IRC log for #asterisk on 20161229

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01:02.29funxionanyone here
01:27.51SamotThere isn't "hunting"
01:28.10SamotThey are going to present a list of codecs in the order they want to use them, individually..
01:28.19SamotThey go down the list until there's a match.
01:29.57SamotSo it would depend on the codecs and their order you are sending.
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02:34.03*** join/#asterisk Maver|cK (~Mav-fp@43.246.224.90)
02:34.06Maver|cKCan i Make the Caller id of number which show in my 1st line of dialplan ?
02:34.20Maver|cKExecuting [1122222222@from-goip:1] NoOp("IAX2/1234-9783", "$EXTEN") in new stack
02:34.33Maver|cKFor examplle i want to make 11222222 to my caller id
02:34.37Maver|cKis it possible
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02:38.36Maver|cK?
02:38.46Maver|cKany one alive ? to answer my question
02:42.38Samot${CALLERID(num)}
02:52.56Maver|cKokay let me check
02:54.01Maver|cKSet(CALLERID(dnid)=${CALLERID(num)}
02:54.05Maver|cKlike this ?
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02:57.50WIMPyAre you still in to this thing and still confusing everyone including yourself with your phrasing?
02:58.14WIMPyOr is it really something new?
02:58.47Maver|cKWIMPy bro em Newbie to Asterisk Em Stuck here from last 3 days :(
02:59.31WIMPyAnd I told you two ways to fix your issue yesterday.
03:00.16Maver|cKcan u tell me
03:00.24Maver|cKExecuting [1122222222@from-goip:1] NoOp("IAX2/1234-9783",
03:00.31Maver|cKthis 1122222 i want ot make caller id
03:00.41WIMPyEither you save the extension in some other variable before it changes or you make sure it doesn't change by using Read() instead if WaitExten().
03:01.46WIMPyAnd then you use your variable or EXTEN. Forget about CALLERID(dnid).
03:04.38SamotWow..
03:04.40SamotNo..
03:04.47Maver|cKWIMPy in which variable i can save this extensions ?
03:04.59SamotSet(CALLERID(num)=${EXTEN})
03:04.59WIMPyYou can name it whatever you want.
03:05.15Maver|cKokay
03:05.20Maver|cKlet me try it
03:05.34SamotI think this like the 6th time someone has said "variable"
03:06.46WIMPyAnd it's not about caller ID at all, despite dnid is an option of the CALLERID function.
03:08.36WIMPyAnd using dnid was not a good idea anyway. 1. it's contents may differ depending on tech and 2. it might not contain what you expect.
03:08.46Maver|cKu Guys Are AWesome (Y)
03:08.53Maver|cKThanks ALot I Got it What i Need (Y)
03:09.03Maver|cKGOd BlesS ALL of U (Y)
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03:26.19lvlinuxSamot: nope, when I reloaded features.conf the console said that it was invalid and had to be 'caller', 'callee', or 'both'
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05:36.32igcewielingSamot: Constants aren't.  Variables don't.  -- unknown
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06:39.17denysoniqueDoes ifTime even work in AEL?
06:39.34denysoniqueDo timelimited includes work?
06:39.39denysoniqueI tried, and they don't seem to...
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09:34.57WildyLionHi guys. I need a bit of advice on queue logic
09:36.11WildyLionHow do I write a dialplan (AEL) rule for a case when the caller hung up waiting in a queue vs just going to 'h' extension unconditionally?
09:36.27WildyLioneven after the queued call has been completed successfully
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15:29.21funxionAnyone here?
15:31.39[TK]D-Fendernope
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15:58.58_boothi, i've built asterisk 13 on a debian box, i'm trying to set up pjsip and the only thing i've been unable to do so far is get a transport with protocol=tls recognised - i see no errors in asterisk's startup log but the transport isn't there (it works fine if I use udp). libssl-dev is installed, what could i check?
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16:29.46funxionwhats the most common reason a sip call dies after 8 seconds ?
16:30.27eric_hillDoes anyone have recommendations on an IP-enabled FAX machine?  Something that does iFax so I can integrate with Asterisk fax send/receive?
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16:47.50igcewielingfunxion: direct media enabled when one end is behind NAT
16:48.06igcewielingiFax?
16:48.12funxionyup
16:48.30igcewielingis that a protocol or a product?
16:49.13igcewielingfunxion: set directmedia=no and see if that helps.
16:50.13funxionthat wasnt the problem
16:50.48igcewielingI did answer your question though.
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16:51.43funxionI was uplate troublshooting a carrier issue and tried adding their subnet as a localnet  in sip.conf and forgot to remove
16:51.53funxionyou did answer my question though yes
16:51.55funxionthanks
16:52.20funxionIm going nutz trying to figure out this issue with one of my carriers
16:52.25igcewielingfunxion: misconfigured asterisk would also cause that symptom. 8-)
16:52.26funxiontrying everything
16:52.32funxioncorrect
16:53.08funxionMaybe you can help me with it
16:53.49funxionI havea DID provider where some DID's work just fine and others work every 15 calls
16:54.09funxionI have iptables rules allowing there entire subnet
16:54.39funxionthe calls that fail I see invites in ngrep but not in sngrep
16:55.01igcewielingdoes disabling iptables fix the issue?
16:55.08funxionnope
16:55.31igcewielingcarriers, such as Verizon SIP use many different devices outside their SIP signaling IPs.
16:55.43funxionneither does allowing anonymous sip
16:56.11funxionI know all of the media gateways for the provider fall into the subnets allowed in iptables
16:58.05igcewielingNice to have a provider like that.   Most only tell customers where the SIP proxies are, not the RTP devices.
17:13.56eric_hilligcewieling: iFax means the machine sends outbound faxes as email to 5551212@somedomain.com and checks an IMAP box for inbound faxes.
17:14.39eric_hilligcewieling: I set up dovecot and postfix with asterisk to handle it, but I'm looking for a smaller device.  The large copiers do it fine.  I need a desktop version.
17:18.27igcewielingeric_hill: that has nothing to do with Asterisk.
17:18.59igcewielingYou could build something with Asterisk if you really wanted to, I guess.
17:34.18funxionwhy would you be able to see an invite in ngrep but in sngrep?
17:34.56SamotI'm not sure that question is phrased properly.
17:37.10igcewielingfunxion: I can think of no reason other than you are using them wrong.
17:37.15funxionIm troublshooting an issue where certain DID's work from from a specific carrier and others do not consistantly.  The calls follow the same paths.  The only thing I have noticed is that on the DID's that fail I can only see the invites in ngrep but not in sngrep.  The calls that terminate properly the invites show up in sngrep at the same time as ngrep
17:37.46funxionigcewieling: I don't think I am using them wrong
17:38.00funxionI'm using sngrep filtering only for invites
17:38.09igcewielingah, I suppose if the packet is not valid SIP, sngrep would fail.
17:38.28funxionngrep -pqtN 'carrier subnet here' -W byline -d any port 5060
17:38.53igcewielingI normally use tcpdump.
17:38.59funxionigcewieling:  thats what I think but carrier is arguing
17:39.08funxionI see the same in tcpdump
17:39.54igcewielingdon't filter port 5060, filter on source IPs if the carriers.
17:40.00igcewielings/if/from
17:40.59SamotAre you using Chan_SIP or Chan_PJSIP?
17:41.22funxionchan_sip
17:41.47SamotIs "sip set debug on" not giving you the information you need?
17:42.06SamotOr even "sip set debug ip <peer ip>"?
17:42.10funxionon the calls that fail there is no output in sip debug
17:42.18funxionI only see the invites in ngrep
17:42.21funxionor tcpdump
17:42.50SamotAnd are they being sent to the right IP and port that Chan_SIP/ASterisk is listening on?
17:43.17igcewielinghow about posting ngrep if the failed call?  someone might be interesting slogging through the info.
17:43.22funxionyup
17:43.32funxionsure
17:43.36SamotIf Apache isn't listening on 8080, ngrep will show the traffic but the apache log will not.
17:43.39funxiongive me a few to gather the pieces
17:44.38igcewielingexcept for the most obvious errors, I don't troubleshoot packet captures, but there are others on this channel who might.
17:50.07funxionhttp://pastebin.com/LRv3G4Mp
17:50.21funxionAll there is is inbound invites without responses
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17:53.02SamotShow a "working" one.
17:56.39funxionI love the quotes around working BTW
17:57.45igcewielingheh "User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150416T155219Z~6e247a3f3d~64bit."
17:59.00igcewielingtry setting nat=yes.  U(17) 2016/12/29 17:45:40.899262 8.27.107.6:5060 -> 10.100.253.4:5060 looks like a NAT issue, depending on the interface
17:59.03SamotIs it "coming"?
17:59.15funxioninbound yes
18:00.09igcewielingwhat codecs do you have enabled?  try enabling only ulaw (or alaw if you are outside north america)
18:00.49SamotCodecs or NAT issues would still show up in a sip debug in the CLI
18:00.51funxionI only have ulaw enabled
18:00.59funxionI beleive it to be a NAT issue
18:02.35igcewielingmake sure you have nat=yes localne and externip set.
18:04.55SamotWhere is a working DID call?
18:06.41igcewielingalso set qualify=10000 and qualifyfreq=30 to see if that helps
18:08.24Kaianfunxion, are you using sngrep with port 5060 filters? Your packages are fragmented by IP, so try capturing without bpf filters
18:08.59igcewieling(12:39:54 PM) igcewieling: don't filter port 5060, filter on source IPs if the carriers.
18:10.50SamotWhy is all this being suggested?!
18:11.09SamotThere is an issue with certain DIDs from the same provider on the same peer not working.
18:11.27SamotThey show up in ngrep but not in an Asterisk console sip debug...
18:11.58igcewielingHmm?  NAT translations on routers sometimes clear the translation if no UDP traffic has been received.   Causes inbound calls to randomly fail.  That is why I suggested qualify and qualify freq.
18:12.02SamotSo they are hitting the server but are they hitting an ip and port Asterisk is listening on?
18:12.19SamotI want to see the difference between a failed DID and a working DID
18:12.35SamotTo see if they are being sent the same way with the same information.
18:12.56SamotCompare the two data sets before making a bunch of change suggestions.
18:13.21igcewielinghave fun, I'm going back to useful stuff.
18:13.29SamotI'm about to do the same.
18:13.53SamotI've waited 20 minutes for requested debug information and it's not coming.
18:15.27funxionsorry carrier called
18:16.14funxionand I think they changed something because I cant get a "working" call now
18:17.34funxiongot one
18:17.37funxioncoming in a sec
18:17.53funxionhttp://pastebin.com/aXTSHasF
18:21.25[TK]D-Fenderfunxion, Stop showing external dumps and prove whateach channel driver is listening on, etc
18:28.07SamotThat's the issue. The call doesn't show up in the Asterisk CLI with a sip debug.
18:29.32Samotfunxion: OK, now do this:
18:29.36Samotasterisk -rvvvvv
18:29.49Samotsip set debug ip <peer ip>
18:29.58SamotMake a call on a DID that doesn't "work"
18:30.20SamotCopy and paste EVERYTHING from the moment you enter the Asterisk CLI to hanging up the call.
18:30.21Samot~pb
18:30.22infobotfrom memory, 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.
18:30.29Samot^ Give us the link.
18:33.05funxionthere is a lot of output its a live pbx
18:33.25funxionand the call that is failing does not show up in sip debug in asterisk
18:33.37[TK]D-FenderGo prove firewalls and SIP settings
18:34.21funxionbindport=5060 is in sip.conf
18:36.13[TK]D-Fender"sip show settings"
18:36.19[TK]D-Fenderdump your firewall.
18:36.37[TK]D-FenderWe need actual proof &dumps shown
18:36.44[TK]D-Fenderanything less is a waste of everyone's time
18:37.02funxionk
18:37.19funxionhttp://pastebin.com/VZb6wAb8 sip settings
18:39.05funxionhttp://pastebin.com/xdvSjQCr  iptables -L output
18:42.22[TK]D-Fender10.100.253.4/255.255.255.255
18:42.33funxioninternal ip of asterisk box
18:42.39[TK]D-FenderWhy do you have a host entry as NETWORK one?
18:42.52[TK]D-FenderYou should not be making host entries like this
18:43.01[TK]D-Fenderit's not loke * should be talking to itself
18:43.11[TK]D-Fenderlike*
18:44.01funxiondoes that really matter
18:45.11funxionremoved
18:47.00[TK]D-FenderACCEPT     udp  --  anywhere             anywhere            state RELATED,ESTABLISHED
18:47.06[TK]D-FenderUDP = stateless
18:47.10[TK]D-Fenderso this looks funny.
18:48.17funxionwhat would you suggest
18:49.13[TK]D-Fenderwell first if you see see sip debug for other things, but not what that external dump showed, then it's pretty much hard proof that your firewall is responsible
18:49.41[TK]D-FenderSo validate taht by temporarly flushing it entirely so you should expect to see your call attempt
18:49.52[TK]D-FenderThen time to reinstate and review it all
18:50.43funxionI have flushed it and get the same results
18:51.03[TK]D-Fender* has not filtering of its own.
18:51.06[TK]D-FenderThis has to be external
18:53.35funxionNoramally I would agree with you on this but I have  had iptables  off with the same results
18:53.49funxionI can see invites in ngrep and tcpdump but not snmgrep or asterisk
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18:53.57[TK]D-FenderThis particular point isn't really up for debate.
18:53.59[TK]D-Fender* has no firewall
18:54.04[TK]D-FenderSo whatever you have is not part of *
18:54.20[TK]D-FenderSo either the defaults or rules or bad or something else is in the way
18:54.54[TK]D-FenderTher is no filter, nothing to "bug" no extra settings.  * shows the packets when you enable SIP debug.  Nothing shown = nothing made it to the daemon
18:55.10funxionagreed
18:55.25funxionmy thoughts is that the packets are malformed
18:55.40funxionthats why they do not show up in sngrep either
18:56.15funxionbut I can't see anything wrong with them
18:56.22funxionwas hoping someone else would
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20:06.11igcewielinggads, I really hate Javascript.
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21:13.26funxionso I think I've zeroed in on the cause of my problem just not sure how to resolve it
21:13.52funxionthe sip invite is getting fragmented and centos is timing out on reassembly
21:14.02funxionthus not answering the invite
21:14.09funxionor getting to asterisk
21:14.16funxionnot an iptables issue
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22:21.06WIMPyArgl
22:21.12tomcruise? :)
22:21.59WIMPyI just hit the mouse again with clutter.
22:22.03tomcruisehaha ok
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22:26.43funxionSamot: [TK]D-Fender
22:26.52funxionI figured out my issue and wanted to share
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22:30.06funxionThe reason the daemon didn't see the invites is because the invites were being fragmented
22:30.24funxionfreeswitch was adding too many information elements
22:30.42funxiononce the stuffing was removed fromt he freeswitch side everything is working
22:31.01WIMPyICE?
22:31.38funxion?
22:31.56WIMPyCaused by ICE?
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