IRC log for #asterisk on 20170311

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01:21.25*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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12:52.13sarthorHI, Current Asterisk Version: 13.9.1. here is error https://paste.ubuntu.com/24157459/ I can connect in local lan but can not connect via Internet, Please help need for guidance. thanks in advance.
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14:35.39CryptoManiacCan Asterisk run without dahdi if using just sip channels? Or dahdi has to be installed regardless of channel use?
14:37.32WIMPyNot needed.
14:38.14CryptoManiacWIMPy: Thanks
14:39.32CryptoManiacAnother issue I'm having... When dialing from my external softphone to one of the extensions which is a desk IP phone on the internal network, there i sno audio. I am  pretty sure my NAT settings are correct because I am able to use the external softphone to dial into other applications and echo test has two-way audio.
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14:45.32SamotThere's no audio completely? No audio on one side? Or one-way audio?
14:46.28CryptoManiacno audio completely
14:46.52SamotSo neither side can hear each other/
14:47.05CryptoManiacCorrect
14:47.16SamotShow a SIP debug on a call this is happening with
14:47.19Samotasterisk -rvvvvvvvvvv
14:47.22Samotsip set debug on
14:47.26CryptoManiacok
14:47.30CryptoManiacwill pastebin it shortly
14:47.35Samotpjsip set logger on <-- if you are using that
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15:05.28CryptoManiacSamot: May I PM you the pastebin link? (as it contains my public IP's)
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15:11.55SamotI guess but then others cant see it and make assessments..
15:12.23WIMPyThe bots have probably already found you anyway.
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15:24.52SamotWell?
15:29.47CryptoManiacApologies guys
15:30.03CryptoManiacI do not mean to waste your precious time
15:30.22CryptoManiacIt's against IT department policy to post our public IP's. :-(
15:39.05SamotI said you could PM them.
15:39.08CryptoManiaccame up wtith a solution, i replaced the public IP with <PUBLIC IP>
15:39.14CryptoManiachttp://pastebin.com/k7U2g0t6
15:40.27SamotDoes 11 even support Opus?
15:41.44SamotMake all sides of the call ulaw.
15:42.01SamotLike your Bria client, which is only offering Opus from what I can see.
15:42.45CryptoManiacah...
15:44.17Samot#CodecsMatter
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15:45.31CryptoManiacI have disallow=all under my sip user settings, and then allow=ulaw allow=alaw allow=g729
15:45.43CryptoManiacbut I guess the problem is that Bria is only offering Opus
15:45.55CryptoManiacTo be honest I don't even know what Opus is. I'm a Voip newbie
15:45.56CryptoManiac:-)
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15:51.22CryptoManiacSamot: I'm afaraid that was not the problem, I disabled Opus in Bria and set ulaw/alaw as priority followed by g7211 g729 and gsm but still the problem persists
15:51.46SamotThere is no g7211
15:51.57SamotYou really don't need g729 or gsm.
15:52.12Maliuta_Samot: it's the upgraded g711 ;)
15:52.20Maliuta_CryptoManiac: http://opus-codec.org/
15:53.26CryptoManiacMaliuta_: Thanks
15:53.45CryptoManiacSamot: Noted
15:53.58CryptoManiachowever, still no audio
15:54.52Maliuta_Samot: looks like opus is better than g729, I would probably use opus over g729 - if my devices supported it
15:55.09Maliuta_CryptoManiac: is NAT involved anywhere?
15:55.57Maliuta_CryptoManiac: and if you are terminating a trunk on opus you might have to transcode to a codec that your SIP devices can handle
15:57.10CryptoManiacMaliuta_: Yes there is NAT. There is two-way audio between the Bria softphone client and asterisk, for example when I dial the demo extensions 500 an 600. However when I dial a desk IP phone which is on the internal network there is no audio at all...
15:57.13WIMPyIs there any ITSP that supports opus?
15:57.46Maliuta_CryptoManiac: try turning of reinvite for the softphone
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15:58.13Maliuta_or just turn it off fullstop
15:58.31Maliuta_for smaller setups it's worth keeping * in the loop
15:58.52Maliuta_there are even cases for keeping * in the loop in larger setups
15:59.03CryptoManiacMaliuta_: I actually want Asterisk to be in the loop
15:59.30SamotThen you want directmedia/reinvite off.
15:59.42Maliuta_Samot++
15:59.55Maliuta_WIMPy: if there is not we can create one!
16:01.51*** part/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew)
16:02.15SamotNo ITSP I know of openly admits they support Opus.
16:03.18CryptoManiacSamot: I turn that off by canreinvite=no ?
16:03.20SamotBut then again a lot are "ambiguous" when it comes to the RTP media.
16:03.34Samotcanrevinite= was replaced by directmedia=
16:03.46SamotBut yes, they should be no
16:03.52CryptoManiacok
16:04.46Maliuta_Samot: it really is a wideband codec, so passing to/from PTSN is going to create degredation. Probably a good solution for VoIP only, if it has to go to the PTSN use g729 or g711
16:05.01SamotI know what Opus is.
16:05.10SamotYes.
16:05.25SamotI know it's why a lot of providers do offer anything outside of g711 or g729
16:05.40SamotAnd don't do STRP/TLS.
16:05.47Maliuta_that and laziness
16:05.49Samots/STRP/SRTP/
16:05.56SamotPerhaps..
16:06.08SamotOr the fact it will require a lot more processing power.
16:06.18SamotPuts a lot more load on the systems.
16:06.19Maliuta_think of all the time/money they save by not implimenting it
16:06.32Maliuta_CPU is cheap these days
16:06.33SamotOr the fact that a lot of the PSTN doesn't use it.
16:07.37Maliuta_here in .au pretty soon old school PTSN will be dead, it will be all VoIP over the NBN (National Broadband Network) - even if 99.999999% of the populus don't know that
16:07.40SamotWhen you're taking thousands of calls per second and have to transcode or decrypt/encrypt them all....
16:08.22Maliuta_most mobile networks are VoIP inside, and I've even heard that most copper PTSN is VoIP from the exchange now
16:08.47SamotA lot of carriers are IP based.
16:09.03SamotBut the entire PSTN backbone is not and will not be for some time.
16:09.16Maliuta_depending on where you are
16:09.48SamotWhich means if an IP based carrier wants to exchange calls from a non-IP based carrier, what happens?
16:10.14Maliuta_when you sign up to the NBN here your phone line (if you have one) goes with it. Comes out of an RJ11 port on the their termination box
16:10.39SamotYeah that's not uncommon.
16:10.44SamotATT, Comcast, they all do it here.
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16:11.59Maliuta_Samot: I'd have to look into it, but I think all of our international calls are going out via fibre links (probably IP based) I guess the problem is always going to be "the other end". Once it's out of your network you have NFI what tech is being used to deliver it
16:12.52Maliuta_Most ISP's have been doing it on DSL connections for a while. The just have VoIP enabled modems they give people. Most don't even know it's VoIP
16:13.03SamotI know.
16:13.06SamotThat's how it is now.
16:13.24SamotVoIP is becoming the standard delivery method to the end user..
16:13.30SamotThe *end user*
16:13.41Maliuta_I tried to trunk directly with one a while ago (because friends of mine knew it was a VoIP line) and the ISP and firewalled it all off to only their servers
16:14.24Maliuta_end users can bite me. They cause more problems than they're worth :D
16:15.05Samot"If it wasn't for all my end users, my business would be awesome"
16:15.14Maliuta_devs are enough of a PITA for me :)
16:15.32Maliuta_although I work with only DBA's ATM
16:15.42Maliuta_they are almost as bad as devs
16:16.13Maliuta_Samot: end users are SEP
16:17.08Maliuta_I don't do that shite anymore, cause me to develop a drinking habit. Also the primary cause of my leukaemia :D
16:17.26Maliuta_but the caffiene helped treat that
16:17.31SamotEnd users are everyone's concern at a provider.
16:18.23Maliuta_Not if you work far enough up the food chain. Then they are just accounts, or DB entries :)
16:18.32Maliuta_my concern is servers and services
16:18.48Maliuta_if they are up and running properly it's SEP
16:19.07SamotSo basically, the end users are your problem.
16:19.27Maliuta_only the ones I work with
16:19.28SamotServers or services don't work, that impacts the end user and means you have to get involved. So they are your problem.
16:21.26Maliuta_"CC all mail from these wordpress installs to this address - I already get them from another address, but add this one anyway" ... "it's not working I haven't seen an email from those addresses" /me provides logs proving delivery of said email, user doesn't look at his filtering rules that are removing duplicate messages
16:22.22Maliuta_I only do internal systems, client systems aren't my problem. Our company specifically doesn't offer that service. Only DB design/optimisation (for mysql/mariadb)
16:23.04CryptoManiacFound this in my console verbose but that URL is not valid
16:23.06CryptoManiac[Mar 11 19:16:40] WARNING[4161]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission e360524b5c1941bf9ef0236b28ed3821 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
16:23.33Maliuta_occaisionally I might have to look at a client system to tell the DBA where the bottle neck is, but then it's up to the client to fix it
16:23.44Maliuta_CryptoManiac: firewall issue somewhere?
16:24.02SamotThat URL works fine.
16:24.24CryptoManiacMaliuta_: Have opened SIP port TCP/UDP and RTP ports 10000 to 20000 as configured in rtp.conf
16:25.06SamotUDP, right?
16:25.27SamotYou have all those ports open on UDP?
16:26.17Maliuta_I limit mine to about 10 ports for my small install. If I am doing a 30 phone setup for a campaign I open more for them - but never that many
16:26.40Maliuta_also tend to firewall them off so that only the ITSP can access them
16:26.54Maliuta_from the outside - inside is a different matter
16:27.32Samot10 RTP ports?
16:27.51Maliuta_CryptoManiac: you'd need to see which device that was failing with to see. Just having the ports open might not be enough
16:28.12Maliuta_Samot: not making a massive number of calls from here
16:28.26SamotYeah, 5 is the limit.
16:28.42SamotRTP requires two ports per call
16:28.57Maliuta_Samot: like I said, when I do larger installs I open more
16:29.04Maliuta_yup
16:29.36Maliuta_which is why I open 10 - allows 10 concurrent calls from here
16:29.43Maliuta_sorry 5
16:29.58SamotIt also means that people can figure out what RTP ports you are using.
16:30.31Maliuta_firewalled off to the ITSP
16:30.45Maliuta_not allowing just anyone to trunk into my systems
16:30.58Maliuta_ may be crazy, just not _that_ crazy
16:31.04SamotOK
16:32.17Maliuta_if someone can look at the number of filetered/rejected ports on a scan of my system and figure out which ports are for RTP then they have all the network foo
16:32.27SamotNot like anything is actually listening on the ports until they are in use.
16:32.44Maliuta_this is also true
16:33.08SamotThe other side doesnt even know what ports until you tell them.
16:33.36SamotThere is no unestablished, new incoming RTP requests
16:34.08Maliuta_someone at an ISP level could probably look at the netflows and tell
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16:34.30SamotRTP ports arent known by the provider until you tell them what ports to use.
16:34.42SamotThey don't "listen" for anything.
16:34.42Maliuta_traffic patterns can tell you alot
16:35.05SamotRight, the ports are assigned randomly based on the range.
16:35.18SamotSo 10000-20000 means you have 10,000 ports to randomly choose from.
16:35.22Samot10 means just that.
16:35.36SamotWhich limit makes it easier to guess patterns?
16:35.38Maliuta_good point
16:35.49Maliuta_yeah yeah
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16:35.56Maliuta_I see what you're getting at
16:36.17Maliuta_throw in some random skype/IM activity and it gets harder
16:36.27Maliuta_but still
16:36.29SamotOpening the RTP ports in your NAT rules is more for making sure NAT isn't lost once the audio stream is established.
16:36.51SamotI rarely have to open SIP ports or RTP ports on my routers.
16:37.14Maliuta_I don't NAT my * servers - I got away from that habit ages ago
16:37.29SamotI'm talking about on premise systems.
16:37.47Maliuta_and I don't allow directmedia, so everything inside (on private IP's) goes to * to be sent out
16:37.53SamotThis also includes phones.
16:37.58SamotI don't allow directmedia either.
16:39.30Maliuta_I try not to run VoIP through NAT if I can avoid it
16:39.49SamotClient premise systems can't avoid it.
16:39.53Maliuta_most of the installs I do are to systems that have a real static IP, run the * server there
16:42.17Maliuta_RTP is a bigger issue on the Big Blue Button servers we use for training than it is on any of the * installs I do
16:43.01SamotI'm not sure what that means.
16:43.15Maliuta__if_ we move to a matrix server to replace our company IRC channel it could be an issue there if people want to do voice/video via that
16:44.02Maliuta_Big Blue Button is bit of software we use that allows over internet training. With video/voice and desktop sharing
16:44.46Maliuta_it's normally used as part of a moodle install. But we don't need all the moodle foo (although for one the bosses other companies it might be needed soon)
16:46.26Maliuta_BBB uses RTP for all kinds of things, so having it configured properly, and the firewalls setup to allow it through properly (including the fact the some of the client ends are going to be comming through NAT) is an issue on those servers.
16:47.11Maliuta_approaching 3am - I should probably get some sleep
16:47.34Maliuta_tonnes of code to write, and other work I need to do later today
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17:24.14CryptoManiacSamot and Maliuta_ thanks a lot for your help. Fixed my NAT issues...
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21:37.45zendhi! Can someone help me to understand this Warning in Asterisk, Abnormal 'Gosub(ackcall,~~s~~,1(61,00000000056,0437767327))' exit.  Popping routine return locations.
21:39.19dan_jzend: Do you have a 'return' at the end of the gosub?
21:40.00zenddan_j: yes I do
21:40.30dan_jzend: http://lists.digium.com/pipermail/asterisk-dev/2013-September/062428.html
21:40.38zenddan_j: What happens is callee hangs up, while in the macro. I'm doing this https://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall
21:41.06dan_jIf they hang up inside the macro, then return won't get run. I believe you can ignore this warning if that is the case.
21:41.46CryptoManiacwhatever happened to TK Defender?
21:41.57dan_j[macro-ackcall] in the link you posted does not have a Return
21:42.38zenddan_j: i modified mine to have the return.... i'm using second example
21:43.12zenddan_j: THE AEL version
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21:43.52zenddan_j: do you think it ever leads to any imbalances in asterisk stacks & leads to Unexplained behaviour...
21:44.08dan_jI dont know AEL. But if the HANGUP gets executed, then it doesnt hit the RETURN. I dont think it's an issue.
21:44.40dan_jMaybe ask again during the week when this channel is more active with people who are more 'in the know'
21:45.44zenddan_j: Ok , thanks
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22:42.23igcewielingI suspect using a goto to jump out of a gosub (or macro) would cause that sort of error.
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