IRC log for #asterisk on 20160704

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00:13.17*** join/#asterisk Prelude2004c (~Prelude20@iptv-management.zazeen.com)
00:14.38Prelude2004chey guys.. good day... some question.. i got opus going on my phone with asterisk but something is not right.. i setup codecs.conf to only have opus8 and its sample rate of 8khz and maxbitrate of 6000 , yet when on my phone i use full or wideband it still works... does it not force the system to use 8khz and respect the bw ? i mean the idea is to save bw and it doesnt' matter what i chooce its always the same bit rate
00:14.40Prelude2004cany ideas?
00:15.44shido6limit the codecs the phone accepts
00:15.54Prelude2004ci did.. i only allow=opus
00:16.13Prelude2004coh yes.. i did.. but still my bw stats still how 60kbs
00:16.17shido6show the call - the list of codecs are listed in the sip trace
00:16.26Prelude2004cdown is 60kbs and up is 60kbs
00:18.13Prelude2004ci select only 8khz and still in logs it says " codec_opus.c:203 lintoopus_frameout: [Encoder #98 (48000)] 960 samples, 1920 byte "
00:18.32shido6http://listening-tests.hydrogenaud.io/igorc/results.html
00:18.39shido6https://www.opus-codec.org/comparison/
00:20.44Prelude2004cyes i have seen all that
00:21.36shido6show the call , please :)
00:21.58Prelude2004ci need to be on the low size of opus where it auto adjusts to g729 or better quality when needed but if bw is good and can pickup then it uses better quality
00:22.23Prelude2004ctrying to get to all the logs :( .. its a lot , i set verbose at 255 and debug at 255
00:22.32shido6and your codecs.conf
00:22.51shido6no not ALL the logs - just the sip call would be a nice start [pastebin.com]
00:23.26shido6heh
00:24.43Prelude2004chttp://pastebin.com/raw/8YsbW3S8
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00:52.36Prelude2004cso any ideas?
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01:43.58[[thufir]]inbound calls show up in the Asterisk CLI:  http://pastebin.com/QitSuwjZ  but there's a problem with SIP retransmits.    Basically, start with turning on sip debug?
01:45.23[[thufir]]I can dial out, for what it's worth.  does that exclude NAT and firwall issues?
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02:46.11snadge"Is it possible to add H.264 for video on that trunk?" .. heh
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03:02.58boris_rrhhello, I am trying to enable ARI at my fresh asterisk installation
03:03.24boris_rrhtrying  curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/channels"
03:04.11boris_rrhand get 404 with  "message": "Resource not found"
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03:04.42boris_rrhwhen I try endpoints instead of channels everything works fine
03:04.59boris_rrhI also get events over websocket
03:05.05boris_rrhwhat might be wrong?
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03:26.15[TK]D-Fender[[thufir]], yes you should be looking with sip debug enabled
03:27.24[[thufir]][TK]D-Fender: thanks.  can I eliminate NAT and firewall issues because I'm able to dial out?  (using telnyx)
03:27.33[TK]D-Fenderno
03:31.07[[thufir]]ok
03:31.26[[thufir]]sipscanner.voicefraud.com can detect 5060 on my IP address.
03:35.18[[thufir]][TK]D-Fender: what positive tests can I run to check NAT and firewall?
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03:35.46[TK]D-FenderLook at the call
03:35.50mnathaniwhats the best codec for use with voip calls on a PC
03:36.05[TK]D-Fendermnathani, the biggest one the other side supports
03:36.09mnathaniand which provider / voip software supports such codec
03:36.32mnathaniare you familiar with voip.ms offerings
03:36.33[TK]D-FenderBasically the PST runs on G.711.  So that's where you hsould probably b
03:36.35[TK]D-Fenderbe
03:36.45[TK]D-FenderPSTN*
03:37.03[TK]D-Fenderso that'd be ulaw
03:37.49mnathanithanks
03:38.22mnathaniyour nick makes me curious - are you affiliated with any software release group?
03:38.51[TK]D-Fendernoe
03:38.52[TK]D-Fendernope
03:45.10drmessanoLies
03:45.34drmessano[TK]D-Fender wrote the keygen for PKZIP for the Kaypro
03:46.17[TK]D-Fenderis afraid the Osbourne he used to have will reaniamte and hunt him down...
03:47.24drmessanoAlso, he cracked the app that turns your iPad into a bathroom scale
03:47.30drmessanoWould you like a copy?
03:55.05[[thufir]]I can't seem to unregister telnyx:  http://pastebin.com/LxvPFd3J   because it doesn't show in "sip show peers".  How do remove it from the results of "sip show registry"?
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04:01.22[TK]D-FenderWell since it's been a half hour wirthout seeing the actual call as requested ..
04:01.24[TK]D-FenderI'm out.
04:01.27[TK]D-Fenderheads off
04:18.15[[thufir]][TK]D-Fender: wait one sec.
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04:19.26[[thufir]][TK]D-Fender: failed call http://pastebin.com/upwU3G8X
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07:16.14[[thufir]]inbound calls show up in the Asterisk CLI:  http://pastebin.com/QitSuwjZ  but there's a problem with SIP retransmits.    Sip debug on shows http://pastebin.com/upwU3G8X for a failed call to the DID from outside.  Why?
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09:30.52confused_recruitHello - is there someone I could have a quick chat with (community member is fine) re: Asterisk and the coding behind it?
09:32.45confused_recruitI'm a recruiter - I work for a company which uses the ASterisk system to underpin their phone systems - they have asked me to recruit a Release Manager to managage their development process.  Which is fine - although they can't answer me what language(s) Asterisk is written in.  It's Linux based for them, so I'm guessing PHP / Python or something similar?  Can someone confirm this for me?   Many thanks! :)
09:33.37WIMPyYou can't do telephony in a scripting language. It's all C.
09:34.33confused_recruitC - sweet, that works for me :)
09:35.55confused_recruitThey were going on about projects with AngularJS and CSS3 and I was trying to explain to them that's the front end, not the back end - but the HR lady didn't understand and just kept giving me words like JavaScript etc - so I thought perhaps it has some PHP/Python type code behind it.   But IIRC both are based on C anyway and C is a fine answer, thank you for your time :)
09:37.53WIMPyAsterisk dies not come with a frontend. So if it's about something like that, you have to find out what it it they're using.
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10:02.25[[thufir]]how do I know or see, from the dialplan, whether a number is allocated for incoming calls?  I have:  http://pastebin.com/ZP1E6Nd4   apparently I'm returning a 404 not found to the sip provider for my DID.
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10:59.03Samot[[thurfir]] Change your inbound to match the DID as it's sent to you.
10:59.26SamotAnd do you have your trunk setup properly?
11:01.38[[thufir]]Samot: one sec.
11:04.22[[thufir]]Samot: right now I'm using '_x.' =>          1. NoOp()  but have tried many patterns.  Outbound dialing works.  AFAIK the trunk is correct.  I'm reading about NAT and am leaning towards thinking that it's NAT at the root of this.  How do I run a SIP trace?
11:04.44SamotWhat's the context set to?
11:04.52SamotIn the trunk settings?
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11:05.53[[thufir]]Samot: inbound, line 22
11:05.55[[thufir]]http://pastebin.com/M3Nn41Vw
11:06.38Samotasterisk -rvvvvvvvvvvvv
11:06.46Samotsip set debug on
11:06.52SamotMake the call, watch for the output
11:06.58Samot~pb the results.
11:07.04Samot~pb
11:07.05infobotA "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
11:08.00[[thufir]]that's alota alota v's.  ok.
11:08.58SamotNo pms, please.
11:09.02SamotJust paste it here.
11:09.13[[thufir]]it shows my public ip, though.
11:09.18SamotSo?
11:09.35SamotIt's not like it's a secret to the world what your public IP is.
11:09.41SamotYou send it to everyone you talk to.
11:09.49[[thufir]]ok
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11:12.06[[thufir]]Samot: calling the DID:  http://pastebin.com/UfEyVUB7
11:13.07SamotNope.
11:13.14SamotThere is zero verbosity in there
11:13.19Samotcore set verbose 10
11:13.27SamotThat was the point of asterisk -rvvvvvvvvvvvv
11:13.37SamotTo put you in the proper verbosity for a debug.
11:14.07SamotDo it again, as told. Then pastebin the results.
11:14.47[[thufir]]I did all those v's.  asterisk -rvvvvvvvvvvvv  I checked just now.  I'll do core set verbose 10 also.
11:15.13SamotAnd call the DID from the outside
11:15.22SamotNot from your PBX.
11:15.28Samot<PROTECTED>
11:17.15[[thufir]]Samot:  http://pastebin.com/LwN4aBWg
11:18.10SamotHow are you making the test call?
11:18.28[[thufir]]skype
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11:18.48SamotTo the DID?
11:19.08[[thufir]]yes.   skype dial out to a phone number.  it rings.
11:19.23SamotThe call isnt hitting the system.
11:19.37[[thufir]]telnyx say that I'm returning back a 404.
11:20.05SamotThen fix your firewall
11:20.20SamotBecause the call is not hitting the pbx
11:20.37[[thufir]]what's weird is that I can dial out.  how do you know that it's a firewall?   you are inferring this because it doesn't hit the pbx?
11:21.45SamotYes.
11:22.05[[thufir]]is there some positive test I can do?  sip trace?
11:22.12SamotAlso, i doubt telnyx is behind NAT
11:22.23SamotYou just did one
11:23.32[[thufir]]ahh, ok. thanks.
11:23.45SamotAnd dialing out is outbound traffic. I doubt you have outbound rules in your firewall.
11:24.24[[thufir]]dmz?
11:24.29SamotMake sure you have it set up with the proper nat rules
11:24.31SamotNo.
11:26.53[[thufir]]which NAT rules? listen on port 5060 (?) and forward to ip address x.x.x.x.8 for Asterisk?
11:27.22Samot5060, 10000-20000 on UDP for them.
11:29.39[[thufir]]5060 is tcp/udp?  10000-20000 is udp only? ok.
11:30.34SamotIt's all UDP.
11:30.50[[thufir]]ok.  no tcp then.  I'll fix that.  thanks.
11:35.50[sID]Samot: Hi, I wrote and posted about a2 me that is From the UI
11:36.18SamotOK.
11:37.01[sID]Only probably lie because I do not see anything like that :)
11:43.13SamotAre you saying they replied?
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12:11.07[[thufir]]I'm looking at http://the-asterisk-book.com/1.6/einleitung-regex.html  but don't see how match a number starting with a + and then NANP?
12:12.02WIMPyJust add a +1 to the pattern.
12:12.24WIMPyOr are you really using REGEX?
12:13.24[[thufir]]change this:  '_1NXXNXXXXXX' => 1. NoOp()   to    '_+1NXXNXXXXXX' => 1. NoOp()    ? easy enough.  Not trying to use regex.
12:13.48WIMPyYes, just that.
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12:46.10[[thufir]]telnyx tells me that I'm sending many, many 200 OK's but that otherwise it looks fine for incoming calls to the DID.  It rings, I answer, then hangup.  I forwarded 5060 UDP and 10000-20000UDP to the internal IP address for the Asterisk box.
12:46.40[[thufir]]The messages are that a critical packet wasn't received, reference https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions.
12:47.56TandyUKI get that a LOT when using TCP/TLS for handsets
12:48.09TandyUKif youre using udp exclusively, not sure whats wrong
12:48.24[TK]D-Fenderthat makes no sense for TCP
12:48.39[TK]D-FenderTCP takes care of retransmission
12:48.43TandyUKit does, when the tcp session has dropped
12:48.59[TK]D-FenderNormally shouldn't happen
12:49.13TandyUKit still tells asterisk it fails, so once asterisk tries again, but now theres no tcp session to send the packet down, and a tthat point asterisk realises the phone is unavailable
12:49.22[TK]D-Fenderbut I suppose if your router gets flooded it may rank things as idle and allowed to flush
12:49.31[TK]D-FenderBetter routers let you set rules for that though
12:49.40TandyUKI had my voip platform provider modify the code to make it a notice not a warning as it was just flooding our logs
12:50.17TandyUKits not the router lol, its when the connection (from the other end) has been interrupted, eg customer broadband went down, or mobile lost 3g data, etc
12:50.42TandyUKpossible customer end routers not being conservative enoug hwith states, but thats all out of our control
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12:57.41[[thufir]]here is the sip trace for an inbound call.  outbound calls are fine.  again, I setup port forwarding, but "guess" that it's NAT and/or firewall problem.  http://pastebin.com/P76z3zzZ
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13:00.10[TK]D-Fender<--- Transmitting (NAT) to 192.76.120.10:5060 --->
13:00.24[TK]D-FenderVirtuall no provider is behind NAT.  Fix your peer
13:00.42[TK]D-FenderContact: <sip:+16044494243@192.168.1.8:5060>
13:01.09[TK]D-FenderAnd you seem to have been sending the wrong IP for contact
13:01.33[TK]D-FenderSo it's looking like you messed up your otehr basics in [general]
13:01.58[[thufir]]pardon:  http://pastebin.com/HjeG086P
13:03.26[[thufir]]the provider isn't behind NAT, but I am.  the wrong IP for contact for ____?
13:03.54[TK]D-FenderFIX YOUR TRUNK
13:04.03[TK]D-Fender<[TK]D-Fender> Virtuall no provider is behind NAT.  Fix your peer
13:04.55[TK]D-FenderAnd you have still screwed up your general NAT settings.  You are transmitting your PRIVATE IP as the Contact: and not your PUBLIC ONE.
13:05.08[TK]D-Fender225-237
13:05.09[[thufir]]ah.
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13:09.44[[thufir]]nat=never ?
13:10.04[TK]D-FenderNO
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13:30.17[[thufir]]I set externip in the peer to transmit my public IP as the contact.  correct?  nat=force_rport,comedia because I'm using NAT.  correct?
13:30.40[TK]D-Fenderexternip is NOT a peer-level parameter
13:30.45[TK]D-FenderYou're doing it wrong
13:31.04[TK]D-Fender<[TK]D-Fender> <[TK]D-Fender> Virtuall no provider is behind NAT.  Fix your peer
13:34.13sharadI have two asterisk boxes one is using as voip gateway another is using as media server. when i make a call through my SIP phone which is registered on media server, using the voip gateway server to the PSTN network and PSTN guy hold the call then i get default MOH from my VOIP gateway How is it possible to get this MOH from the media it self
13:34.16[[thufir]]externip goes under [general] then.
13:49.34[[thufir]][TK]D-Fender: thanks, it works.  I don't understand, why, though.  what does [general] do?  by putting externip in general, that...sends my external IP?  but why can't I put it in [telnyx]?  what's the distinction?
13:50.05SamotBecause it's not a peer level setting.
13:50.05[TK]D-Fenderbecause you don't get so specify the advertieed address PER PERR
13:50.55[[thufir]]ok
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13:51.10[TK]D-Fender* will send either the interface's IP, or what you set as external
13:51.36[TK]D-Fenderbased on your localnet
13:51.44[TK]D-Fenderwhich is ALSO not a peer setting
13:51.49[TK]D-Fenderread the SAMPLE CONFIG
13:51.53[TK]D-FenderThe local matters
13:51.58[TK]D-Fenderlocation*
13:53.49filewobbles
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14:40.40ronybeckHi all.  I have a problem with a SIP trunk on Asterisk.  When Asterisk sends an invite, the SIP Trunk provider sends back a 401 unauthorized (authentication challenge).  Asterisk should now answer the authentication challenge.  But it doesnt.  Who do I fix that?
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15:00.47mirela666ronybeck, did you try setting trunk with insecure=invite ?
15:07.46mirela666maybe even you should remove the insecure section
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15:10.28ronybeckMilenco: I did
15:10.31[TK]D-FenderNot relevant
15:10.32ronybeckI just tried removing it
15:10.38ronybeckMakes no difference
15:12.07ronybeckIt is like Asterisk doesn't see the 401 unauthorized message
15:12.26ronybeckEven though it prints it in the asterisk console when I enable sip debugging
15:13.16[TK]D-FenderYou should probably be showing us the debug & config masking only the secret at this point....
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15:15.46ronybecksure
15:16.42ronybeck[mdc_trunk_conf-1]
15:16.42ronybeck; phonestar - friend
15:16.42ronybecktype=friend
15:16.42ronybeckcontext=from-trunk
15:16.42ronybeckdefaultuser=YYYYYYYYYYY
15:16.42ronybecksecret=XXXXXXXXXX
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15:16.45ronybeckhost=ps1.voipgateway.org
15:16.46ronybecknat=force_rport
15:16.50ronybeckprogressinband=yes
15:16.52ronybeckqualify=no
15:16.54ronybeckfromuser=YYYYYYYYYYY
15:16.56ronybeckThat is the trunk configuration
15:17.06mirela666use the pastebin ~pastebin
15:17.11ronybeckSorry
15:17.12mirela666~pb
15:17.12infobotA "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://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
15:17.16ronybeck~pastbin
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15:22.50ronybeckhttp://paste.lisp.org/display/319814    <----------------------- here is the TCP dump from the invite
15:23.01[TK]D-Fenderno providers should be behind NAT
15:23.42ronybeck[TK]D-Fender: are you referring to this:  nat=force_rport
15:23.48[TK]D-Fenderyes
15:23.57ronybeckWhat does it do?
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15:24.14[TK]D-Fenderset's who to trust for ports & IP's
15:24.23[TK]D-Fenderyou should be taking what they tell you at face value
15:25.09ronybeckJust remove that line then?
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15:26.09[TK]D-Fendernat=NO
15:26.27[TK]D-Fenderand type=peer
15:26.49ronybeckBut with type=peer I can't receive calls right?
15:26.57[TK]D-Fenderincorrect
15:27.40ronybeckok
15:28.06[TK]D-Fendermake the changes and show a new call with * SIP debug (not external)
15:28.45ronybeckok
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15:52.51mirela666ronybeck, type=friend never use unless you know what you are doing, it will create both user and peer and it can be very insecure if you don't set a password
15:53.32ronybeckok
15:53.35ronybeckUnderstood
15:53.44ronybeckHow do I call "*"?
15:53.59ronybeckI know how to do this stuff on Freeswitch.  Asterisk is a new world to me.
15:54.32[TK]D-Fender<ronybeck> How do I call "*"? <- as in?
15:54.59ronybeck<[TK]D-Fender> make the changes and show a new call with * SIP debug (not external)
15:55.15[TK]D-FenderAsterisk CLI
15:55.16[TK]D-Fendersip debug
15:55.23ronybeckah
15:55.26ronybeckunderstood
15:55.45ronybeckI thought you meant don-t call external.  But you mean don't use an external tool
15:55.47ronybeckright
16:01.45ronybeckhttp://paste.lisp.org/display/319817  <------------ here is the logs
16:02.50ronybeckThat is with the changes suggested so far
16:03.19[TK]D-FenderThat is a pretty old release, you should upgrade
16:03.32[TK]D-Fender<PROTECTED>
16:04.04ronybeckIt is part of a software suite called MobyDick.  This is the latest supported version
16:05.10ronybeckOtherwise I would have taken the latest version :-(
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17:23.00drmessanoWow
17:23.02drmessanohttps://www.pascom.net/en/pricing/
17:28.03filehmm?
17:28.12fileoic
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20:21.48_abc_Hello. Can anyone suggest a iax2 phone client for linux, non gnome, non kde? Preferrably a statically compiled one?
20:21.59_abc_sflphone seems to be one but no idea what libs it requires.
20:23.05_abc_Strangely sflphone is advertised as SIP and something else (its own protocol) but the debian docs claim it is IAX2 compatible too.
20:27.08_abc_http://www.voip-info.org/wiki/view/IAXComm is gone gone?
20:36.47_abc_Basically I can't find a iax2 client for linux which is included in any distribution besides sflphone.
20:38.47_abc_https://sourceforge.net/projects/kiax/ is still around but not included in any distribution as far as I checked
20:41.21_abc_Ahh kiax is static compiled. Nice.
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20:45.32_abc_Package verified and installed: lies, it is not static. Dll compiled and depends on a lot of distribution specific libs
20:46.10_abc_Package name is kiax-2.1-beta1-linux-static.tar.gz
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22:37.07drmessanoIAX Softphone = Uncommon
22:37.18drmessanoIAX Softphone for Linux = Less Common
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