /PCMu |
15:22.09 | [TK]D-Fender | just do it |
15:25.41 | apb1963 | http://pastebin.com/spzpaA3h |
15:26.27 | Samot | No. |
15:26.32 | Samot | Turn of DEBUG |
15:26.36 | Samot | core set debug 0 |
15:26.45 | Samot | That is just too much garbage we don't need. |
15:27.01 | *** join/#asterisk cmendes0101 (~cmendes01@47-144-223-7.lsan.ca.frontiernet.net) |
15:27.40 | *** join/#asterisk miralin (~Thunderbi@194.8.128.50) |
15:34.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
15:36.04 | apb1963 | http://pastebin.com/pN1PiYRU |
15:39.05 | [TK]D-Fender | What kind of password do you have on it? |
15:39.57 | apb1963 | "it" |
15:40.58 | apb1963 | normal passwd? There's also a TLS cert. Not sure... |
15:41.10 | apb1963 | It's setup to use the cert |
15:41.24 | [TK]D-Fender | LENGTH, CHARACATER MIX< ETC |
15:42.24 | apb1963 | 10, normal ASCII chars. |
15:42.43 | [TK]D-Fender | Show both sides config in full |
15:44.32 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-cqfstshuhpmsgzoi) |
15:44.32 | *** mode/#asterisk [+o rmudgett] by ChanServ |
15:45.07 | apb1963 | http://picpaste.com/pics/Screenshot_from_2017-02-10_07-43-08-9HH0MyaU.1486741479.png |
15:45.09 | *** join/#asterisk Tiffon (~name@unaffiliated/tiff0n) |
15:47.28 | apb1963 | http://picpaste.com/Screenshot_from_2017-02-10_07-45-24-6Szd8UKa.png |
15:48.57 | apb1963 | http://picpaste.com/pics/Screenshot_from_2017-02-10_07-47-47-l17Hdlen.1486741716.png |
15:49.02 | [TK]D-Fender | Something tells me you should ahve the IP in the ID there |
15:49.10 | [TK]D-Fender | shouldn't |
15:52.06 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
15:58.15 | *** join/#asterisk lankanmon_ (~LKNnet@2607:fea8:d20:239:11e0:707c:2961:d41e) |
16:01.12 | *** join/#asterisk apb1963 (~apb1963@107-146-220-94.res.bhn.net) |
16:03.16 | apb1963 | http://pastebin.com/4Gf0hL52 |
16:04.12 | apb1963 | [TK]D-Fender, sans IP |
16:08.50 | apb1963 | needs a slow break. brb |
16:18.50 | [TK]D-Fender | I never saw the ASTERISK SIDE CONFIGS |
16:19.30 | *** join/#asterisk samwierema (~samwierem@095-097-255-066.static.chello.nl) |
16:20.12 | *** join/#asterisk BrencoInc (~textual@50-205-197-158-static.hfc.comcastbusiness.net) |
16:23.50 | apb1963 | ok |
16:23.53 | *** join/#asterisk BrencoInc (~textual@50-205-197-158-static.hfc.comcastbusiness.net) |
16:25.02 | *** join/#asterisk BrencoInc (~textual@50-205-197-158-static.hfc.comcastbusiness.net) |
16:26.14 | apb1963 | http://picpaste.com/Screenshot_from_2017-02-10_08-25-00-s8DWRNa0.png |
16:27.48 | apb1963 | http://picpaste.com/Screenshot_from_2017-02-10_08-26-56-toK8Dp6C.png |
16:28.09 | igcewieling | oh. FreePBX. |
16:28.39 | *** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
16:29.16 | apb1963 | [TK]D-Fender, that enough or what else? |
16:29.33 | [TK]D-Fender | Where is the EXTENSION? |
16:31.05 | *** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp130-03-74-14-78-25.dsl.bell.ca) |
16:34.13 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
16:34.16 | wasanzy | hi |
16:34.26 | apb1963 | http://picpaste.com/Screenshot_from_2017-02-10_08-26-56-toK8Dp6C.png I'm guessing that you're going to notice something on the port side... |
16:34.30 | wasanzy | does PlayBack() support gsm format? |
16:35.50 | [TK]D-Fender | apb1963, that is not the extension |
16:36.12 | [TK]D-Fender | wasanzy, Yes |
16:38.10 | apb1963 | http://picpaste.com/Screenshot_from_2017-02-10_08-33-27-h6RzvEq7.png sorry about that |
16:39.56 | apb1963 | [TK]D-Fender, change the port? |
16:40.26 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
16:49.51 | *** join/#asterisk axp (~axp@mail.hasinet.at) |
16:50.31 | axp | does anyone know how to debug asterisk 100% cpu usage on an very small system? |
16:50.38 | *** join/#asterisk lankanmon (~LKNnet@CPE1cabc0702d13-CM1cabc0702d10.cpe.net.cable.rogers.com) |
17:07.31 | apb1963 | axp, does it happen immediately on startup or when? |
17:07.46 | apb1963 | axp, is there something that seems to trigger it? |
17:08.34 | apb1963 | axp, crank up the logs... for starters |
17:12.44 | axp | apb1963, hi, thx for info, but logfiles are empty, i do restart asterisk once a day via a cron job |
17:13.33 | *** join/#asterisk brokensyntax (~quassel@45.62.240.131) |
17:16.30 | apb1963 | axp, try sip set debug on |
17:22.10 | *** join/#asterisk Oatmeal (~Suzeanne@2001:558:600d:c:35f1:4a2f:63ab:d8ac) |
17:25.47 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
17:27.42 | *** join/#asterisk thiagoc (~thiagoc@unaffiliated/thiagoc) |
17:29.28 | *** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net) |
17:30.48 | igcewieling | updated iptables which filters out some of the common sip scanners, added a new rule today: http://pastebin.com/2mKSw1nZ |
17:30.56 | igcewieling | maybe it will help someone |
17:31.24 | Samot | No siparmyknife? |
17:31.53 | igcewieling | I've not seen it attacking |
17:32.01 | Samot | Oh |
17:32.04 | Samot | I don't wait for it. |
17:32.14 | Samot | It's a known sip scanner UA |
17:32.31 | igcewieling | I could add it. |
17:32.34 | drmessano | Yeah I have it in mine |
17:33.08 | igcewieling | the exact string "siparmyknife" is anywhere in the fist 1500 bytes? |
17:33.46 | drmessano | Im SSHing into one of my boxes.. I wanna check my list against yours |
17:34.54 | Samot | friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent|VaxIPUserAgent|sip-scan|sipsak|sundayddr|iWar|SIVuS|Gulp|sipv|smap|siparmyknife |
17:35.07 | Samot | friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent|VaxIPUserAgent|sipscan|sipsak|sundayddr|iWar|SIVuS|Gulp|sipv|smap|siparmyknife |
17:35.15 | Samot | That's what I have right now. |
17:36.52 | igcewieling | I called the extension they registered as and got tt-monkeys. 8-| |
17:38.24 | drmessano | Yeah Samot and I overlap.. he actually has a few I dont |
17:38.28 | drmessano | So I have nothing to add |
17:39.02 | igcewieling | thanks for the info |
17:39.06 | drmessano | I have sip-scan with a hyphen |
17:40.16 | Samot | I check "friendly scanner uas" or "sip scanner uas" |
17:40.50 | Samot | On google every couple of months or so..just to see if any UA is overly reported a lot. |
17:41.19 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
17:43.05 | drmessano | VaxIPUserAgent and VaxSIPUserAgent? |
17:43.21 | igcewieling | The SIP VOIP v11.0.0 useragent I see must be dumb. many scripts seem to stop registering once they successfully register, the SIP VOIP one didn't. |
17:44.11 | igcewieling | drmessano: you never know when you need a 20 year old minicomputer made by a company which does not exist anymore to run a softphone! |
17:44.29 | drmessano | heh |
17:44.49 | drmessano | Samot: |
17:45.29 | drmessano | sip-scan seems to be prevalent on google, not "sipscan". Unless you have seen it, and want to add the hyphenated as well |
17:51.34 | igcewieling | Hmmm...I wonder if I could use a useragent whitelist instead of blacklist. We don't have a lot of different user agents |
17:53.08 | drmessano | The iptables rules are such a lifesaver |
17:59.12 | *** join/#asterisk Demon_VoIP (~demon@109.60.222.253) |
18:03.02 | *** join/#asterisk skywayskase (~skywayska@67.139.42.219) |
18:37.13 | *** join/#asterisk Follow-me (~JACK@86.108.39.224) |
18:38.16 | Follow-me | guys how do i change asterisk database location |
18:39.06 | Follow-me | Database: mor mor@185.23.2.2:1201 |
18:39.21 | *** join/#asterisk pdugas (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
18:40.09 | [TK]D-Fender | What database? |
18:40.29 | Follow-me | connected to asterisk |
18:40.56 | [TK]D-Fender | What database? |
18:41.01 | [TK]D-Fender | Databases don't connect to tAsterisk |
18:41.05 | [TK]D-Fender | that is backwards |
18:42.30 | Follow-me | mysql database |
18:42.56 | [TK]D-Fender | Mysql does not connect to Asterisk. |
18:43.25 | [TK]D-Fender | Other things use databases. Databases don't use Asterisk |
18:44.02 | [TK]D-Fender | If your * talks to a datyabse tthere is going to be a blatantly obvious config file that specifies where itt is connecting |
18:44.20 | [TK]D-Fender | If your * talks to a database there is going to be a blatantly obvious config file that specifies where it is connecting to it |
18:53.39 | apb1963 | [TK]D-Fender, so I take it you've given up on my issue? |
18:54.30 | drmessano | You never answered him |
18:54.35 | apb1963 | drmessano, how so?\ |
18:54.46 | drmessano | Looks like he asked for the EXTENSION |
18:54.50 | drmessano | and you didnt provide it |
18:54.54 | apb1963 | drmessano, yes... I did. |
18:55.12 | apb1963 | of course at this point the file has expired |
18:55.49 | apb1963 | http://picpaste.com/Screenshot_from_2017-02-10_08-33-27-h6RzvEq7.png |
18:55.58 | apb1963 | which has expired |
18:56.02 | apb1963 | so don't look |
18:56.05 | drmessano | I dont know why people expire pastes |
18:56.15 | apb1963 | why not? |
18:56.27 | drmessano | Set it to a day or something.. not 30 mins |
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18:56.39 | apb1963 | usually that's enough |
18:56.56 | drmessano | Awfully presumptions that someone is sitting there RIGHT NOW EAGERLY waiting to help |
18:57.04 | drmessano | presumptious |
18:57.07 | apb1963 | if someone is paying attention it takes much less than that before it expires |
18:57.24 | apb1963 | I don't paste until someone is paying attention |
18:57.27 | apb1963 | as a general rule |
18:57.43 | drmessano | Lot of good it did you here |
18:58.45 | Samot | Yeah, I went to click on it a while ago and it was expired. |
18:59.38 | drmessano | I usually put serial paste expiration offenders on ignore |
18:59.59 | apb1963 | http://picpaste.com/pics/Screenshot_from_2017-02-10_08-33-27-P3iOQD0y.1486753167.png |
19:00.01 | drmessano | If your paste is national security enough that you have to expire it in 30 seconds, you should have someone you can call |
19:00.33 | Samot | That's not even the entire thing. |
19:00.53 | apb1963 | was there something in particular you were looking for? |
19:01.04 | drmessano | Yeah the REST of it |
19:01.26 | apb1963 | The whole page doesn't fit in one screenshot and I snapshotted the salient piece. |
19:01.31 | Samot | We know. |
19:01.40 | Samot | You have to take a couple. |
19:01.45 | drmessano | Thats your opinion |
19:01.47 | drmessano | and its wrong |
19:01.58 | drmessano | "the salient piece" |
19:02.01 | drmessano | Keep going |
19:02.10 | apb1963 | I take shots all year long... |
19:02.13 | Samot | You can either pastebin the details from sip_additional.conf |
19:02.22 | Samot | Or you can take more screenshots.. |
19:02.38 | apb1963 | What? You're giving me actual detailed instructions as to what you want to see? Say it isn't so Ethel! |
19:02.46 | Samot | Dude. |
19:02.54 | Samot | We asked to see the extension's settings. |
19:02.56 | drmessano | ..... |
19:03.00 | Samot | You gave us half of it. |
19:03.02 | drmessano | You posted..... |
19:03.03 | drmessano | "the salient piece" |
19:03.09 | drmessano | Like WTF |
19:03.15 | Samot | Also, we've told you numerous times. GO TO #FREEPBX |
19:03.15 | drmessano | Dont ask for help |
19:03.22 | drmessano | Then post what YOU fucking think is important |
19:04.07 | Samot | Do not sit here and act like you haven't been given implicit instructions, when you have and have failed to follow them. |
19:04.11 | drmessano | You do this same crap when you're asked for debug |
19:04.25 | drmessano | You obviously dont know what is important.. so post what is ASKED |
19:05.46 | [\\\] | Another day, another dollar. |
19:06.11 | apb1963 | http://pastebin.com/7mVVviQV |
19:06.54 | Samot | OK, I was expecting just the extension were are supposed to look at.. |
19:06.58 | Samot | But which one IS IT? |
19:07.16 | apb1963 | 3304 |
19:07.20 | drmessano | lol |
19:07.50 | Samot | avpf=no <--- ????/ |
19:07.58 | Samot | You are ATTEMPTING a VIDEO call |
19:08.17 | Samot | But you don't have anything in there for a video call. |
19:09.24 | Samot | I have said this before, make your softphone ONLY use the codecs that you have allowed in Asterisk. |
19:10.14 | Samot | Stop offering up garbage in the SDP that doesn't apply to anything. |
19:13.35 | apb1963 | strange.. I could have sworn I added all video codecs in for * |
19:14.16 | Samot | sip show settings |
19:14.20 | Samot | prove it. |
19:15.08 | apb1963 | well I'm looking at it... I left out a couple.. that's why I say it's strange because I thought I got them all |
19:15.24 | apb1963 | hence "could have sworn..." |
19:15.50 | apb1963 | so... added those in... working on testing. |
19:20.23 | drmessano | well? |
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19:35.46 | [TK]D-Fender | <apb1963> strange.. I could have sworn I added all video codecs in for * <- NO codecs were specified in the peer at all |
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19:48.55 | *** join/#asterisk foobar2017 (bebb81f2@gateway/web/freenode/ip.190.187.129.242) |
19:49.02 | foobar2017 | hi |
19:49.49 | foobar2017 | I'm getting a load of 100% from mysql in my asterisk installation. Any help¡ |
19:49.50 | foobar2017 | ?? |
19:50.10 | igcewieling | foobar2017: did it run out of disk space? |
19:51.31 | foobar2017 | igcewieling: I have more then 400 GiB of free space. |
19:54.07 | foobar2017 | *than |
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19:59.16 | apb1963 | Thanks guys, I appreciate the help. I have to pick this up later, have a nice day! |
19:59.30 | Samot | We probably won't. |
19:59.49 | drmessano | Yeah not even close |
20:01.04 | igcewieling | foobar2017: is it running in a VM? |
20:01.21 | drmessano | I am starting to hate systemd |
20:02.30 | igcewieling | drmessano: centos 7 ? |
20:02.37 | drmessano | Nah, Ubuntu |
20:03.31 | drmessano | Trying to delete some lock files before I start a service |
20:04.24 | foobar2017 | igcewieling: no, it's baremetal |
20:04.52 | foobar2017 | igcewieling: it's running on debian |
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20:28.06 | BrencoInc | foobar2017 dumb question but do you have any calls looping or sip scaner hitting you? |
20:36.06 | foobar2017 | BrencoInc, igcewieling: I've just cleared out the cdr table. |
20:36.12 | Alex_Bkash | anyone compiled asterisk in cygwin? |
20:36.20 | foobar2017 | And now it's normal. |
20:42.05 | Alex_Bkash | anyone compiled asterisk in cygwin? |
20:49.29 | igcewieling | edges away from the crazy person |
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20:56.15 | [TK]D-Fender | ~polls |
20:56.18 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
20:56.22 | [TK]D-Fender | ^^^^^^^^^^^^ |
21:03.54 | *** join/#asterisk klow (~klow@66.114.139.162) |
21:04.12 | johnny_|_ | ~ask |
21:04.12 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
21:09.03 | drmessano | and asking 6 minutes apart |
21:09.08 | drmessano | With no channel scroll |
21:09.35 | [TK]D-Fender | Wouldn't believe the time I wasted in PM finding out what the project was in the first place... |
21:09.56 | drmessano | ROFL |
21:10.05 | drmessano | Do I even want to? |
21:10.51 | Samot | Of course. |
21:11.00 | Samot | What was it? |
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21:14.19 | drmessano | if (cash me ousside) echo "how bow dah"; |
21:14.32 | drmessano | That's how you compile Asterisk in Cygwin |
21:16.31 | drmessano | [TK]D-Fender: TELL US |
21:19.34 | [TK]D-Fender | drmessano, PMM |
21:32.20 | [TK]D-Fender | checkout time, BBIAB |
21:32.57 | Samot | Now tell me! |
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22:13.28 | Sean-Der | Samot: thanks again for the help last night, SIP provider said I wasn't responding to INVITES fast enough |
22:14.04 | Sean-Der | My first OK took about 5 seconds, and they sent the same call through another friend |
22:14.22 | Sean-Der | which would get the 4xx thrown |
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