IRC log for #asterisk on 20170210

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01:56.29wonderworldis Opus for Asterisk 13 available yet? http://blogs.digium.com/2016/09/30/opus-in-asterisk/
01:58.15filein latest version, yes
02:01.36wonderworldi run 13.13.1. is it a seperate download or should it be available already after building from source?
02:02.22fileyou have to select it in "make menuselect" under Codecs to have it download, and that option requires the "xmlstarlet" utility be installed to use
02:02.39wonderworldtnx let me check
02:04.37wonderworldok i didn't compile it XXX codec_opus
02:05.52wonderworldwhat dazzeled me was that asterisk announces to be able to handle OPUS via SDP/webrtc
02:06.13wonderworldbut it didn't work. asterisk always fell back to ulaw
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02:14.49wonderworldwhere would i get xmlstartlet?
02:37.58wonderworldi manualy installed opus. now asterisk crashes in the middle of loading the module. https://ghostbin.com/paste/66kqz
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03:35.03wyoung@file is a perl thingy isn't it?
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04:21.56*** join/#asterisk Sean-Der (~Sean-Der@siobud.com)
04:27.14Sean-DerDoes anyone know how to turn off Loop Detection, for some reason certain incoming calls from a SIP trunk are getting a 482 (Loop Detected)
04:27.21Sean-DerI can anonymize the inbound SIP if it helps
04:27.55Sean-Derbut 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
04:28.18Sean-DerI am running 13.13.1
04:28.50SamotSo they are sending you a call?
04:28.57SamotOr are you sending them a call?
04:29.09Sean-DerThey are sending me a call, and I respond to the INVITE with a 482
04:29.51Sean-DerThe 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)
04:29.59Sean-Derlike I am not responding to the first INVITE fast enough
04:30.47SamotWhy are you responding with a 482?
04:30.54SamotThat shouldn't be a reply to an INVITE
04:30.59SamotIt should be a 1xx reply.
04:31.04SamotAt least.
04:32.05Sean-DerI have no idea, Asterisk is doing it for me. I have no logic in my dialplan (or sip config) relating to this
04:32.05SamotShow one of these calls..
04:32.13Samotasterisk -r
04:32.17Samotsip set debug on
04:32.23Samotor pjsip set logger on
04:32.27SamotDepending on your tech..
04:32.29SamotMake a call
04:32.31Samot~pb
04:32.31infobotwell, 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.
04:32.38SamotShow the SIP debug..
04:32.47SamotWant to see the INVITE and the 482 reply to it.
04:32.56Sean-DerYep!
04:35.54Sean-DerSamot: mind if I PM you the paste (has some numbers and IPs) not a big deal if not
04:35.59Sean-Derand we can keep chat in here
04:36.36Samotok
04:38.06Sean-DerI don't know what Asterisk uses to throw a Loop Detected, but gonna grep around and see if I can figure it out
04:38.30SamotWhy are they sending you calls on a 10.10.x.x network?
04:38.38SamotDo you have a private LAN connection with them?
04:39.23Sean-DerIt is a DMZ, the calls the majority of the time
04:40.31drmessanoO.o
04:40.43Sean-Dermaybe not the right term, but the interface is a 10.10, but it is a world routable box
04:40.54Sean-Dersorry the calls work the majority of the time
04:41.04Sean-Derand this call would work, if Asterisk wouldn't throw the 482
04:42.02Samot10.10 is not world routable.
04:42.28Samot10.x.x.x is a completely private /8
04:42.40Sean-Der10.10 is the local interface, but it is setup so that all traffic to a world routable address forwards to it
04:42.48SamotOK.
04:42.55SamotSo it's NAT'd.
04:44.55Sean-Deryes
04:45.23SamotIs this just coming straight into your Asterisk box?
04:46.03SamotProvider --> Asterisk
04:46.10Sean-Deryep
04:46.19SamotWhich line in the debug is your Asterisk IP
04:46.27SamotTell me the first line it appears.
04:47.36drmessanoIts a secret
04:47.40SamotAnd not the INVITE with the 10.10.
04:47.46SamotNo, he sent me the debug.
04:48.05SamotI have no issue with him not wanting information logged in botbot..
04:48.38SamotAnd it's not sanitised, so yay!
04:49.10SamotI think I can see why Asterisk is replying with a 482 but need to confirm.
04:49.23Sean-DerI actually don't see it, I checked a couple times
04:49.34SamotIt has to be there.
04:49.43SamotThe WAN IP.
04:49.48SamotThe external IP for it.
04:49.51SamotNot the local.
04:50.06Sean-DerThat is the WAN IP
04:50.07SamotK
04:50.32Sean-DerI guess I don't understand how this was ever working, if I restart Asterisk it works for a while
04:50.42Sean-Derbut eventually gets into a bad state
04:51.42SamotWhat is the IP in the TO header?
04:51.47SamotIs that yours?
04:51.52SamotBecause if it's not...
04:52.03SamotThis is a poorly routed SIP message.
04:52.13SamotAnd that's why Asterisk is seeing a loop.
04:52.47SamotMax-Forwards: 67 <-- You see this?
04:53.03Sean-Deryep! default is 70, only thing that rings a bell
04:53.09SamotOK
04:53.21SamotIt goes down by one for every route in the path.
04:53.31SamotSo it it was a straight route..
04:53.36SamotIt would always be 69
04:53.38SamotDUDE
04:53.51SamotThis took two extra routes.
04:54.36SamotAnd the TO URI is not your IP
04:54.38*** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos)
04:54.46SamotIt's one of theirs in the Via headers.
04:54.57Sean-DerSo this is out of my control?
04:55.25Sean-DerLooking through calls that worked it has always been this way
04:55.32Sean-DerSo somehow this was working in Asterisk 1.8
04:55.41SamotWell..
04:55.43Sean-Derbut I am taking 1.8 -> 13 boxes, or trying at least
04:55.47SamotThat would have been a bug
04:55.49SamotOr..
04:55.57SamotThings have changed in the last 5 years.
04:56.14SamotThink of it this way...
04:56.16Sean-DerI really should have upgrade sooner, thank you very much for your help Samot
04:57.02SamotOriginates at PSTN -> Routes to A -> A routes to B -> B routes to C -> C routes to A
04:57.14SamotLoopish.
04:57.19Sean-DerI 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
04:57.32SamotYour RR headers and Via headers look loopish..
04:57.53SamotIt may not pick it up right away...
04:58.11Sean-DerSo I am pretty lost when it comes to anything besides the basics
04:58.35SamotSIP in generally has improved a lot over the years...
04:58.49SamotSo things that worked 10 years ago...
04:59.06SamotMight not work now because of improvements and better ways.
04:59.27Sean-Derok so I restarted the server, and it is working
04:59.37Sean-DerDo you mind if I dump pcap again, and compare maybe?
04:59.37SamotShow another call.
04:59.41Sean-DerACK
04:59.48SamotI want to see the difference.
05:08.38Sean-DerSamot: so it looks like it still is malformed, but this call worked
05:08.54Sean-Dersorry for reason ZNC keeps timing out, I think my VPS is having issues
05:08.57SamotOne sec.
05:09.53SamotYes but a completely different set of routes completely.
05:10.31SamotAnd handled differently.
05:11.16SamotIn the one the failed, the first record-route has the standard lr tag.
05:11.21SamotWhich can be ignored..
05:11.55SamotThe working call, not only does it come from a different route. It has the lr=on tag which forces lr to be respected.
05:12.36SamotAs well the second record-route on the working call, while it has just a normal lr tag (like the non-working one)...
05:12.48SamotIt has a ftag attached...
05:13.37SamotThis routing path is more sane and has better transaction and dialog management it looks like.
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05:14.18SamotI'm speculating on dialog management since  I can't see the dialog happening in this.
05:14.37SamotBut the transaction is being handled better.
05:16.16Sean-DerI am sorta stuck, because that is stuff I can't change
05:16.37Sean-DerI can make a ticket and send data to them
05:17.06SamotYes.
05:17.12SamotShow them both.
05:17.28Sean-Derwill do, and will try my best to relay your knowledge
05:17.36SamotThey can make it so your route isn't load balanced or whatever they are doing.
05:17.39Sean-Derthrough the sieve of my cluelessnes
05:17.54SamotYou don't need to say all I said..
05:18.12SamotJust say "This way fails, this way doesn't. Please always route us to the way that doesn't."
05:18.22SamotIf you want, make a few calls in a row.
05:18.34SamotGive them, here's X that failed and Y that didn't.
05:18.41SamotNotice X is always the same route.
05:18.51SamotY is always the same route. Please always make Y
05:18.54Sean-Derwill do, thank you very much for the help again that was super helpful
05:19.22SamotThese are Sonus boxes too..
05:19.45Sean-DerI saw people complaining about them online
05:19.47Sean-Dergenerally a pain?
05:19.59SamotThey are expensive.
05:20.30SamotWho's your provider, if you don't mind..
05:20.57Sean-Derdon't mind! not closely tied to them https://www.nexvortex.com/
05:21.26Sean-DerI didn't chose them and honestly has worked pretty well (up until now)
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05:21.41Sean-Derbeen with them since 2011 and haven't had issue 1.6 and 1.8 boxes
05:21.44SamotThere are two types of companies that use Sonus SBCs...
05:22.37SamotReal carriers/providers and those that think they can be one because they had the money to spend on a really shiny box.
05:23.25SamotIt don't matter you driving a Porsche when all you do is grind the clutch....
05:24.03Sean-Derheh
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05:24.47Sean-Derwell let's hope that tomorrow I send them these captures and they can just fix it
05:25.04SamotOr 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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05:26.03drmessano4 on the floor too
05:26.05SamotHe'll tell Neil straight up, no southern man want him around anyhow.
05:26.43SamotAnd a GUN RACK
05:26.59drmessanoAh its definitely a new day
05:27.12Samot"I don't even have A gun, let alone many that would facilitate a gun rack"
05:27.21drmessanoAround 23:00 I said I had my fill of idiots
05:27.39drmessanoSo a buddy of mine posts asking for a recommendation for battery repair
05:27.44drmessanofor an iphone 5s
05:28.04drmessanoSome says
05:28.13drmessano"Try T-Mobile or Apple or online"
05:28.25drmessanoTry shutting the fsck up
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05:30.39drmessanoFor $60 I may get my 5s done
05:34.27wyoungSamot: Doesn't sound like a real southerner then
05:35.37SamotNo.
05:35.48SamotBut I was quoting a classic movie.
05:35.52wyoungright
05:35.57wyoungNever heard of it :P
05:36.23wyoungDo they use asterisk in the movie?
05:37.35SamotNo.
05:37.47SamotBecause, 1992.
05:56.51drmessanoSo back when Asterisk was in beta
05:59.59SamotI though ti was like 98/99 when it was first released...
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06:02.50SamotWhen did Kelly Ann go to the Obama pronunciation of ISIS?
06:05.22drmessanoIt was late 99, I think
06:05.57drmessanoKAC is such a train wreck
06:06.11drmessanoWorst part is she's not even as amusing as Spicer
06:06.50drmessanoHe should definitely be invited to Astricon
06:08.22SamotAt least he took SNL in stride.
06:20.09drmessanoYeah
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07:43.16tparcinaGood morning.
07:43.48tparcinaSamot: What makes you think it is an network issue?
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08:42.56RomanDcozHello Guys, when enable TCP protocol i am getting below error, Please suggest me.
08:42.57RomanDcozhttp://pastebin.com/h1SbyuHa
08:44.08RomanDcozerror 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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09:03.19junedHi all, is it possible to add IP range for peer ? I want receive incoming calls from IP range instead of single IP.
09:04.06juned[provider_1_trunk]
09:04.06junedtype=102.254.233.23/24
09:04.06junedcontext=provider_1_incoming
09:04.15junedsomething like this
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09:21.01bouncemanHowdy, 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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09:35.00bouncemanAlternative 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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10:05.37dokmaMy 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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11:42.27Samottparcina: Because no audio/one way audio issues area always network issues.
11:43.50Samotjuned: Not with Chan_SIP
11:44.02tparcinaSamot: I have look at captured network traffic, and I have managed to narrow down the problem.
11:44.31tparcinaSamot: A is behind my VoIP provider.
11:44.56SamotRight.
11:45.03junedHi Samot, can we do that using pjsip  ?
11:45.05SamotRomanDcoz: Via: SIP/2.0/TCP 202.MY.asterisk.IP:5060;branch=z9hG4bK238009e6;rport
11:45.21tparcinaAnd the problem is with RTP stream that * is sending to my ISP (to A), after the transfer.
11:45.24Samot^^^ Does your provider support TCP transport.
11:45.33tparcinaSamot: Then new RTP stram starts.
11:45.35Samotjuned: yes.
11:45.43junedThanks Samot...
11:45.52SamotYes.
11:45.58SamotOf course, it's going to change.
11:46.01tparcinaSamot: It goes from same address:port to the same address:port as previous RTP stream.
11:46.11SamotUh?
11:46.12SamotNo.
11:46.17RomanDcozSamot: dont know, i think need to ask provider for this
11:46.23SamotThe RTP ports between the provider and the PBX will NOT change.
11:46.29SamotThat channel is never changed.
11:46.32SamotIt's always the same.
11:47.10SamotProvider to PBX = 1 channel
11:47.11tparcinaSamot: Just I have noticed on this one case that when there was the problem the used RTP codec has changed.
11:47.14SamotThat's it.
11:47.26SamotThat channel is all that exists between Asterisk and the Provider..
11:47.39SamotThose RTP ports are established when the call comes in.
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11:47.49tparcinaSamot: Now I'm trying to find more failed transfer examples to see do they all have codec change.
11:48.14tparcinaSamot: I don't know, we are using UDP with our provider.
11:48.24SamotThat TCP comment wasn't for you.
11:49.06tparcinaok :)
11:53.24RomanDcozis this correct way to set cdrs value using socket fputs($oSocket, "SetCDRUserField: 123456789\r\n"); ?
11:56.40RomanDcozI need to set value  like custom in any cdrs field , I am using cdr_mysql module to insert value directly
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12:06.49MaliutaLapdokma: the format that matches the codec you're using
12:07.29MaliutaLapdokma: also you can't just use any music, most codecs have limited frequency ranges - so choose wisely
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13:16.59apb1963http://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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13:55.37SamotINVITE sip:1-703-376-3246@192.168.0.102 SIP/2.0 <-- WTF?
13:56.37apb1963?
14:01.46SamotWhy are there dashing in the INVITE
14:02.02apb1963why not?
14:02.22SamotAnd how are you matching or stripping them?
14:02.34apb1963I'm not
14:02.45apb1963I'm not doing anything special
14:02.56[TK]D-FenderI'm not seeing an answer to the 401 challenges
14:04.48apb1963Anybody got a D-Fender to English translator handy?
14:05.12[TK]D-Fenderyou send the invite, * sends a 401 CHALLENGING them, and Jitsi isn't coming back with papers
14:05.22apb1963ok so it's a jitsi issue
14:05.39apb1963but.. not configuration right?  Bug yes?
14:06.43SamotYeah, that's what I'm seeing to.
14:07.05SamotI'm also seeing a bad INVITE structure..
14:07.32SamotAll a million codecs and other crap that really don't need to be involved but hey..
14:07.44SamotINVITE sip:1-703-376-3246@192.168.0.102 SIP/2.0 <--  This will NOT work.
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14:08.09apb1963Why not?
14:08.13Samot1) You have no way to match that and strip the -
14:08.19Samot2) No carrier accepts that format.
14:08.38SamotThe dashes are human readable things.
14:08.55SamotThey are not part of any routing of a DID.
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14:11.00apb1963well, it's never been a problem prior to TLS
14:12.26SamotSo sending a call like that from Jitsi works on plain ole UDP?
14:12.30SamotNon-TLS?
14:13.50[TK]D-Fender<apb1963> but.. not configuration right?  Bug yes? <- configuration could be an issue
14:14.58apb1963well, 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.
14:15.06apb1963Samot, yes
14:16.01apb1963[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.
14:16.43SamotSo you have a pattern match of _1-NXX-NXX-XXXX?
14:17.42apb1963Samot, Looks somewhat familiar what's automatically set for me in the gooey.
14:17.52apb1963^to
14:19.29SamotAnd you're sending it like that to the provider?!
14:19.35apb1963yummy yummy gooey gooey rich and chewy inside out
14:19.50apb1963whatever you see in the log
14:20.00SamotI see a failed call.
14:20.05SamotI see a call that never makes it to dialplan.
14:20.19SamotSo no, I can't see how it was accepted by the provider in that format.
14:20.32SamotNor do I know of ANY provider that would accept that format.
14:21.07apb1963I"m assuming at this point that it doesn't make it to the dialplan because it doesn't answer the 401 challenge.
14:21.13SamotYes.
14:21.48SamotSo why don't you show us a non-TLS call that works..
14:21.50apb1963so.. once jitsi fixes that issue, i'll revisit that issue
14:22.14SamotThat's kind of a show stopper bug.
14:22.21SamotNot responding to 401 challenges.
14:22.31SamotMake 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
14:26.06wonderworldwhich libsrtp version is recommended for use with asterisk?
14:26.16SamotI want to see proof that backs up the claim that it works non-TLS and that call actually makes it to carrier.
14:26.18apb1963Well, 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.
14:26.34apb1963s/5061/5060
14:26.44SamotNo, that's a cofiguration issue.
14:26.49apb1963i.e. the wrong port.
14:26.51apb1963No, that's a bug.
14:27.00SamotWhere do you put the port in the host name?
14:27.13apb1963?
14:27.27SamotWell there's generally two options for hosts.
14:27.33apb1963jitsi has a configuration window.   It allows me to specify a port.
14:27.34SamotThe Registration/SIP Proxy host
14:27.41SamotOr the Outbound Proxy Host
14:27.57apb1963Both
14:28.15SamotOB proxy is never required.
14:28.18SamotOr shouldn't be.
14:28.25apb1963With UDP, it's not.
14:28.59SamotWith TLS it should be either.
14:29.05apb1963Hence the bug
14:29.35apb1963Using the proxy settings forces it to use the right port.
14:29.57apb1963It'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)
14:30.16SamotDo you have the right port in the main SIP proxy?
14:30.21Samot:5061?
14:30.23apb1963Yes
14:32.47[TK]D-FenderJust show all of it
14:32.58apb1963http://picpaste.com/pics/Screenshot_from_2017-02-10_06-30-52-WsA4kBzA.1486737153.png
14:33.01[TK]D-FenderWe shouldn't be wasting time doing this blind
14:33.28[TK]D-FenderAnd ditch that keepalive
14:33.36[TK]D-Fenderit's flooding pointlessly
14:34.10apb1963None, Options, Register, CRLF are my options
14:34.21[TK]D-FenderDITCH <-
14:34.25apb1963None
14:34.42apb1963I 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)
14:37.03apb1963summary window: http://picpaste.com/pics/Screenshot_from_2017-02-10_06-35-13-1k0yakSn.1486737387.png
14:38.09wonderworldwhats 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)
14:38.42EmleyMoorI 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
14:38.48EmleyMoorsy way I can implement this? (I have ...
14:38.51EmleyMoor... processing for presence/validity of caller ID on my dialplan already, just not passed through for fax.
14:38.54EmleyMoor)
14:39.51SamotWhere does the fax number go?
14:40.07SamotDoes it route to an endpoint like an ATA?
14:40.17[TK]D-FenderIIRC you should not be looking at ZRTP if you were looking for SRTP
14:40.22WIMPyEmleyMoor: Experiment with different cause codes in Hangup().
14:40.56EmleyMoorSamot: To an iaxwodem
14:40.59EmleyMoormodem*
14:41.04SamotOK
14:41.12SamotSo they are repeatedly calling?
14:41.16EmleyMoorWIMPy: Ah... good point...
14:41.24SamotLike one right after the other?
14:41.38WIMPyEmleyMoor: Why is a software modem busy?
14:41.40EmleyMoorSamot: In this case, that is what happened (fake Danish number, tech support scam call)
14:41.53SamotBut they are making multiple calls at once?
14:41.57EmleyMoorWIMPy: Because it's already dealing with the first call
14:42.08SamotAre they making enough calls to trigger the modem to throw a busy?!
14:42.08EmleyMoorSamot: Yes
14:42.12WIMPyWhy do you only have one running?
14:42.34SamotSo everything is acting as it should.
14:42.58Samot1) you cannnot take a fax call and put it in the queue
14:43.15EmleyMoorI could add more, perhaps... but I actually very rarely receive calls genuinely for it anyway
14:43.24WIMPyWhy not?
14:43.36SamotPut a fax call in the queue?
14:43.48WIMPyyes
14:43.55*** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212)
14:43.59SamotUhm.
14:44.11SamotNegotiation...
14:44.15SamotWhen answered.
14:44.29SamotA sending fax machine is expecting tones..
14:44.34WIMPyYou don't have to answer to use Queue.
14:44.39SamotOK.
14:44.43SamotSure.
14:44.52SamotOR you can just handle your calls correctly.
14:44.58SamotI'm a BIG fan of that.
14:45.14WIMPyWhy wouldn't that be correctly?
14:45.44SamotWell he has a failover to his cell phone triggered when the FAX line is busy.
14:45.53SamotI'm not sure that's the best way to handle a fax call failover.
14:45.55WIMPyMaybe it doesn't give yu enough time, but it surely increases the chances to receive that fax earlier.
14:46.04SamotUnless 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
14:46.34WIMPyNo, but that's a question of what kind of configuration options the ITSP offers.
14:46.35EmleyMoorHmmm... good point - may be able to disentangle this at the ITSP end
14:46.35[TK]D-FenderEmleyMoor, Of just answer the call and ditch if you don't want it to remain unanswered
14:47.38Samot1) Why are you sending busy FAX calls to a cell phone as the failover/CFWD Busy option??!
14:47.59Samot2) Why don't you block the call that keeps flooding you?
14:48.23Samot3) If you need to answer more faxes, do what TK said. Make more modems.
14:48.58EmleyMoorSamot: 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
14:49.50SamotSo you never get flooded from the same "fake" source twice?
14:53.38EmleyMoorSamot: No.
14:53.51*** join/#asterisk u0m3 (~u0m3@188.25.22.193)
14:54.07EmleyMoorAnyway, I've regrouped my numbers... so that should resolve it
15:00.07apb1963http://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)
15:03.41Samot<--- Reliably Transmitting (NAT) to 192.168.0.11:58840 ---> <-- ARe you really behind NAT on the same network?
15:04.50SamotCall-ID: bb93e88c37069908664f2ab4677e8939@0:0:0:0:0:0:0:0 <--- I would say that's a bad Call-ID
15:06.32[TK]D-FenderSIP/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.
15:08.52SamotThat is.....impressive.
15:09.14SamotConsider the fact he offers just about every codec known to man from Jitsi.
15:09.17apb1963Samot, 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.
15:09.39SamotNo, it will determine if there is NAT or not.
15:09.46apb1963Samot, I was also under the impression that the caller id was irrelevant.
15:09.47SamotBut if there are issues with network traffic..
15:09.57SamotPerhaps just set nat to NO to skip the detection.
15:10.02Samot??
15:10.10SamotCall-ID
15:10.13SamotNot caller id.
15:10.48SamotCall-ID is kinda one of the things that are used to track calls.
15:10.51SamotIn transactions.
15:10.54apb1963ok... no idea what I'm supposed to do about it.  Neither that nor the compatible codec issue, etc.
15:11.19apb1963Samot, oh, well then that's not important at the moment since right now I'm not doing any tracking that I'm aware of.
15:11.20SamotWell the codecs issue is, fix what codecs you allow on Asterisk.
15:11.25SamotNot YOU
15:11.34SamotThe actual SIP TRANSACTION
15:11.48SamotIt should have an IP
15:12.16SamotCall-ID should be in a valid URI format.
15:12.49SamotHonestly, go get a softclient that doesn't suck.
15:14.18SamotJitsi is like the PFSense of softphones.
15:14.24apb1963and what makes you think it sucks?
15:14.34SamotWell..
15:14.52SamotEvery time someone with Jitsi has an issue...
15:14.58SamotOK, not every time
15:15.09SamotBut like 98% of the time, it's due to Jitsi just sucking.
15:15.13apb1963jitsi, like asterisk... has many configurable options.
15:15.51SamotYou're sending a poorly formed Call-ID
15:15.55apb1963Well, so far I'm hearing to change asterisk config to presumably allow more codecs, but why that would be I don't know.
15:16.06SamotYour Asterisk side doesn't have codecs enabled proberly...
15:16.23SamotWhat codecs do you have enable on Asterisk?
15:16.38apb1963I've got ulaw, alaw, and... I think it's GSM but will have to doublecheck
15:16.42SamotOK
15:17.08SamotLook at that mess of a SDP section you have.
15:17.22SamotNone of the codecs Asterisk supports...
15:17.25SamotIs offered first.
15:17.38SamotThey are like the 8th or 9th offering...
15:17.42SamotIt never makes it to them.
15:17.59apb1963oh and g726
15:18.02SamotOK
15:18.12SamotSo make it so Jitsi is ONLY using those codecs.
15:18.19SamotWhy offer 25 codecs?
15:18.26apb1963I don't know which is the best one to use
15:18.26SamotIn random order...
15:18.37SamotTHEY HAVE TO MATCH
15:18.42apb1963that much i know
15:18.45SamotOK
15:18.51igcewielingwhen testing use ulaw or alaw
15:18.55SamotSo if you are using ulaw, alaw, g726 and gsm..
15:19.03SamotThat's ALL Jitsi should have ENABLED
15:19.22SamotYes, ulaw should always be first
15:19.34Samotor alaw, depending on your country of origin.
15:20.07Samotg711 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-FenderULAW / G711u /PCMu
15:22.09[TK]D-Fenderjust do it
15:25.41apb1963http://pastebin.com/spzpaA3h
15:26.27SamotNo.
15:26.32SamotTurn of DEBUG
15:26.36Samotcore set debug 0
15:26.45SamotThat 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)
15:36.04apb1963http://pastebin.com/pN1PiYRU
15:39.05[TK]D-FenderWhat kind of password do you have on it?
15:39.57apb1963"it"
15:40.58apb1963normal passwd?  There's also a TLS cert.  Not sure...
15:41.10apb1963It's setup to use the cert
15:41.24[TK]D-FenderLENGTH, CHARACATER MIX< ETC
15:42.24apb196310, normal ASCII chars.
15:42.43[TK]D-FenderShow 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
15:45.07apb1963http://picpaste.com/pics/Screenshot_from_2017-02-10_07-43-08-9HH0MyaU.1486741479.png
15:45.09*** join/#asterisk Tiffon (~name@unaffiliated/tiff0n)
15:47.28apb1963http://picpaste.com/Screenshot_from_2017-02-10_07-45-24-6Szd8UKa.png
15:48.57apb1963http://picpaste.com/pics/Screenshot_from_2017-02-10_07-47-47-l17Hdlen.1486741716.png
15:49.02[TK]D-FenderSomething tells me you should ahve the IP in the ID there
15:49.10[TK]D-Fendershouldn'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)
16:03.16apb1963http://pastebin.com/4Gf0hL52
16:04.12apb1963[TK]D-Fender, sans IP
16:08.50apb1963needs a slow break. brb
16:18.50[TK]D-FenderI 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)
16:23.50apb1963ok
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)
16:26.14apb1963http://picpaste.com/Screenshot_from_2017-02-10_08-25-00-s8DWRNa0.png
16:27.48apb1963http://picpaste.com/Screenshot_from_2017-02-10_08-26-56-toK8Dp6C.png
16:28.09igcewielingoh.  FreePBX.
16:28.39*** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net)
16:29.16apb1963[TK]D-Fender, that enough or what else?
16:29.33[TK]D-FenderWhere 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)
16:34.16wasanzyhi
16:34.26apb1963http://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...
16:34.30wasanzydoes PlayBack() support gsm format?
16:35.50[TK]D-Fenderapb1963, that is not the extension
16:36.12[TK]D-Fenderwasanzy, Yes
16:38.10apb1963http://picpaste.com/Screenshot_from_2017-02-10_08-33-27-h6RzvEq7.png sorry about that
16:39.56apb1963[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)
16:50.31axpdoes 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)
17:07.31apb1963axp, does it happen immediately on startup or when?
17:07.46apb1963axp, is there something that seems to trigger it?
17:08.34apb1963axp, crank up the logs... for starters
17:12.44axpapb1963, 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)
17:16.30apb1963axp, 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)
17:30.48igcewielingupdated iptables which filters out some of the common sip scanners, added a new rule today:   http://pastebin.com/2mKSw1nZ
17:30.56igcewielingmaybe it will help someone
17:31.24SamotNo siparmyknife?
17:31.53igcewielingI've not seen it attacking
17:32.01SamotOh
17:32.04SamotI don't wait for it.
17:32.14SamotIt's a known sip scanner UA
17:32.31igcewielingI could add it.
17:32.34drmessanoYeah I have it in mine
17:33.08igcewielingthe exact string "siparmyknife" is anywhere in the fist 1500 bytes?
17:33.46drmessanoIm SSHing into one of my boxes.. I wanna check my list against yours
17:34.54Samotfriendly-scanner|sipcli|sipvicious|VaxSIPUserAgent|VaxIPUserAgent|sip-scan|sipsak|sundayddr|iWar|SIVuS|Gulp|sipv|smap|siparmyknife
17:35.07Samotfriendly-scanner|sipcli|sipvicious|VaxSIPUserAgent|VaxIPUserAgent|sipscan|sipsak|sundayddr|iWar|SIVuS|Gulp|sipv|smap|siparmyknife
17:35.15SamotThat's what I have right now.
17:36.52igcewielingI called the extension they registered as and got tt-monkeys. 8-|
17:38.24drmessanoYeah Samot and I overlap.. he actually has a few I dont
17:38.28drmessanoSo I have nothing to add
17:39.02igcewielingthanks for the info
17:39.06drmessanoI have sip-scan with a hyphen
17:40.16SamotI check "friendly scanner uas" or "sip scanner uas"
17:40.50SamotOn 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)
17:43.05drmessanoVaxIPUserAgent and VaxSIPUserAgent?
17:43.21igcewielingThe 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.
17:44.11igcewielingdrmessano: you never know when you need a 20 year old minicomputer made by a company which does not exist anymore to run a softphone!
17:44.29drmessanoheh
17:44.49drmessanoSamot:
17:45.29drmessanosip-scan seems to be prevalent on google, not "sipscan".  Unless you have seen it, and want to add the hyphenated as well
17:51.34igcewielingHmmm...I wonder if I could use a useragent whitelist instead of blacklist.   We don't have a lot of different user agents
17:53.08drmessanoThe 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)
18:38.16Follow-meguys how do i change asterisk database location
18:39.06Follow-meDatabase: 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-FenderWhat database?
18:40.29Follow-meconnected to asterisk
18:40.56[TK]D-FenderWhat database?
18:41.01[TK]D-FenderDatabases don't connect to tAsterisk
18:41.05[TK]D-Fenderthat is backwards
18:42.30Follow-memysql database
18:42.56[TK]D-FenderMysql does not connect to Asterisk.
18:43.25[TK]D-FenderOther things use databases.  Databases don't use Asterisk
18:44.02[TK]D-FenderIf 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-FenderIf your * talks to a database there is going to be a blatantly obvious config file that specifies where it is connecting to it
18:53.39apb1963[TK]D-Fender, so I take it you've given up on my issue?
18:54.30drmessanoYou never answered him
18:54.35apb1963drmessano, how so?\
18:54.46drmessanoLooks like he asked for the EXTENSION
18:54.50drmessanoand you didnt provide it
18:54.54apb1963drmessano, yes... I did.
18:55.12apb1963of course at this point the file has expired
18:55.49apb1963http://picpaste.com/Screenshot_from_2017-02-10_08-33-27-h6RzvEq7.png
18:55.58apb1963which has expired
18:56.02apb1963so don't look
18:56.05drmessanoI dont know why people expire pastes
18:56.15apb1963why not?
18:56.27drmessanoSet it to a day or something.. not 30 mins
18:56.31*** join/#asterisk skywayskase (~skywayska@67.139.42.219)
18:56.39apb1963usually that's enough
18:56.56drmessanoAwfully presumptions that someone is sitting there RIGHT NOW EAGERLY waiting to help
18:57.04drmessanopresumptious
18:57.07apb1963if someone is paying attention it takes much less than that before it expires
18:57.24apb1963I don't paste until someone is paying attention
18:57.27apb1963as a general rule
18:57.43drmessanoLot of good it did you here
18:58.45SamotYeah, I went to click on it a while ago and it was expired.
18:59.38drmessanoI usually put serial paste expiration offenders on ignore
18:59.59apb1963http://picpaste.com/pics/Screenshot_from_2017-02-10_08-33-27-P3iOQD0y.1486753167.png
19:00.01drmessanoIf your paste is national security enough that you have to expire it in 30 seconds, you should have someone you can call
19:00.33SamotThat's not even the entire thing.
19:00.53apb1963was there something in particular you were looking for?
19:01.04drmessanoYeah the REST of it
19:01.26apb1963The whole page doesn't fit in one screenshot and I snapshotted the salient piece.
19:01.31SamotWe know.
19:01.40SamotYou have to take a couple.
19:01.45drmessanoThats your opinion
19:01.47drmessanoand its wrong
19:01.58drmessano"the salient piece"
19:02.01drmessanoKeep going
19:02.10apb1963I take shots all year long...
19:02.13SamotYou can either pastebin the details from sip_additional.conf
19:02.22SamotOr you can take more screenshots..
19:02.38apb1963What?  You're giving me actual detailed instructions as to what you want to see?  Say it isn't so Ethel!
19:02.46SamotDude.
19:02.54SamotWe asked to see the extension's settings.
19:02.56drmessano.....
19:03.00SamotYou gave us half of it.
19:03.02drmessanoYou posted.....
19:03.03drmessano"the salient piece"
19:03.09drmessanoLike WTF
19:03.15SamotAlso, we've told you numerous times. GO TO #FREEPBX
19:03.15drmessanoDont ask for help
19:03.22drmessanoThen post what YOU fucking think is important
19:04.07SamotDo not sit here and act like you haven't been given implicit instructions, when you have and have failed to follow them.
19:04.11drmessanoYou do this same crap when you're asked for debug
19:04.25drmessanoYou obviously dont know what is important.. so post what is ASKED
19:05.46[\\\]Another day, another dollar.
19:06.11apb1963http://pastebin.com/7mVVviQV
19:06.54SamotOK, I was expecting just the extension were are supposed to look at..
19:06.58SamotBut which one IS IT?
19:07.16apb19633304
19:07.20drmessanolol
19:07.50Samotavpf=no <--- ????/
19:07.58SamotYou are ATTEMPTING a VIDEO call
19:08.17SamotBut you don't have anything in there for a video call.
19:09.24SamotI have said this before, make your softphone ONLY use the codecs that you have allowed in Asterisk.
19:10.14SamotStop offering up garbage in the SDP that doesn't apply to anything.
19:13.35apb1963strange.. I could have sworn I added all video codecs in for *
19:14.16Samotsip show settings
19:14.20Samotprove it.
19:15.08apb1963well 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
19:15.24apb1963hence "could have sworn..."
19:15.50apb1963so... added those in... working on testing.
19:20.23drmessanowell?
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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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19:49.02foobar2017hi
19:49.49foobar2017I'm getting a load of 100% from mysql in my asterisk installation. Any help¡
19:49.50foobar2017??
19:50.10igcewielingfoobar2017: did it run out of disk space?
19:51.31foobar2017igcewieling: I have more then 400 GiB of free space.
19:54.07foobar2017*than
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19:59.16apb1963Thanks guys, I appreciate the help.  I have to pick this up later, have a nice day!
19:59.30SamotWe probably won't.
19:59.49drmessanoYeah not even close
20:01.04igcewielingfoobar2017: is it running in a VM?
20:01.21drmessanoI am starting to hate systemd
20:02.30igcewielingdrmessano: centos 7 ?
20:02.37drmessanoNah, Ubuntu
20:03.31drmessanoTrying to delete some lock files before I start a service
20:04.24foobar2017igcewieling: no, it's baremetal
20:04.52foobar2017igcewieling: it's running on debian
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20:28.06BrencoIncfoobar2017 dumb question but do you have any calls looping or sip scaner hitting you?
20:36.06foobar2017BrencoInc, igcewieling: I've just cleared out the cdr table.
20:36.12Alex_Bkashanyone compiled asterisk in cygwin?
20:36.20foobar2017And now it's normal.
20:42.05Alex_Bkashanyone compiled asterisk in cygwin?
20:49.29igcewielingedges away from the crazy person
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20:56.15[TK]D-Fender~polls
20:56.18infobot"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)
21:04.12johnny_|_~ask
21:04.12infobotQuestions 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.
21:09.03drmessanoand asking 6 minutes apart
21:09.08drmessanoWith no channel scroll
21:09.35[TK]D-FenderWouldn't believe the time I wasted in PM finding out what the project was in the first place...
21:09.56drmessanoROFL
21:10.05drmessanoDo I even want to?
21:10.51SamotOf course.
21:11.00SamotWhat was it?
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21:14.19drmessanoif (cash me ousside) echo "how bow dah";
21:14.32drmessanoThat's how you compile Asterisk in Cygwin
21:16.31drmessano[TK]D-Fender: TELL US
21:19.34[TK]D-Fenderdrmessano, PMM
21:32.20[TK]D-Fendercheckout time, BBIAB
21:32.57SamotNow tell me!
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22:13.28Sean-DerSamot: thanks again for the help last night, SIP provider said I wasn't responding to INVITES fast enough
22:14.04Sean-DerMy first OK took about 5 seconds, and they sent the same call through another friend
22:14.22Sean-Derwhich would get the 4xx thrown
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