IRC log for #asterisk on 20160722

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01:23.42raspberrypifanso how can i use wireshark to check which ports are being blocked by my nat/firewall. If i dmz my computer i get audio both ways but otherwise it seems i am missing some ports
01:26.00freebsjust open the correct ports to the pbx
01:26.10freebsrtp 10000-20000
01:26.17freebsnormally
01:29.21raspberrypifanwell those are the ports that are open and more actually
01:29.25raspberrypifanbut when i do i get one way audio
01:29.39raspberrypifani only manage to get two way audio if i dmz my laptop with the softyphone
01:30.02raspberrypifanalso the sip port on the softphone is a very random port number
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01:30.07freebssounds like natting issue .. dmz is taken out of NAT
01:30.18raspberrypifanyea tho the router i have is a ddwrt router
01:30.40raspberrypifanthe weird part is the abnormally high sip port numbers
01:36.07raspberrypifanits also weird because i have asigned the rtp ports from 10,000 to 20,000 in the file but it uses 8000
01:36.09raspberrypifanlike wtf
01:41.49ChannelZYou only have control of where the remote side sends RTP to you (a range via rtp.conf) - those need to be port-forwarded to your asterisk.
01:42.18ChannelZThe remote end can request you send RTP to them on just about any port; So you have to be pretty lax in what you allow _out_
01:45.19raspberrypifanah i see
01:45.44raspberrypifanbut it just seems like it doesnt listen to what im asking from rtp.conf
01:50.56ChannelZIt should unless you didn't reload
01:54.05ChannelZBut again the other trick is you need to port-forward that range (and 5060) to your asterisk box's IP behind the firewall.
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01:58.42lukeescudeHello all - New installation of Asterisk 13.8, and it crashes every time UDPTL begins. Any ideas?
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02:10.42raspberrypifanwell my asterisk box is actually hosted on a cloud somewhere
02:10.44raspberrypifanwith open ports
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02:33.43lukeescudeNew installation of Asterisk 13.8, and it crashes every time UDPTL begins. Any ideas?
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08:16.33Rasputin3711Does the bug https://issues.asterisk.org/jira/browse/ASTERISK-16115 exist in the 13.9.1 release?
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09:33.19fileRasputin3711, yes. you can see based on the Target Release Versions where it was fixed.
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11:01.32UncleKiwihey is there anyone here with a voip business ?
11:02.00TandyUKyup lots of us
11:02.08sekilno usually only lawyers and economists
11:02.14UncleKiwinice
11:02.18TandyUKanyone know how to get 'sip show peers' to show which protocol a given phone is using
11:02.21UncleKiwii want to learn something about marketing
11:02.23TandyUKeg TCP / UDP / TLS
11:02.29UncleKiwiwhat works best
11:02.39TandyUKword of mouth is by far the best
11:02.43TandyUKive never advertised
11:02.45sekilUncleKiwi: like Cisco sucks?
11:03.03UncleKiwiaha
11:03.17UncleKiwiword of mouth is good
11:03.23UncleKiwia website with good seo is good
11:03.37UncleKiwiphoneing around and getting meetings is good
11:03.47UncleKiwibut i need to get things moving more
11:03.57UncleKiwihow do i take it to the next level
11:04.17UncleKiwii have been thinking about facebook advertising
11:04.19TandyUKgo to more business networking meetings
11:04.38TandyUKintroduce incentives for your existing customers to refer you to their contacts
11:04.45UncleKiwii got invited to a business networking meeting
11:04.52UncleKiwinext week
11:04.52TandyUKi do like 4 each week
11:05.07TandyUKbusiness networking is about people getting to know and trust you
11:05.26TandyUKone particular networking group im in is now responsible for ~30% of my income
11:05.41UncleKiwiyeah the owner of callcenter i setup invited me to one he is holding
11:06.06UncleKiwiwhat do you mean by these groups
11:06.10Rasputin3711https://www.amazon.com/How-Start-VoIP-Business-Six-Stage/dp/6094088301/ref=sr_1_1?ie=UTF8&qid=1469185558&sr=8-1&keywords=voip+business
11:06.30TandyUK[12:06] <UncleKiwi> what do you mean by these groups << huh?
11:06.41UncleKiwibusiness networking
11:07.07UncleKiwiis it like somthing that is common and people pay to belong to these things
11:07.09TandyUKhttp://www.bni.co.uk/what-bni
11:07.10UncleKiwior is it informal
11:07.20TandyUKthats one group i am a member of (actually vice-president of my chapter)
11:07.24TandyUKvery formal
11:07.28UncleKiwiah  ok
11:07.30TandyUKvery structured
11:07.36UncleKiwiyes we have one in our city
11:07.55TandyUKI can sort you out an invite to that if you like
11:08.05UncleKiwiim a little worried the leaders will pry into the inner workings of business and share my knowledge
11:08.08TandyUKBNI has chapters everywhere :)
11:08.18TandyUKwell thats kinda  the point
11:08.26TandyUKthey get to understand how you work, and trust you
11:08.29UncleKiwito my competition
11:08.39TandyUKand then they are putting their own reputation on the line when referring you to their contacts
11:08.50TandyUKwhy would they have anythign to do with your competition?
11:09.07TandyUKBNI is a lockout group - there can only be one VOIP provider in a chapter
11:09.17UncleKiwimaybe im not the only voip guy going to the meetings
11:09.26TandyUKso if you joined your chapter as the voip provider, no other voip provider is allowed to go
11:09.33TandyUKthats how it works
11:09.52UncleKiwii think the one in our city has different groups
11:10.05TandyUKyou join, you lock out your competition from even going as a visitor
11:10.11UncleKiwiso they keep the voip guys appart
11:10.18TandyUKbni works the same way in every chapter
11:10.39UncleKiwiare you in the UKJ?
11:10.41UncleKiwiUK ?
11:10.43TandyUKmy town we have 2 bni chapters, one meets in the morning, and one meets at lunchtime
11:10.54TandyUKmy nick means you really need to ask that?
11:11.27UncleKiwiaha
11:11.38TandyUKhonestly, go visit your local bni chapter, and see how it works for yourself
11:11.40UncleKiwiok so bni is valuab;e
11:11.43UncleKiwiok so bni is valuable
11:11.47TandyUKtheres lots of other networking groups that arent bni too
11:11.48UncleKiwiim i have been thinking about it
11:11.50TandyUKhell yeah
11:12.04TandyUKits more valuble than any other networking i do
11:12.33UncleKiwithe one here they make ya wake up early and have breakfast with them all
11:12.37UncleKiwionce a week
11:12.49TandyUKyeah most bni meetings are first thing, so you meet up once a week before work
11:12.59TandyUKmy chapter is more civilised - we meet at lunch :)
11:13.03UncleKiwiok maybe i need to do it
11:13.31UncleKiwiso this is by far the biggest way you get clients
11:13.43TandyUKfor some hard stats, it costs me about £1500/yr to be a member, including travel, parking, lunch, and the 2 hours a week i lose
11:14.09TandyUKSo far in 2016, ive had £17000 turnover as a direct result of being a member
11:15.10UncleKiwihmm
11:15.12UncleKiwiok
11:15.21TandyUKand as a service provider, we gain the most too
11:15.31UncleKiwiok
11:15.33TandyUKas just because someome leaves bni, doesnt mean they stop using your services
11:15.49UncleKiwitrue
11:15.52TandyUKand they'll keep referring you
11:17.13TandyUKI supply telecoms, broadband, voip, it support etc to around half the members of my chapter atm
11:17.15UncleKiwithanks TandyUK im a bit of an introvert and I know some of these business owners are such confident loudmouths
11:17.24TandyUKyeah so was i 2 years ago
11:17.32TandyUKbni has helped me massively on that front too
11:17.36UncleKiwiok
11:17.43TandyUKi have no trouble standing up and giving a talk to a room of 50 people now
11:17.50TandyUK2 years ago, id have been shitting myself
11:18.11UncleKiwii have done that once or twice - it gets better the more you do it
11:18.24TandyUKyeah, and you're doing it every week, so it becomes natural
11:18.27UncleKiwifirst tims is painful
11:18.29UncleKiwiahaha
11:18.42TandyUKwhich also helps when you go to other netowkring things too
11:19.17TandyUKbut anyway, back to my problem :P
11:19.23UncleKiwiok
11:19.29UncleKiwithanks i'll do bin
11:19.31UncleKiwibni
11:19.38TandyUKi have a customer with 6 phones behind a draytek router (same as 5 or so other customers)
11:19.51UncleKiwiok
11:19.58TandyUKthis one in particular is getting one way audio when transferring calls, or taking someone off hold
11:20.19TandyUKtheir router and phones are setup the same as my other working customers
11:20.30TandyUKwhat can i do to debug why this is happening
11:33.38SamotSIP debug
11:34.31SamotWhat the packets and see if you can see the audio change from sendrecv to sendonly or recvonly..can't remember exactly the names.
11:34.52SamotBut you'll see the SIP message to request the hold and another to request the hold be released.
11:35.00TandyUKyeah following that on a system with 400 extensions is kinda hard :P
11:35.12SamotNo it's not.
11:35.17SamotDebug the peer only
11:35.23SamotThe IP that phone is on.
11:35.28TandyUKtheres 8 phones on that ip
11:35.48SamotThen I guess you'll have to watch the packets and look through them.
11:35.55TandyUKother than sngrep, are there any good wireshark-like tools for analysing the sessions?
11:36.06Samottcpdump
11:36.15scvsngrep is the easiest tool by far
11:36.18SamotBut why aren't you doing a sip set debug peer <peer>
11:36.23TandyUKi am mate
11:36.26scvsamot that just limits by ip
11:36.26SamotAlong with a verbose
11:36.36SamotThat's what we want to do.
11:36.36TandyUKbut there multiple phones with these issues, and only sporadically
11:36.42scvTandyUK clearly said there are multiple peers on the same address
11:36.48SamotI understand that.
11:36.49scvso it's still mixed data
11:36.51scv?
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11:36.54SamotFFS, sometimes you have to read through data.
11:36.55TandyUKi was debugging ext 2001, and then 2004 had the problem
11:37.11SamotYou will see what happens with the audio and the call with the debugs.
11:37.54SamotBut if there is one way audio after the transfer or taking someone off hold..
11:37.57SamotThat's a NAT issue
11:39.16SamotSo it doesn't matter how many phones are on that IP, if they all have the same problem and they're remote to the PBX, NAT issue. You'll see it on any of those endpoints on that IP.
11:39.25SamotSo yeah, watching the IP will tell you what is going on.
11:39.28TandyUKit seems like its a nat issues the wrong
11:40.47TandyUKwrong way..
11:40.55TandyUKits the outgoing audio stream that fails
11:41.05SamotAre they remote to the PBX?
11:41.05TandyUKid expect the incoming audio stream to stop working
11:41.09TandyUKyea
11:41.09SamotThen look at the PBX.
11:41.13TandyUKpbx has no nat
11:42.09SamotWhat do the debugs show when you transfer a call or take it off hold?
11:42.23TandyUKive just been on the phone with this customer testing
11:42.38SamotAre you even looking in the PBX during the call?
11:42.39TandyUKand i can replicate it even after 5 transfers, and being on hold for 3 minutes
11:42.46TandyUKffs yes
11:42.56SamotSo what does the debug show?
11:42.57TandyUKbut its really hard keeping up with ~400 lines of text per second
11:43.26TandyUKhence my question if there is anythign sngrep/wireshark-esque for breaking the logs down into somethign manageable
11:46.52Samottcpdump -nqt -s 0 -A -i eth0 "host <ip> and (port 5060 or portrange 10000-20000)" -w testcall.pcap
11:47.46TandyUKyou also seem to be ignoring the fact that this is nor replicatable on demand
11:47.56TandyUKit doesnt happen every time
11:48.00SamotOK.
11:48.02SamotSo when it does.
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11:48.03TandyUKive just dont that, and have a perfect log
11:48.07SamotOK.
11:48.13SamotAnd what does it say?
11:48.19SamotWhere does the RTP die?
11:48.21TandyUKlol, i dont know when it does, only when the customer phones me afterwards
11:48.32TandyUKam i supposed to leave that pcap running 24/7?
11:48.32SamotYou just said you replicated it.
11:48.44TandyUKno i didnt, i said we could NOT replicate it
11:48.53TandyUKwe tried, sure
11:49.17SamotTandyUK> James
11:49.17Samotand i can replicate it even after 5 transfers, and being on hold for 3 minutes
11:49.25TandyUKoh sorry that was a typo
11:49.33TandyUKand i canT replicate it even after 5 transfers, and being on hold for 3 minutes
11:49.58TandyUKthis keyboard is waering out lol
11:50.11SamotYou have to be able to do the capture when it's happen.
11:50.20TandyUKright but im not psychic
11:50.20SamotThat is the down side of intermittent problems.
11:50.30TandyUKso i either leave that trace running 24/7 in the hope it happens again
11:50.44GuggeTandyUK: just leave tcpdump running and make it output a new file every 15 minutes, and then delete old files from cron :)
11:51.05SamotExcept how will he know if he's deleting the actual trace that shows the problem?
11:51.31Guggehes gonna home the customer tells him within a few days
11:51.31TandyUK^^ exactly lol
11:51.35Guggehope
11:51.49Guggeand then keep a few days worth or files
11:51.52SamotAnd just chew up diskspace with random traces.
11:52.02TandyUKoh the customer will tell me within about 20 mins of it happening
11:52.10TandyUKit happened once yesterday, and once so far today
11:52.12Guggethen you only need to keep the last 2-3 files :)
11:52.21SamotYou need to tell them to tell you immediately.
11:52.33SamotYou need to stress it's an issue that must be handled when it's happening..
11:52.39SamotNot 30 minutes after the fact.
11:52.53SamotTransfer call, gets one way audio, call tech support.
11:54.02SamotSo when you say outbound, you mean the caller can no longer hear the your customer?
11:56.16TandyUKCaller  <>  Hosted PBX  <DSL Link>  Customer Router < Wired LAN>  Extension
11:56.29TandyUKThe extension can hear the caller,
11:56.34TandyUKbut the caller cant hear them
11:56.41SamotFor both holds and transfers?
11:56.43TandyUKso if its nat, its on the outbound nat side
11:57.04TandyUKwell, im still trying to debug this myself, but ive been told it happens (occasionally) when
11:57.17TandyUKa)  putting the customer on hold, and a few minutes later they take the caller off hold
11:57.28TandyUKb) when transferring the call from Extension1 to Extension 2
11:58.09SamotIf the hold someone for a  minute or so it's fine?
11:58.58ruiedHello. I'm getting a call in at phoneA, Is there a way to inform phoneB of the callerID that is ringing to PhoneA. I want to be able to pickup the call  at phoneB (working) but knowing ho is calling to PhoneA.
11:59.02TandyUKwell i was on hold to them for 3 minutes, and it was fine
11:59.14TandyUKif it was nat timeout related, id expect that to be long enough
12:02.58ruiedsomekind of messaging the callerid to the screen of phoneB without ?
12:03.39ruiedwithout a connected channel
12:04.05SamotTandyUK: Both hold and transfer send INVITES which could be mapping the RTP on a different source port.
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14:34.08ruiedhow can I send text directly to the screen of a phone without needing to press something to read? I'm using messagesend() but I need to press a key to read it...
14:34.52ruiedIs there a way to send a message directly from the dialplan to the screen of a phone? from asterisk
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18:06.36freebsfor drmessano: https://www.youtube.com/watch?v=kfVsfOSbJY0
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18:20.19Samotshut up freebs.
18:20.36freebssettle down little fella
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20:33.25deweydbHi, i'm looking for a high reliability VOIP provider to port some of my numbers to.
20:34.33deweydbi'm using vitelity right now. but they haven't been the greatest in terms of uptime/reliability
20:35.39freebsflowroute...
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20:52.20MiccI'm having a problem with dynamic queue members in asterisk 13.8-cert1
21:26.31SpaceInvadersHow do you like to test scripts?  What are some convenient ways?  I want to play with strings and in bash it's easy to build small scripts and just run them immediately from the CL but  the only way I can think of in Asterisk is setting an extention to execute my test script and issuing a core reload now for every little change.
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21:30.51SpaceInvaderssorry, back!
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21:42.43Get_The_Fishhello all, I'm trying to set the variable MESSAGE(body) in LUA and I'm having a tough time with the syntax. would channel.MESSAGE.body = foo be correct?
21:43.03Get_The_FishI tried channel.MESSAGE(body) = foo and it didnt like that one
21:44.39SpaceInvadersDon't you need the set command?
21:45.38SpaceInvadershttp://www.voip-info.org/wiki/view/Asterisk+cmd+Set
21:47.59Get_The_Fishtried a couple of iterations of that and nothing has worked.
21:49.34Get_The_FishI dont think that this is a channel variable, so the docs arent really that helpful
21:50.21SpaceInvadersThat was what I was wondering
21:50.50SpaceInvadersI initially thought it was your own unique var.  But it's an asterisk var, isn't it?
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21:56.00SpaceInvadersOK it's not in the asterisk detailed variable list
21:57.53SpaceInvadersok given I don't see MESSAGE(body) listed in the detailed list of asterisk variables I expect it is handled like an array except you don't list a 2nd variable.  Is it generaiting an error?  That's what I'd expect.
21:59.09SpaceInvadersok no, I'm wrong.
21:59.23Get_The_Fishyeah its a variable
21:59.39Get_The_Fishsorry an asterisk var
21:59.43SpaceInvadersbut I don't see it listed
22:00.00SpaceInvaders*yet*
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22:01.15Get_The_Fishhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend
22:01.36Get_The_FishIt refers to a MESSAGE(to), MESSAGE(from), and MESSAGE(body)
22:02.02Get_The_Fishall of which are set when asterisk receives a SIP message with text
22:02.51SpaceInvadersOK my 1st thought--have you looked to see if it has to be set like an array? http://www.voip-info.org/wiki/view/Asterisk+func+array
22:03.11SpaceInvadersit states the format as almost exactly the same, right there.
22:04.01SpaceInvadersmy guess would be Set(MessageSend(to,from,body)=to\,from\,body)
22:04.39SpaceInvaderswell except  it doesn't list body.  it only says MessageSend(to,[from])
22:05.45Get_The_Fishthats a good thought I'm going to try it
22:05.56SpaceInvadersty :D
22:06.20Get_The_Fishwell actually in the docs it says "body of the message that will be sent is what is currently set to MESSAGE(body)"
22:08.14SpaceInvaderscould it be order?  Have you tried set(MESSAGE(body)=body) then setting the SendMessage per the format?
22:08.33SpaceInvaderser...
22:08.40SpaceInvaderscould it be order?  Have you tried set(MESSAGE(body)=body) then setting the MessageSend per the format?
22:09.56SpaceInvadersHEY!
22:10.05SpaceInvadersdid you see this? http://stackoverflow.com/questions/7755245/in-asterisk-10-how-can-i-use-messagesend-in-dialplan-to-send-messages-outside
22:10.55SpaceInvadersI'd start there and see if that works--assuming you haven't already :)
22:11.36Get_The_Fishyeah it doesnt like the nested parens
22:12.38MiccMy queue is not ringing any members, what could be the problem?
22:13.09MiccThis worked in previous version of asterisk, 13.7.2, but I upgraded to 13.8-cert1 and now it won't ring anyone even though they are available.
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22:15.14Get_The_FishSpaceInvaders: yeah, I did, and that works (well, mostly, but thats not really the point - unless you have experience sending SIP messages on SIPStation)
22:16.21Miccand now it crashed.
22:16.27SpaceInvadersI don't use SIPStation, sorry!
22:16.29Micc13.8-cert1 seems to be bad.
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22:23.27Get_The_Fisharray didnt work
22:23.34SpaceInvaders:/
22:23.39SpaceInvaderswhat'd you try?
22:24.39Get_The_Fishhttp://pastebin.com/1RyqP9rv
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22:25.14SpaceInvaders0.0
22:25.17Get_The_FishI think that the array index needs to be numeric
22:26.55SpaceInvadersok 1st question
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22:27.32SpaceInvadersMESSAGE[body] = "testing message body 0" <-- shouldn't that be set(MESSAGE(body)="testing message body 0")
22:27.48SpaceInvadersor at the least change the brackets to ()
22:28.16SpaceInvadersso MESSAGE(body) = "testing"
22:31.44Get_The_Fishchannel.MESSAGE("body"):set("testing message body 0")
22:31.56Get_The_Fishthats a set
22:32.05Get_The_FishmsgBody = channel.MESSAGE("body"):get()
22:32.16Get_The_Fishthats a retrieve
22:32.19Get_The_Fishreading is fundamental I suppose :)
22:32.24SpaceInvaders:)
22:32.41Get_The_Fishthanks though, sometimes just the act of explaining what you're stumped on helps you figure it out
22:33.12SpaceInvadersand I was reading the wiki like mad and learning because I want to tackle messaging at some point in the not-too-distant future
22:33.17SpaceInvadersso thank you, too :)
22:33.38SpaceInvadersI take it you were able to verify, with the retrieve, that you successfully set the var?
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