IRC log for #asterisk on 20130924

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00:07.06Technodromeits crazy how much latency 2 cell phones have, not to mention any voip
00:17.23*** join/#asterisk jsjc (~Adium@225.Red-88-12-12.staticIP.rima-tde.net)
00:21.45jsjcHi!
00:22.02jsjcI have a little ssue… with asterisk registering to one of my trunks
00:22.21jsjcVia: SIP/2.0/UDP xx.xx.xx.225:5060;branch=z9hG4bK0a46a632;rport
00:22.44jsjcThe IP shown in the debug is correct for all my trunks beacause is the externhost IP or externip that I have set.
00:23.03jsjcbut the issue comes with certain trunk that connection must come from other interface
00:23.19jsjcit should appear no IP so eveything gets sorted on the way...
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00:55.27X-RobOh look
00:55.41navaismobye
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02:06.46CRCinAU_has anyone come across this before: receivefax_t38_init: error reading frame while generating CED tone
02:07.08CRCinAU_Someone just tried to send what I assume was a fax to me - but it bailed with this.
02:07.52CRCinAU_the only real hits I get from google are source code or unanswered forum posts :(
02:11.35CRCinAU_also, in 11.5.1, has the f option to fall back from T38 -> g711 been removed? o_O
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03:06.42*** join/#asterisk Quest (~syncsys@pool2-80-213.brain.net.pk)
03:06.43QuestI have cisco voip phones. they can call via internet. but when I plug the same wire (attached to cisco phones) to my pc. , I do get an ip assinged and can ping other ips on the same net mask. but I cant browse or ping ips outside the network. (internet). Why?
03:09.12PenguinAre you plugging into your own network or someone else's network?
03:10.28Questcompany network
03:11.19PenguinIf I had to guess, it's probably because of some network policies.
03:11.47Questya
03:11.54Questwhere would those policies be made?
03:11.59PenguinWith the PC, can you do DNS lookups?  nslookup, host, dig?
03:12.18QuestPenguin,  i tried tracerout only. no use
03:12.22PenguinYour network administrator would have created them and installed them in the switches/routers.
03:12.38WIMPyWhat makes you think there might be a connection to the internet?
03:13.04PenguinIf he can make calls "via internet," there must be a connection to the internet.
03:13.15QuestPenguin,  yes. i suspect that too.  how can i be sure
03:13.23QuestPenguin,  yes. thats my poing
03:13.27QuestPenguin> If he can make calls "via internet," there must be a connection to the internet.
03:13.36WIMPydoesn't believe the "via internet" part.
03:13.45PenguinI also have my doubts.
03:13.46Questif i  can make calls "via internet," there must be a connection to the internet.
03:14.00WIMPyVia IP !=  via internet.
03:14.26Questhow can i be sure whats stopping my access to internet. is it a vlan or the proxy server. or anything else?
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03:14.30PenguinInternet Protocol _is_ misleading.
03:14.31[TK]D-Fender[23:06]QuestI have cisco voip phones. they can call via internet. but when I plug the same wire (attached to cisco phones) to my pc. , I do get an ip assinged and can ping other ips on the same net mask. but I cant browse or ping ips outside the network. (internet). Why? <- fix your PC
03:16.54Quest<PROTECTED>
03:17.33[TK]D-FenderFirewall on PC
03:17.41[TK]D-FenderWindows network zone restrictions
03:17.44[TK]D-FenderSUN-SPOTS
03:17.46[TK]D-FenderCould be anything
03:18.02[TK]D-FenderHell lets just blame your browser proxy settings
03:18.12[TK]D-Fenderor the fact that browser != SIP
03:18.30[TK]D-FenderAnd you're comparing a SIP phone being allowed out to a web browser.
03:18.48[TK]D-FenderTraffic is totally different... and you've actually told us nothing about your operating environment
03:19.15WIMPyI still miss any proof that there is any calling "via internet" going on.
03:19.29[TK]D-FenderWhich is tantamount to asking "My car doesn't work, WHY?" and expecting a meaningful answer with that amount of information.
03:19.31Quest<PROTECTED>
03:19.44[TK]D-FenderQuest: What voip server?
03:20.32CRCinAU_Hmmm
03:20.46CRCinAU_I can't find anything on the fallback of ReceiveFAX in 10.5 o_O
03:20.55CRCinAU_eerrr 11.56
03:20.58CRCinAU_shit... 11.5
03:21.55CRCinAU_there's a c option, but no f.
03:27.56[TK]D-Fender[23:13]WIMPyVia IP != via internet. <- Oh noes!  His internet am fail!
03:28.22PenguinO nose
03:30.27CRCinAU_sooo - has the spandsp recievefax stuff been crippled? or am I missing something?
03:30.48CRCinAU_ie 1.8 had fallback from t.38 -> g711, but I can't find anything like that in 11.5
03:31.18CRCinAU_its certainly not mentioned in 'core show application ReceiveFAX'
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06:49.56BKhanHi
06:51.05BKhanI am configuring zaptal with asterisk 1.4. I have 4 ports of FXO card. I want to know on which port telephone line is terminate
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06:52.23kaldemarBKhan: on an FXO port of your choice. why are you using 1.4 and zaptel? they are really old.
06:52.56BKhanyes actually this is a device named ip04
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06:59.00BKhankaldemar: have u any idea about it
07:04.25kaldemarBKhan: about what?
07:05.10BKhankaldemar: I want to know on which port telephone line is terminate
07:05.32BKhanis it any option in Linux or asterisk
07:06.41kaldemarit is up to you to choose if you have more than one FXO port.
07:07.49kaldemarand you must have zaptel and asterisk (chan_zap is configured in zapata.conf) configured.
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07:27.39BKhanok
07:30.44BKhankaldemar: Let me explain. I am remotely configure zaptel . There are four ports and one of them there is a telephone cable. I just want to know on which port cable is attach
07:31.11BKhankaldemar: is there any way to check it
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07:32.54ChannelZconfigure each one and try a call
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08:28.31CRCinAU_so - in Asterisk 11, I'm getting this:
08:28.35CRCinAU_[Sep 24 18:23:54] WARNING[5353][C-0000001a]: chan_sip.c:16374 check_auth: username mismatch, have <spin>, digest has <72275>
08:28.56CRCinAU_now the sip.conf entry starts with [spin] - yet I have defaultuser=72275 in the config.
08:29.24CRCinAU_I also have fromuser=72275 as well
08:29.39CRCinAU_shouldn't that fix the problem? or have things changed in asterisk 11? o_O
08:30.12CRCinAU_it seems username= has been replaced with defaultuser=, but the error still shows.
08:30.37kaldemarCRCinAU_: defaultuser and fromuser are both parameters that affect outbound calls.
08:31.23CRCinAU_oh, I see. Is there a way apart from changing the [spin] section to have it affect inbound calls?
08:31.33kaldemarCRCinAU_: the call matches your [spin] device but has 72275 as authentication user name.
08:31.41CRCinAU_correct.
08:31.59CRCinAU_before, I had insecure=invite - but I'm trying to move away from using that.
08:32.33CRCinAU_as I have multiple sip peers, I'd like to keep it so I'm using Dial(SIP/blah@spin) to identify the VoIP provider easier
08:32.45CRCinAU_instead of using multiple definitions of ultimately the same thing
08:32.56kaldemarwhy is the other end sending 72275?
08:33.15CRCinAU_because that's the username that I register to
08:34.55CRCinAU_ie, its a peer, not a friend.
08:35.46CRCinAU_in most of my peer configs, none of the usernames actually match the phone number used
08:36.13CRCinAU_so ideally, I'd like to keep the peer name as it is - otherwise its difficult to track what Dial uses what provider
08:38.28Guggeno problem, you just have to use spin and not 72275 from the other end too
08:38.45CRCinAU_which I can't control?
08:38.53CRCinAU_ie it's a sip provider?
08:39.20Guggeif the other end have to send to 72275, you have to use [72275]
08:39.50CRCinAU_hmmm - that seems a bit dumb o_O
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08:45.03CRCinAU_I wonder then - is there a ways to alias these things in the dialplan instead?
08:45.32CRCinAU_therefore I could still somehow use an identifier in the form of a name/word instead of a username for a Dial?
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09:18.48*** join/#asterisk Milos|Netbook (~Milos@pdpc/supporter/student/milos)
09:19.02Milos|NetbookWhen using Dial() to a mobile phone, if I reject the call on the mobile, asterisk calls it again
09:19.10Milos|NetbookI have to reject it twice to continue the dialplan
09:19.32Milos|Netbookhttp://bpaste.net/show/tzImSY1FwbTiOOaqu4mC/
09:19.36Milos|Netbooknotice how it says ringing twice
09:19.51Milos|Netbookhow can I fix this, it can't possibly be a feature.
09:20.08GreenlightDidn't you ask this question last week
09:20.10Milos|Netbookno
09:20.18Milos|NetbookI only came in here first time ever yesterday
09:20.21GreenlightSomeone had a similar issue then
09:20.33Milos|Netbookwell, what was the reason and resolution?
09:20.36Milos|Netbookif you can remember
09:20.46GreenlightIn their situation it was their ITSP doing it
09:21.11GreenlightWe'd need to see your dialplan and a log of the call for yours to tell
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09:23.12Milos|NetbookGreenlight, http://bpaste.net/show/HYp8fch9iGeeKPmmerHv/
09:24.03GreenlightYup, this is your ITSP doing it.
09:24.09Milos|Netbookwhy would they call it twice
09:24.15Greenlightshurgs
09:24.16GreenlightAsk them
09:24.27Milos|Netbookhow ridiculous
09:24.33GreenlightWell
09:24.37GreenlightIn fact, get a SIP trace
09:24.50GreenlightSo we can see exactly what's being sent
09:25.08Milos|Netbooksip set debug on? is that a sip trace?
09:25.19GreenlightYea
09:25.27Milos|Netbookhow can I log that?
09:25.34Milos|Netbookit'll reach the end of my framebuffer
09:25.52Milos|Netbookor maybe not, let's see
09:25.54GreenlightAlso
09:26.02GreenlightWhy do you Answer the channel
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09:27.26Milos|Netbookyeah can't really log it
09:27.43Milos|Netbookbecause I'll transfer it to mobile if nobody answers it
09:27.50Milos|Netbookotherwise I have a prompt
09:28.06Milos|Netbookjust made the dialplan simple for this case
09:28.18GreenlightJust seems odd to answer before the other side has
09:28.49GreenlightWithout a SIP trace it's hard to tell exactly what's going on, but Asterisk shound't "retry"
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09:32.05Milos|Netbookwell logging.conf isn't loggin any sip packets despite being set to debug
09:32.35Milos|Netbookoh
09:33.05kaldemarMilos|Netbook: debug in logger.conf is "core debug", not "sip debug".
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09:33.26Milos|Netbookso how do I log sip debu
09:33.26Milos|Netbookg
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09:34.36kaldemarMilos|Netbook: asterisk has no built-in way to log it to a file. script or "asterisk -vvvr | tee /tmp/ast_log.log" will do the job for example.
09:35.15Milos|Netbookit won't
09:35.22Milos|Netbookbecause I need to enable sip debug first
09:35.29Milos|Netbookwhich is done on the CLI
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09:35.50kaldemarwhat stops you from enabling sip debug in the CLI?
09:36.01Milos|Netbookcan I open several CLIs and they are linked?
09:36.04Milos|Netbookoh
09:36.05Milos|Netbookright.
09:36.14Milos|Netbookhaven't really used tee
09:36.18Milos|Netbookit duplicates output
09:36.20Milos|Netbookgotcha
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09:39.32Milos|Netbookhttps://dpaste.de/zWFkW/
09:39.47Milos|NetbookGreenlight, above is the trace
09:42.28GreenlightHow are you calling in?
09:42.58Milos|Netbookwith a landline
09:43.23GreenlightIn over the same trunk yea?
09:44.01Milos|Netbooklandline is completely unrelated to asterisk
09:44.10Milos|Netbookif that's what you mean?
09:48.44GreenlightYup this is your ITSP doing it
09:49.32Milos|Netbookwell why would they be
09:49.36Milos|Netbookyou said someone else had this issue
09:49.43Milos|Netbookthat means it's not really specific to my own itsp
09:49.47Milos|Netbookso why would they do this o_O
09:50.52GreenlightIt's odd. As to why they would do it, ask them
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09:51.13GreenlightIt's a strange trace
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09:51.45GreenlightThey go from PROGRES -> RINGING and *then* ask for 407 auth
09:52.27GreenlightI wonder
09:52.28GreenlightOk
09:52.46GreenlightHere's what I think is happening
09:52.52GreenlightYou send an INVITE
09:53.04GreenlightThey accept that INVITE and start to progress the call,
09:53.12GreenlightThey send a 183 SESSION PROGRESS
09:53.20GreenlightYour mobile starts to ring
09:53.26GreenlightThey send a 180 RINGING
09:53.35GreenlightYou press red button on your mobile to reject the call
09:53.55GreenlightThey now send a 407 Proxy Auth required
09:54.08GreenlightWhich is where things go amiss
09:54.44GreenlightSee line 645
09:54.56GreenlightThe question is why are they rejecting the call incorrectly
09:57.03*** part/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au)
09:57.09GreenlightSInce, asterisk replies with the auth response in the form of another INVITE
09:59.10GreenlightI can't actually see your phone start ringing again after that though
10:00.43Milos|Netbookvery strange
10:00.52Milos|Netbookthanks for going through it for me
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10:40.45lkthomashey guys, which softphone works well with KDE on linux desktop ?
10:47.08sekilkde doesn't make a difference..
10:47.59sekilI had good experience with twinkle..
10:50.52lkthomastwinkle doesn't do pop up
10:50.54lkthomasI just tested it
10:51.52bacobarthave you tried blink? ( http://icanblink.com/ )
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11:02.18lkthomasI will try it, thanks
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11:39.18ocholetrashi guys
11:39.21ocholetras^^
11:40.07ocholetrasTalking about contexts.. is there any kind of "good practices" or "patterns" guide about structure of your contexts?
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12:26.57lxsameerhi guys, does an agi script run blocking ?
12:28.06[TK]D-Fenderyes
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12:29.10lxsameerhow can i make one of them non-blocking ?
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12:29.33lxsameercan i fork any process to make them non-blocking
12:29.43[TK]D-Fenderno
12:30.02[TK]D-Fenderlxsameer: What precisely are you looking to accomplish?
12:30.36lxsameer[TK]D-Fender: i want to put some data in a database on any calles
12:30.50[TK]D-Fenderlxsameer: try to be more specific.
12:30.58Greenlightlxsameer: Why do you need to fork in order to do that?
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12:31.22lxsameer[TK]D-Fender: I want to put some data about an incoming call to a database belongs to one of my softwares
12:31.39[TK]D-Fenderlxsameer: "when" and "how" are very important.....
12:31.45lxsameerGreenlight: forget about it , it was an stupid question
12:31.50[TK]D-Fenderlxsameer: You'll need to be more specific
12:32.10GreenlightOk, at a guess the database query is doing a lookup on callerid, that can take some seconds to run?
12:32.20lxsameer[TK]D-Fender: currently they put an php file in agi-bin folder which runs in case of an incoming call
12:32.54[TK]D-Fenderlxsameer: Who is "they", and what's in the PHP file?  It's not even your own code?
12:33.10lxsameerGreenlight: exactly in addition to that inserting data take time too and this cause caller to hear nothing for couple of minutes
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12:33.20GreenlightMInutes!?
12:33.26GreenlightFix your db tbh
12:33.40lxsameer[TK]D-Fender: unfortunately it's not mine but i have to fix it (damn my boss)
12:33.53lxsameer[TK]D-Fender: let me paste the php code
12:34.10[TK]D-Fenderlxsameer: rather than showing us raw code, you need ot be clear abo9ut what you want to accomplish
12:34.21[TK]D-Fenderlxsameer: Because for allwe know that code is also a terrible approach
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12:34.58bombevhi
12:35.21bombevI have this range of extensions 3XX
12:35.42lxsameer[TK]D-Fender: I'm trying to explain my self. I have a software (CRM) called ABC for example. I want to put the caller info in ABCs database when anyone calls our company. so
12:36.07bombevand I want to restrict call to all extensions 3XX except 360, 330,341
12:36.10lxsameer[TK]D-Fender: I read the astrisk database and insert the extracted data to ABC's database
12:36.12bombevwhat is the best way to do it
12:36.23bombevif statement or match pattern
12:36.30Greenlightlxsameer: The fact that it takes MINUTES is an indicator that something is wrong.
12:36.50GreenlightWe do a lookup against 50 million tel numbers on incomming calls and insert call records and it takes <1s
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12:37.26bombevany idea gyus?
12:37.29bombevguys?
12:37.34lxsameerGreenlight: i think it take time to insert data no looking up for callerID
12:37.38[TK]D-Fenderlxsameer: What "Asterisk Database"?
12:37.54[TK]D-Fenderlxsameer: What "caller info"?
12:38.04Greenlightlxsameer: Inserting records taking minutes also means your db is broken
12:38.05lxsameerGreenlight: i know that current approach is a disaster but I have to fix it quick
12:38.27lxsameerGreenlight: also second db is on another host so its normal
12:38.35GreenlightNormal it is not
12:38.38fileraises eyebrow
12:38.43GreenlightMINUTES to insert a record
12:38.52GreenlightWhoever told you that's normal was lieing
12:38.53lxsameer[TK]D-Fender: channel number start time and end time and status
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12:39.17bacobartare you sure your insert is even working? when the script only does an insert and it takes minutes it's much more likely the connect to the database is timing out
12:39.29Greenlight+1
12:39.29lxsameerGreenlight: not that much my friend It's my fault to show them bigger to you
12:39.44[TK]D-Fenderlxsameer: You don't need an AGI for this at all as far as I can see.
12:39.57lxsameer[TK]D-Fender: good, what can i use then ?
12:39.58[TK]D-Fenderlxsameer: and nothing "background" at all.
12:40.04Greenlight[TK]D-Fender: Nonetheless if his db takes minutes he cant block
12:40.18Greenlightlxsameer: What, that doesn't make sense ?
12:40.24[TK]D-Fenderlxsameer: Call comes in.  Put your "start" info into your DB.  Call ends, put your "end" info into the DB
12:40.34lxsameerGreenlight: which one ? :P
12:40.41Greenlight[01:39pm] <lxsameer> Greenlight: not that much my friend It's my fault to show them bigger to you
12:40.47[TK]D-Fender[08:40]Greenlight[TK]D-Fender: Nonetheless if his db takes minutes he cant block <- this has nothing to do with DB delay
12:40.51bacobartif that's all you want you should just enable cdr/cdl
12:41.07Greenlight[TK]D-Fender: He said his inserts take minutes.
12:41.15lxsameer[TK]D-Fender: unfortunately they one these data real-time, I know that this is a very bad idea but i have to fix it for know ,
12:41.28lxsameerGreenlight: to be honest it take about 30seconds
12:41.48Greenlightlxsameer: Anything more than 0.5 seconds to insert a row and I'd have warning bells going off
12:41.50lxsameerI just want to make a non-blocking code running on each call
12:42.18[TK]D-Fenderlxsameer: You've described notihng about anything happenening between the start of the cal and the end.. so what is there to do in "real time"?
12:42.23lxsameerGreenlight: unfortunately there softwares which handed to me are way worth that you may imagine
12:42.28bacobarti'm pretty sure the script isn't doing what he thinks/expects it's doing
12:42.29Greenlightlxsameer: Ok, so you've a few possible solutions. All are hacks, as your underlying problem is that your db is awful.
12:42.34[TK]D-Fenderlxsameer: This isn't picking up LOGS 10 minutes later.  This is execute on START, and on END.
12:43.09lxsameer[TK]D-Fender: they want to see a popup when someone calling with a link to their profile in that ABC (crm) software
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12:43.31Greenlightlxsameer: So, that's not going to be possible without that delay.
12:43.38[TK]D-Fenderlxsameer: Fine.  Execute the START  OF CALL message script at the START... and then ANOTHER one at the END of the call
12:43.41GreenlightUnless you fix the script or db
12:44.13lxsameer[TK]D-Fender: can you give me a link about these scripts ?
12:44.19bacobartlxsameer: can you put the script on pastebin?
12:44.25Greenlight+1
12:44.33GreenlightLets see if its the sript or the db
12:44.35[TK]D-Fenderlxsameer: No, the code that willactuall tell your CRM this is YOUR job.
12:44.45lxsameerbacobart: sure
12:45.03[TK]D-Fenderlxsameer: Just execute on start, and then on end.
12:45.50Greenlightlxsameer: Quick fix, ditch AGI, use System and execute it backgroud (suffix "&")
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12:47.15GreenlightAnd immediately your users complain that it's 30 seconds into the call before they get their popup
12:47.16GreenlightI know for sure my users complain if their screen pops take more than 1 second.
12:47.23[TK]D-Fender[08:35]bombevI have this range of extensions 3XX   [08:36]bombevand I want to restrict call to all extensions 3XX except 360, 330,341 <- vague description.  What is this "restrict" you're referring to?  What are the actual valid numbers and what at the invalid numbers?
12:47.38lxsameerGreenlight: where can i tell astrisk to run that script ?
12:47.53[TK]D-Fenderlxsameer: Wherever you want
12:47.59GreenlightVarious places. Probably the System command is easiest in your case
12:48.00[TK]D-Fenderlxsameer: It's your dialplan
12:48.09Greenlightcore show application system
12:48.44lxsameer[TK]D-Fender: I meant how can i tell astrisk to run a bashscript on an incoming call
12:48.57Greenlightexten => 1,1,System(ScriptThatTakesAges.php &)
12:49.00[TK]D-Fenderlxsameer: in the DIALPLAN.  You know.. the thing that actually processes your call.
12:49.14[TK]D-Fender~book
12:49.14infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
12:49.16[TK]D-Fender^^^
12:49.19bombev[TK]D-Fender well, I want to allow calls only to 360,341,330 and all other extension in this range 3XX I want to be restricted so the caller cant reach them
12:49.23lxsameer[TK]D-Fender: ok letme find some doc about that
12:49.45[TK]D-Fenderbombev: then make a context with those 4 extens in it
12:50.28lxsameerthanks guys
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13:00.30dongola7anyone have problems with CDRs and ChannelRedirect? I'm seeing an issue where the CDR records disappear when ChannelRedirect is executed in the dialplan.
13:02.09dongola7and by 'disappear', i mean they aren't recorded in the database.
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13:09.57Kattygood morning
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13:11.58CubberI have an IAX2 trunk setup between two asterisk servers running asterisk 11.5.1 on gentoo.  CLI only.  The trunk works, however nothing appears when I do iax2 show stats or netstats on either server.  Both servers report each peer is not monitored when I do iax2 show peers.  How can I enable monitoring and statsitics?
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13:13.05[TK]D-FenderCubber: And how many calls are active?
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13:16.01GreenlightIs it like SIP where you need a qualify=yes ?
13:17.13[TK]D-FenderFor "MONITORED", yes
13:17.34[TK]D-FenderVery much Asterisk 101...
13:19.40Cubbercurrently none
13:20.01Cubber[TK]D-Fender: by that question I take it I need at least one call to see statistics?
13:20.53[TK]D-FenderCubber: * doesn't collect data for some random amount of time for it
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13:23.13Cubber[TK]D-Fender: that make sense, thanks
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13:35.00ACiDVGot a server that look to be victim of a SIP attack. I'm using Asterisk 11 and on the security log files, I got tons of ChallengeSent logs (ex. 1000 attempts in a single second), causing server to crash (Asterisk using 100% of RAM).  What is a "ChallengeSent" request ? Unfortunately I cannot automatically block attempts using fail2ban.
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13:36.59GreenlightYour SIP connections are open to all IPs ?
13:37.31[TK]D-Fender"Unfortunately I cannot automatically block attempts using fail2ban." <- because .... ?
13:37.39Greenlight+1
13:38.05ACiDV[TK]D-Fender: because ChallengeSent isnt an error no ? It only a warning message ?
13:38.31[TK]D-FenderACiDV: The call is getting refused, isn't it?
13:38.45[TK]D-Fenderacixbecause that would cause a real "fail" message to get logged
13:38.46ACiDV[TK]D-Fender: We can block for others logs like AuthenticationFailed, etc but ChallengeSent can be ok for valid registration no ?
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13:39.16[TK]D-FenderAfter the Challenge comes the Fail
13:39.24ACiDVBut no fail on log :(
13:39.48[TK]D-Fendershow us.
13:39.49ACiDVonly ChallengeSent (1gb of logs per day) and no others fail ... server crash before
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13:40.22[TK]D-FenderACiDV: Then limit it on the firewall itself.
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14:22.09Kattywaves to _Corey_
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14:33.09_Corey_wakes up and waves back at Katty ...
14:33.15_Corey_:)
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14:37.08KattyNugget: telnet
14:37.13Nuggettelnet is eeeeeeevil!
14:37.15Katty:>
14:37.24Kattyhugs Nugget
14:37.34Nuggetyay
14:37.58Kattyhow's si nugget?
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15:59.52DpunktCan someone give me some hints how i can get my application to interact with asterisk?
16:00.11pabelanger~book
16:00.11infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
16:00.16[TK]D-FenderYup...
16:00.23[TK]D-FenderBook is a big hint...
16:00.27[TK]D-FenderIndexed even
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16:00.31pabelangerDpunkt, have you read the book?
16:02.04Dpunktno, an scanning the index doesnt point me to the right chapter
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16:02.27DpunktAsterisk->Application works fine via ivr in dialplan, but how do i get my application do do some calls
16:03.01Alex_hanyone got really into the guts of clustering with asterisk? i have a deep ass question for you...
16:03.03pabelangerDpunkt, you need to give us more information, what is this application?
16:03.20Dpunkthome automatisation
16:03.24pabelanger~ask
16:03.25infobotQuestions 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.
16:03.28pabelangerAlex_h, ^
16:03.46Alex_hsorry! ok ill rephrase
16:03.54Dpunktyou should be able to controll thinks via phone- works
16:03.59pabelangerDpunkt, start at the beginning and explain it to us like we don't understand what you are trying to do
16:04.13pabelangerbecause we don't
16:04.28DpunktOK
16:04.59Alex_him interested in finding out if asterisk nodes can maintain a call between fail overs. SIP specifically, so if a call is setup from one node, then the system fails over will the call be maintained?
16:05.26pabelangerAlex_h, no, the call drops
16:05.56pabelangerAsterisk SCF was to be able to handle this, but that project died
16:06.05Alex_hso there is no mechanism within asterisk for maintaining that call? i guess it would require a proxy of some kind which asterisk is not...
16:06.24Dpunkti would like to be able to control some ports of a old isdn-pbx. to do this by phone i would just dial some numbers and im done. but now i would like to do it from the home automation software. so i need to say "asterisk call xyz and hang up after connect"
16:06.39ChainsawDpunkt: Ah, the Quattrovox lives...
16:06.41pabelangerAlex_h, well, no, a proxy will not help either.  The only way to handle it would be to reinvite the audio before the network drops
16:07.06Alex_hpabelanger: thanks!
16:07.10Dpunktthe what?
16:07.52pabelangerDpunkt, CLI*> core show application originate
16:07.54Alex_hDpunkt: You can originate a call via AMI or a callfile, both of which you can involve in a script, if you mean you want to have asterisk make a call itself
16:08.06Alex_hor on the CLI as pabel just said
16:10.24Dpunktthanks now i know where to look in this book, thank you
16:12.33Alex_hpabelanger: thanks again for pointing us in the right direction i see now why the reinvite would be the only way for this to work
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16:56.29jpf31hi
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17:00.54*** join/#asterisk g_r_eek (~g_r_eek@ppp-94-68-160-230.home.otenet.gr)
17:01.49*** join/#asterisk nam3l3zz (~quassel@86-46-207-104-dynamic.b-ras1.pgs.portlaoise.eircom.net)
17:04.11hypknighthey guys, dumb question: I'm accepting a new circuit from a provider that installed a private proxy, but wants me to register to a public host. I have the following set in the sip.conf -- does this look right? All inbound and outbound calls are supposed to come through this: http://pastebin.com/Zs9FEe3C
17:05.44[TK]D-Fenderfromdomain=private asterisk IP <- no, fromdomain is never "you"
17:06.06hypknightsorry
17:06.10hypknightthat's supposed to be the provider
17:06.16[TK]D-Fenderhypknight: As for the rest.... go try it
17:06.41[TK]D-Fenderhypknight: canreinvite=yes <- BAD
17:06.48hypknightreally?
17:06.54hypknightI believe you... but just for my knowledge, why?
17:06.58[TK]D-Fenderand for 1.6+ you should have "directmedia=no"
17:07.16[TK]D-FenderYou could end up allowing a reinvite across a NAT and lose your calls
17:07.22hypknightgotcha
17:07.37[TK]D-Fenderport=5060 <- almost never needed explicitly
17:08.33hypknightfair enough, I put it in there because I put this config in play and I'm having issues with it
17:08.37Kattyrandomly jigs in one door and out the other.
17:08.47hypknightthe provider is just pointing fingers at me, so I figured I'd check
17:09.41[TK]D-Fenderhypknight: Why ask if it's right if you are alrady having actual issues with it?
17:09.55[TK]D-Fenderhypknight: It's a pointless question.  You should be showing us the failure.
17:10.02hypknighti'm not seeing any
17:10.09hypknightlike I said, the provider says it's me
17:10.13[TK]D-Fenderhypknight: Where's the debug?
17:10.16hypknightit was a sanity check, that's all
17:10.29hypknightforget i asked [TK]D-Fender, I knew better
17:10.54[TK]D-Fenderhypknight: If they say your settings are wrong then that means they are hitting your server.  Enable SIP debug and look at what's happening.
17:12.07hypknightthe reason I asked is because I'm registering one place and proxying through another... I haven't done it before. To answer your question.
17:13.47[TK]D-Fenderhypknight: We can't tell if a bunh of other settings are right or not on-sight.  It depends on what the vendor is expecting.  Errors would also give us an indication.
17:16.15hypknightSo, is outboundproxy= still valid? and will it attempt to send media through that address?
17:16.41[TK]D-Fenderhypknight: Media goes where the SDP tells it to.
17:16.51[TK]D-Fenderhypknight: outboundproxy is for SIP
17:17.02hypknightperfect
17:17.15hypknightit'll send though the proxy and SDP will do the rest
17:17.18hypknightthanks much
17:33.19*** join/#asterisk kemmler (tully@r74-192-178-52.htvlcmta01.hnvitx.tl.dh.suddenlink.net)
17:42.27hypknightBracing for impact, but, in the sip.conf - the register=NXXNXXXXXX:password@public.provider.fqdn/NXXNXXXXXX      just confirming, that goes in the [general] context right?
17:43.30[TK]D-Fenderhypknight: After the end of it and before the first other section
17:48.08cuscohi folks
17:49.12Kattywaves to cusco
17:50.49*** join/#asterisk jpoz_ (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com)
17:52.27hypknight[TK]D-Fender: perfect, that's where I thought. When I try and call inbound to a test DID from that provider with the proxy I'm getting: "Sending fake auth rejection for device 'CNAM<sip:NXXNXXXXXX@public.provider.ip:5060'" -- I have sip set debug peer enabled on the context but I don't see anything
17:52.54cuscowaves back at Katty
17:54.10cusco... any bad stuff from placing many calls (say.. 20) in outgoing dir, or using originate? being that these calls stay in queue() until answered ?
17:55.03cusco.oO(can you enable sip debug for a specified context?)
17:55.25cuscohypknight: I normally use sip set debug ip <ip>
17:57.27cuscosay... this messages keep on showing: pbx_spool.c: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/66402.927841053.call
17:58.03cuscoand I'm afraid because there are many concurrent calls in the spool dir and they're queueing ...
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18:02.00[TK]D-Fender[13:52]hypknight[TK]D-Fender: perfect, that's where I thought. When I try and call inbound to a test DID from that provider with the proxy I'm getting: "Sending fake auth rejection for device 'CNAM<sip:NXXNXXXXXX@public.provider.ip:5060'" -- I have sip set debug peer enabled on the context but I don't see anything <- you obviously see something, because sending a fake auth IS something
18:02.04*** join/#asterisk Changos (~Changos@unaffiliated/changos)
18:02.40hypknightI'm sending a 401
18:02.57hypknightand they aren't authenticating
18:03.09[TK]D-Fenderok
18:03.27*** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net)
18:04.11jmetroCan anyone provide a recent-ish example of having a variable say TestExten, calling an agi script, and having ${TextExten} be something the agi script set? I'm using the set_variable command but it doesnt seem to like me.
18:04.41[TK]D-Fenderjmetro: Show us
18:05.08jmetro[TK]D-Fender: I have to go eat lunch first, i was hoping someone had an example while i went and got tacos.
18:05.47[TK]D-Fenderjmetro: Just show us your mistake so we can fix it instead of finding a million guides you still won't follow :p
18:06.10_Corey_jmetro: I want tacos.  My proposition is this: bring me a taco and I will send you examples.
18:11.24cuscodraws a taco in ascii and expects examples
18:14.36hypknight[TK]D-Fender: is there a specific place I need to tell asterisk to accept anonymous calls from the proxy?
18:15.06[TK]D-Fenderhypknight: in a peer entry
18:15.29hypknight[TK]D-Fender: and the one I showed you won't do it?
18:16.06_Corey_jmetro: So, you want to call an AGI like this...  AGI(${variablecontainingpathtoagi})... and then have said AGI manipulate that variable?
18:17.50[TK]D-Fenderhypknight: I don't know.  I haven't seen anything
18:18.00[TK]D-Fenderhypknight: Clearly it's not quite making it yet
18:18.41*** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net)
18:18.42*** join/#asterisk zigg (~matt@unaffiliated/zigg)
18:19.58*** join/#asterisk italorossi (~italoross@187.60.66.11)
18:22.35[TK]D-Fender_Corey_: Not going to work....
18:22.55[TK]D-Fenderactually... nvm
18:22.57[TK]D-Fenderscratch that.
18:23.14_Corey_I'm not sure I understand what he's trying to do...
18:23.19[TK]D-Fenderjust sits and waits to see code and/or debug
18:23.50[TK]D-Fender_Corey_: He's setting vars with the wrong syntax. Lets just see what the actual code looks like
18:26.26_Corey_awaits his taco
18:39.25*** join/#asterisk timahvo1 (~rogue@197.237.174.93)
18:40.23Kattyinfobot: blacklist
18:40.23infobotIn etch, lenny and sid, create/edit /etc/modprobe.d/blacklist.local and add a line similar to this (without quotes): \"blacklist module_name\".  IMPORTANT: ask about <blacklist-initramfs>.  For sarge ask me about <blacklist sarge>.  To blacklist a module at installation time, ask me about <installer blacklist>.
18:42.36[TK]D-Fenderinfobot: plaidlist
18:42.43[TK]D-Fender:(
18:54.09Kattyso i can make asterisk do stuff outside using System()
18:54.18Kattyis there a way to put stuff back in?
18:54.33Kattysomething simple. like. core set verbose 10
18:55.03Kattywithout re-attaching to the CLI
18:58.29[TK]D-FenderKatty: There is no CLI app/function.  If youw ant to use the dialplan to enact it you'll either call asterisk from CLI, or script it through AMI
18:58.41*** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit)
18:58.42Kattywell isn't that a bummer.
18:59.09*** join/#asterisk timahvo1 (~rogue@197.237.174.93)
18:59.23mjordanKatty: be evil. System(asterisk -rx "core set verbose 10")
18:59.57pensmitI've configured two exensions perfectly.  I want to be able to configure a ring group with these two extensions.
19:00.00pensmitHow do I do that?
19:00.16Kattymjordan: now THAT i can work with
19:00.32Kattymjordan: i'm trying to write up a simple IVR where our clients can add their own numbers to the blacklist
19:00.40navaismopensmit: define ring group
19:00.42Kattymjordan: take a variable here, dump it there
19:00.53Kattymjordan: i'll give that a go
19:01.07navaismopensmit: ring at same time, ring 1 after the other?
19:01.13pensmitsame time
19:01.25pensmiti have an ivr i created
19:01.27navaismodial(SIP/dev&sip/dev2)
19:01.51pensmityes but will that go straight to those devices
19:01.52pensmitor
19:01.57pensmitto the extensions i created
19:02.14navaismouh?
19:02.37pensmitsee that will skip the stuff that's in the extension
19:02.57pensmitso i have an ivr
19:03.02pensmitthat when they press 1
19:03.12pensmiti want it to hit two defined exensions
19:03.20pensmitso maybe ring group isn't the thing to do?
19:03.33navaismowait---extensions != devices
19:03.39pensmityes
19:03.43pensmitexactly
19:04.06navaismobased on your example if they press one---> 1,1,Dial(SIP/dev1&SIP/dev1)
19:04.11[TK]D-Fender[15:02]pensmitsee that will skip the stuff that's in the extension <- there is no "stuff".  You create "stuff"
19:04.26pensmitlol
19:04.27pensmitguys
19:04.31pensmiti've created stuff
19:04.33pensmitlmao
19:04.40[TK]D-FenderSo it's your dialplan.. you already know what's there...
19:04.43pensmitso i've defined two extensions
19:04.45[TK]D-Fendergo change your own dialplan
19:04.51pensmitit'snot working
19:04.56pensmitthe extensions work
19:05.02[TK]D-Fenderpensmit: And we can't see why.  The circle is complete
19:05.16pensmitbut jumping from the ivr to the extension is no
19:05.17pensmitnot
19:05.28dongola7could you just use local channels for this? Dial(LOCAL/<ext1>@<context>&LOCAL/<ext2>@<context>? if there is other 'stuff' associated with dialing a particular extension beyond dialing the device?
19:05.44[TK]D-Fenderpensmit: You keep throwing these terms "extensions" and "IVRS" around and we have no idea about your layout.
19:06.15[TK]D-Fenderpensmit: Your hierarchy is entirely your own.
19:06.47pensmitexten => 203,1,Dial(SIP/203,20)
19:06.47pensmit<PROTECTED>
19:06.47pensmitexten => 203-BUSY,1,Goto(203menu,busy,1)
19:06.47pensmitexten => 203-NOANSWER,1,Goto(203menu,unavail,1)
19:06.47pensmitexten => _203-.,1,Goto(203menu,unavail,1)
19:06.54*** join/#asterisk vlad_starkov (~vlad_star@109.188.126.119)
19:06.55[TK]D-FenderPenPASTEBIN
19:07.09[TK]D-Fenderpensmit: PASTEBIN
19:07.14pensmitso for example that'sone
19:07.25pensmitthe other is 205
19:07.30pensmitsame type of stuff
19:07.31[TK]D-FenderPenthat is also jsut some code from some context.  We don't kinow what is going where, what else points to what... or what the heck you're trying to do.
19:07.52navaismouse the dongola7 suggestion, dial local channels
19:08.56pensmitlets say i have two of those
19:08.57pensmitjust like that
19:09.03pensmitnow
19:09.33pensmitin the employees context
19:10.16pensmithow can i create a "ring group" that would go through one of those depending on who answered first
19:10.33pensmitor do I have to define that independently
19:11.02[TK]D-Fendermake an extension that dials BOTH as you've alreadqy been shown
19:11.17pensmitok but wouldnt' that dial the device
19:11.27pensmitand skip the extension stuff
19:11.34pensmitso no 20 sec timeout
19:11.51pensmitno busy or unavailble check
19:12.57pensmitcan you do a GoTo for both maybe?
19:13.16navaismoARGH
19:13.36[TK]D-FenderPenyou make ANOTHER extension
19:13.36navaismoUSE LOCAL CHANNELS as dongola7 said
19:13.47[TK]D-Fenderno need for local channel
19:14.17[TK]D-Fender[15:11]pensmitand skip the extension stuff <- this has no meaning
19:14.24pensmitjesus christ man
19:14.28[TK]D-Fenderpensmit: Make another pattern
19:14.30pensmithelp a brother out
19:14.31pensmitlol
19:14.44pensmitextension stuff
19:14.47pensmiti just put it in ther
19:14.49pensmityou see it
19:14.51pensmitsemantics
19:14.59[TK]D-FenderYou are lacking proper comprehension of dialplan basics and ability to clearly express yoruself
19:15.00navaismohe insist to use those extens they can use with localchannels period, seem like he dont want to create another exten or anything in his dialplan
19:15.14pensmityou are lacking ability to be a human being
19:15.39[TK]D-Fenderpensmit: You description is bad.
19:15.45pensmityour answer is worse
19:16.04pensmitok so navaismo use local channels?
19:16.16pensmiti didn't even know you were talking to me
19:16.19pensmitlol
19:16.26pensmitD-Fender was cool the other day.
19:16.33pensmitsomeone pissed in his cheerios today
19:16.49[TK]D-Fenderpensmit: show your complete code na dpoint out what you are looking to dial and where exactly
19:16.54[TK]D-Fenderwe need to see the full contexts
19:17.02[TK]D-FenderYour description is missing to much
19:17.09[TK]D-Fendertoo*
19:17.11*** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:7847:3c92:2414:bafc)
19:17.15navaismowrites as an idiot to the air
19:17.29pensmitthen i have to edit out damn phone numbers
19:17.31pensmitthis isn't that hard
19:17.40pensmityou just want something for no reason now
19:17.45[TK]D-FenderNobody cares about your numbers
19:17.50pensmiti do
19:17.56[TK]D-FenderNo, your description is terrible
19:18.03pensmitno you're being a goober
19:18.18pensmiti'm going to look into local channels
19:19.11pensmitcause it appears you have to do that if you dont' want to define an extension that does the same thing 2 already do
19:19.48[TK]D-FenderI don't see 2 at all
19:20.00pensmiti told you there was another pretty much the same
19:20.08[TK]D-FenderYou've shown 1 poorly concieved pattern so far and floded the channel doing it
19:20.12*** join/#asterisk vlad_starkov (~vlad_star@109.188.126.119)
19:20.22pensmityou are in a mood today
19:20.24pensmitlol
19:20.29[TK]D-FenderAnd if it's a pure copy then your code is already full of duplicates due to bad coding
19:20.39pensmitso
19:20.40[TK]D-FenderNo, this is not a bad mood.
19:20.40pensmitworks
19:20.45pensmitlol
19:21.04[TK]D-FenderThis is a fair assessment of coding standards and what you've shown
19:21.14pensmitare you a grade school teacher now
19:21.21pensmitam i supposed to feel bad about it
19:21.23pensmitlol
19:21.25pensmitgimme a break man
19:21.34[TK]D-FenderEarn one
19:21.42pensmitthat's ok
19:21.48[TK]D-FenderYou've burnt a pile of freebies already
19:22.00pensmiti take back my internet beers i gave you
19:22.50*** join/#asterisk jmetro (0cac9c1a@gateway/web/freenode/ip.12.172.156.26)
19:22.54[TK]D-FenderI take back and chance for future support from me
19:22.58[TK]D-FenderThere
19:23.07jmetro[TK]D-Fender: Did anyone ever reply to my post about AGI <.<
19:23.23pensmitDude...i really dont' want it now
19:23.24[TK]D-Fenderjmetro: Yes.  It was "show your code so we can show you the error
19:23.26pensmityou're a dick
19:23.33jmetro:<
19:24.07[TK]D-Fenderpensmit: You ask for help.  You provide no proper details.  You make supporting you a pain.  YOU are not worth the effort.  Go play "secret squirrel" with someone else.
19:24.15*** join/#asterisk vlad_starkov (~vlad_star@109.188.126.119)
19:24.29pensmitGo be a dick to someone else
19:24.33Penguin21 questions went into overtime?
19:24.35pensmitI provided you everything you need.
19:24.37pensmitdone
19:24.37navaismo~secret squirrel
19:24.56pensmitthe problem truly is capbability
19:25.07pensmitwhich you simply dont' have if you can't conceive of what i said
19:26.40[TK]D-Fenderpensmit: [15:10]pensmithow can i create a "ring group" that would go through one of those depending on who answered first
19:26.49[TK]D-Fenderit won't do ANYTHING more if one of them answers
19:27.03[TK]D-FenderYou don't understand DIAL if you're asking for that result
19:27.06navaismocan we just view the jmetro code...
19:27.36jmetronavaismo: I think i found my problem, i'm doing one more test to find out. I had originally done a #include of phpagi.php and forgot to put it on this server..
19:27.49navaismooh
19:28.29pensmitD-Fender, I've been offended by you enough.  I'll figure it out by myself.  Go torture someone else with snide remarks when they ask for help.
19:30.05jmetroOkay, i've got phpagi in there, still not setting my extens, reposting..
19:30.24vlad_starkovQuestion: anyone is familiar with Gigaset C610A IP phone? I unexpectedly got a mystery issue with repeatedly beep-beep-beep sounds when dialing. Just had read the manual and didn't find an answer.
19:32.16PenguinThe ONLY way I know of that one extension can dial two other extensions is with the Local channel.
19:32.35jmetroDial(Sip/Exten1&Sip/Exten2)
19:33.18dongola7jmetro: he wants to redirect to the actual extension in the dialplan, not dial the device directly.
19:33.43PenguinYou've erroneously mixed up devices and extensions.
19:34.03PenguinSIP/device1&SIP/device2
19:34.25PenguinHuge difference when talking dial plan.
19:34.36navaismoHe already solved right? can we move on? I need money hehe
19:35.00PenguinI don't think it got solved.  I think he gave up.
19:35.00jmetrohm
19:35.02pensmitI haven't solved it but thanks for the suggestions.
19:35.09pensmitI'm still looking.
19:35.24jmetroDial(Local) is the way then
19:35.39pensmitThanks jmetro
19:35.41PenguinI'm not going to scroll back to the beginning of time, but if you want one extension to dial two other extensions, the local channel is the ONLY way I can think to make it happen.
19:35.48dongola7Penguin: https://wiki.asterisk.org/wiki/display/AST/Introduction+to+Local+Channels
19:35.56pensmitthanks penguin
19:36.03Penguindongola7: I already know how to use local channels.  :)
19:36.12navaismook dongola7 and I wrote to the air
19:36.29dongola7Penguin: sorry. auto-complete got the better of me :)
19:36.35PenguinI knew it did.
19:36.49navaismonow i can join fender about wasting the time
19:37.32jmetroOk, my agi script isnt erroring out anymore, but i still cant figure out how to set a variable that i can use with asterisk dialplan, posting pb
19:38.27navaismojmetro: are you using phpagi?
19:38.36navaismook ill wait to the pb
19:38.37PenguinSet a variable inside the agi and use it outside the agi?
19:40.25jmetronavaismo: yes indeed, i think i mysteriously solved it but let me find out.
19:41.22jmetroYep. Dont know what was wrong, but this code originally did not work. I commmented out the set_variable line and made a duplicate and it works.
19:41.40navaismoi use: $agi->Set_Varibale("LIFE","42"); then in the dialplan just in the next step NoOP(${LIFE})
19:41.44navaismouh
19:41.45jmetrohttp://pastebin.com/5McF09Ch
19:41.50navaismolatest phpagi
19:41.57navaismo?
19:42.38navaismo<goatvoice>weeeeird</goatvoice>
19:43.31*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
19:46.10[TK]D-Fenderjmetro:      same => n,Agi("test-exten.php") <- no quotes
19:46.29jmetroIt seems to work with it, but if it will work without, looks cleaner.
19:46.32[TK]D-Fenderjmetro: Also you are calling it wrong
19:46.56[TK]D-Fenderjmetro: Your php would have to be EXECUTABLE and also is missing the usual headers
19:47.08[TK]D-Fenderjmetro: This is several missed AGI requirements
19:47.29jmetroAll of those, very valid points, but are taken care of already.
19:47.47jmetroi cut-pasted the code and didnt have the #! in there.
19:48.01jmetroand, you also cant tell that the php is executable from a pb.
19:48.06[TK]D-Fenderjmetro: Don't hold back on the details or we'll point things out as errors
19:48.16[TK]D-Fenderjmetro: And get the output...\
19:49.30[TK]D-Fender[15:47]jmetroi cut-pasted the code and didnt have the #! in there. <- still can't tell if it's right
19:49.50hypknight[TK]D-Fender: I think I'm getting somewhere with it... wanna take a look at this debug for an inbound call? http://pastebin.com/G1HyZm4X     What happens is a hear a brief second of white noise when the call connects and then it disconnects
19:50.13hypknightAll SIP and RTP is supposed to go through that private 172 address
19:51.50[TK]D-Fenderhypknight: No, their invite clear shows the media is elsewhere
19:52.11[TK]D-Fenderhypknight: and it shows that your peer is set wrong.
19:52.19[TK]D-Fender<--- Transmitting (NAT) to 172.16.1.1:5060 ---> <---------------------
19:52.19hypknightmy peer is set wrong?
19:52.20jmetroHm. Error- broken pipe :<
19:52.33[TK]D-Fenderjmetro: Yup, and I'm pretty sure I know why....
19:52.48jmetroIt works, but that error is annoying
19:53.10[TK]D-Fenderjmetro: Show and we'll fix it
19:55.54jmetroDoesnt do it now. I'
19:57.00jmetroyep, no warnings or errors, flawless.
19:57.05[TK]D-Fender\o/
19:57.18[TK]D-Fenderjmetro: so... '-q" right?
19:58.57jmetroNope, i had ended my code with a verbose when i was testing, changed it to a set, and now it doesnt do it anymore.
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19:59.43hypknight[TK]D-Fender: could you throw me a hint as to why my peer is wrong?
20:00.00[TK]D-Fenderhypknight: NAT <-
20:01.14pensmitsame => n,Dial(Local/203@employees&Local/205@employees)  worked like a champ guys
20:01.16pensmitthanks a lot
20:03.34hypknight[TK]D-Fender: Here's with nat=no    http://pastebin.com/8x2jWVxv      it shows (no NAT) but the symptom is the same
20:04.38[TK]D-Fenderhypknight: What symptom?
20:04.54hypknight[TK]D-Fender: half second of audio and a disconnect
20:04.58[TK]D-Fenderhypknight: You are missing basic verbose in there.  We aren't seeing dialplan execute so we have no idea what else * is trying to do...
20:05.49hypknight[TK]D-Fender:  it's a really busy system, I was trying to keep the other stuff out
20:06.00[TK]D-Fenderhypknight: I don't debug half-blind
20:08.37[TK]D-Fenderhypknight: What is this proxy you're routing through exactly?
20:08.50hypknight[TK]D-Fender: with core set verbose 3: http://pastebin.com/kLB1FvR9
20:08.59hypknightit's a provider adtran
20:09.46[TK]D-Fenderhypknight: Who can't hear whom?
20:10.02hypknightnoone can hear anyone... the call immediately disconnects
20:10.07[TK]D-FenderWell.. and an immediate disconnect?
20:10.20[TK]D-FenderOH
20:10.24[TK]D-Fenderwell.. there's the problem
20:10.32[TK]D-Fender<PROTECTED>
20:10.42[TK]D-Fender<PROTECTED>
20:10.43hypknightI'm trying to play a sound
20:10.48[TK]D-FenderAll you did was answer
20:10.52[TK]D-Fenderand ran out of things to Do
20:10.56hypknighthmmm
20:11.00[TK]D-FenderyOUR DIALPLAN FAILED
20:12.37hypknighti completely missed that
20:13.11pensmitD-Fender you shouldn't stop there.  Tell him he lacks comprehension and his dialplan is child like.
20:13.30hypknight...
20:13.40navaismoLMAO
20:13.50PenguinHe hasn't demonstrated those characteristics yet.
20:14.13PenguinGive it time.  If he does, he'll get called out on it.
20:14.36PenguinIf he doesn't, don't worry over it.
20:14.36hypknightway to bolster confidence in asking questions...
20:14.54navaismomy wife look at me with the face: wtf with you, why are you laughing
20:15.50hypknightdoes everyone who asks a question get raked over the coals?
20:15.56[TK]D-Fendernope
20:15.58PenguinYou didn't.
20:16.02[TK]D-Fenderindeed
20:16.10hypknightjust checking.
20:16.49PenguinBut people with poor behaviors more often get fires lit under them.
20:16.55pensmithypknight I'm just crying because D-Fender hurt my iddy biddy feelings earlier.
20:17.48hypknightI'm used to D-Fender... I've been here for years. He is abrasive every time
20:18.01[TK]D-FenderOnly when people waste my time
20:18.07pensmitlordy
20:18.21jmetrohypknight: more like O-fender amirite
20:18.29hypknight[TK]D-Fender: You feel someone asking a question is a time waster?
20:18.42[TK]D-Fenderhypknight: No.
20:19.05PenguinIf you're trying to illustrate a concept to be played out within asterisk, it's a really good idea to learn the terminology prior to the exhibit being displayed.
20:19.18[TK]D-Fenderhypknight: You asked, I looked, I asked for more, you gave.  I found something for you to fix.  No issues there
20:19.37hypknight[TK]D-Fender: Agreed. I was just asking as not to irritate you in the future.
20:19.48[TK]D-Fenderhypknight: So far our conversation traces back as "all clear"
20:20.03hypknightI still don't know why the sound isn't being played...
20:20.14hypknightI'm grabbing my dialplan for you now
20:20.25[TK]D-Fenderhypknight: and the new complete debug...
20:20.39hypknightthe debug hasn't changed
20:20.48[TK]D-Fenderhypknight: It HAS to have.. before it HUNG UP.
20:20.49hypknighti haven't changed anything at all since you saw that
20:21.06hypknighteasy, easy
20:21.07[TK]D-Fenderhypknight: My last call had no playback.  the call DIED, remember?
20:21.10hypknightyes
20:21.15[TK]D-Fenderhypknight: there you go...
20:21.18jmetrohypknight: are you doing a same => n,Playback(tt-monkeys) ?
20:21.20hypknightwould you like me to post an identical debug?
20:21.21hypknightI will
20:21.25hypknightyes
20:21.48jmetrowell in that case you should definitely be listening to screaming primates for a good 10 seconds or so
20:21.48hypknightI even changed it to same => n,Playback(hello-world)
20:21.51hypknightdoes the same thing
20:22.03hypknightmy question is why
20:22.08hypknightbut the debug is the same
20:22.20jmetrodid you show us the dialplan?
20:22.22PenguinIf you've changed the dial plan, the only way the debug is the same is if you failed to reload the dial plan.
20:22.40*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
20:22.50hypknightbut i didn't change it... i just changed the sound
20:22.59PenguinThat's changing it.
20:23.01hypknightit only just says answer() then die
20:23.08hypknighti'll grab it and post it.
20:23.38*** join/#asterisk i_am_good (864a491f@gateway/web/freenode/ip.134.74.73.31)
20:24.03PenguinRemember what I said about reloading.
20:24.26PenguinAlso, I wouldn't even bother with Answer().  Playback() does that for you.
20:24.43*** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou)
20:24.47hypknightI thought it did
20:24.58hypknightbut I wasn't seeing the Answer way back when
20:25.08hypknightso I added it in there so I could at least see SOMETHING
20:25.15PenguinIf you didn't provide one, why would you have seen it?
20:25.27PenguinApps that you do not use do not appear.
20:25.34hypknightwhat?
20:25.34jmetrohypknight: did you post the dialplan?
20:25.53PenguinIf you do not use Answer() in your dial plan, why would you expect to see an Answer() in the verbose output?
20:26.03hypknightI didn't see a Playback()
20:26.07PenguinIf you don't use the Answer() app, it will not appear.
20:26.08hypknightI didn't see anything at all
20:26.12hypknightso I put in an Answer()
20:26.21PenguinLet's see the dial plan.
20:26.27jmetro^
20:27.47hypknightpastebin.com/WVW8FDHQ
20:28.39[TK]D-Fenderexten => n,Playback(hello-world)
20:28.41[TK]D-Fenderthi is not "same"
20:28.42PenguinYou have no extesnions listed past the first line.
20:29.03[TK]D-Fenderthis is a new pattern called "n", with no prioirty in the proper slot
20:29.06[TK]D-FenderSyntax error
20:29.29PenguinYou either have to use  exten => 6516413456,n,   or   same => n,
20:29.45hypknightfair enough, thanks for the clarification
20:30.56PenguinI'm surprised "dialplan reload" didn't throw a big red flag at you.
20:31.09i_am_goodHi, are ASR: 31.1% and ACD: 5.7 bad?
20:31.10PenguinMaybe it scrolled by and you didn't scroll back to read it.
20:31.34*** join/#asterisk jhlavacek (~jirka@87.89.218.63)
20:31.41[TK]D-Fendercheckout time, BBIAB
20:32.52hypknightperhaps
20:34.42jmetrohypknight: why wont you post the dialplan?
20:34.47PenguinHe did.
20:34.51PenguinWe fixed it.
20:34.53jmetrooh its not highlighted.
20:35.31i_am_goodSorry if not directly related to Asterisk but if ACD means Average Call Duration, would this carrier's quality of service be bad? http://tiny.cc/ratelist
20:38.12hypknightSame symptom =\,    pastebin.com/s2GX7HVZ
20:38.36PenguinStill failed at syntax.
20:38.56hypknightI did?
20:39.01hypknightoh hell
20:39.05PenguinYou've got extenstion 6516413456 priority 1.  You've then got extension n priority 2 and 3.
20:39.30PenguinYou should have copied and pasted my last answer.
20:39.33hypknightI actually do apologize... I've been staring at this dealing with Integra for 4 hours now
20:39.36Penguin(1529.28) <Penguin> You either have to use  exten => 6516413456,n,   or   same => n,
20:39.43hypknightI apologize
20:40.23PenguinAnd get rid of the Answer().  It's just wasting resources.
20:42.47Kattymjordan: she's workin like a charm
20:42.56Kattymjordan: http://pastebin.ca/2457908 <- if you're interested.
20:44.04PenguinHoly numbered priorities, Batman!
20:44.10hypknightPenguin: Ok, so we're to the root of the issue I think... I still get no audio, but it doesn't disconnect:   http://pastebin.com/4bzf6rH4
20:46.08PenguinAt least it played the sound (executed the dial plan correctly).
20:47.14hypknightexactly
20:47.18hypknightbut I can't hear anything
20:47.50*** join/#asterisk danjenkins (~danjenkin@cpc3-folk2-2-0-cust34.1-2.cable.virginmedia.com)
20:48.03PenguinI'm not seeing a second INVITE, but did you specify directmedia=no in your peer entry in sip.conf?
20:48.24PenguinI noticed the media was at a public address rather than the local one.
20:48.28*** part/#asterisk aelliott22 (~aelliott@198.41.29.45)
20:48.37PenguinMakes me think you're allowing directmedia.
20:48.50hypknightI've been toggling that
20:48.53hypknightlet me toggle it back to no
20:49.04PenguinRemember sip reload after you're done.
20:49.23hypknightyup!
20:51.11hypknighthttp://pastebin.com/4kdUgy5p
20:52.04PenguinIt's still saying the same thing regarding the media.
20:52.25hypknighti know :(
20:52.38PenguinIs the Adtran device providing a SIP proxy or a SIP server/client?
20:52.45hypknightit's a SIP proxy
20:53.15*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2)
20:53.23PenguinThat's probably why the directmedia setting didn't make a difference.
20:53.41PenguinGive [tk]d-fender that last debug and see what he thinks about it.
20:54.18hypknight[03:49.23] <hypknight> yup!
20:54.18hypknight[03:51.11] <hypknight> http://pastebin.com/4kdUgy5p
20:54.42hypknight[TK]D-Fender: Got the dialplan sorted out, but still no audio
20:54.50PenguinDial plan executes the playback of the sound file, but there's no sound coming from the phone.
20:55.08hypknightdirectmedia=no is set on both contexts
20:56.38[TK]D-Fenderhypknight: Why is it that you have this proxy on your LAN exactly?
20:56.51hypknight[TK]D-Fender: becuase they make me do it that way
20:56.57[TK]D-Fenderhypknight: because this looks like it's likely to cause som serious sourcing issues
20:57.20[TK]D-Fenderhypknight: Does this thing actually VPN somewhere?
20:57.23hypknight[TK]D-Fender: the primary IP of this server is a public one, the 172.16.x.x address is a second NIC, with no gateway on it
20:57.39hypknight[TK]D-Fender: i don't know what the provider does with it once it hits .1
20:57.44[TK]D-FenderYour * server has a public IP exposed?
20:57.49PenguinIt can get even worse when I'm told to go put a fortigate in front of the adtran.
20:57.52[TK]D-FenderDirectly on an interface on it?
20:58.10hypknightit's publicly addressed yes
20:58.19[TK]D-Fenderhypknight: Please elaborate
20:58.57hypknight[TK]D-Fender: eth1 <public IP with Gateway as normal> Provider wouldn't touch that and demanded private space on the inside of their Adtran
20:59.14hypknight[TK]D-Fender: added a second NIC, with no gateway, 172.16.1.2 in the same /30
20:59.34hypknight[TK]D-Fender: at the network layer it works fine, pings and whatnot
21:00.34[TK]D-Fenderhypknight: problem arises that when your server tries contacting the remote for audio, it's told to go to a remote public IP.  That will go out your primary interface
21:00.49[TK]D-Fendermeanwhile the SIP is still pointing to the local gateway
21:00.54hypknightright
21:01.06[TK]D-FenderAnd you tell THEM that your audio is on a PRIVATE IP.
21:01.12[TK]D-FenderThis is a circular clusterfuck
21:01.29[TK]D-FenderThey tell you public, you tell them private
21:01.34hypknightI tried to do the register against their proxy
21:01.40hypknightit won't accept it
21:01.42[TK]D-Fenderthat's not the issue
21:01.51[TK]D-Fender* sorta needs to tell them your audio is on your WAN IP.
21:01.52PenguinYou'll have to set up a static route.
21:02.00[TK]D-Fenderthat's the problem of this being dual-zoned
21:02.12[TK]D-FenderPenguin: It's worse than that...
21:02.47[TK]D-FenderPenguin: its the RTP srouce that * advertises as private even as the remote, though passed through a private proxy... is still a public IP
21:03.02PenguinEww.
21:03.09[TK]D-Fenderyup
21:03.16[TK]D-Fenderdual-homed HELL
21:03.27PenguinThat says to me that the only way to fix it is stop with multi-homed asterisk.
21:03.30hypknight[TK]D-Fender: ok, so I'm retarded. Is what they're trying to make me do going to work at all?
21:03.50[TK]D-Fenderhypknight: I can't see how to keep the public and still do that...
21:03.57PenguinIt sounds like a limitation of asterisk's chan_sip.
21:03.59[TK]D-Fenderhypknight: What is it doing sitting publc?
21:04.02hypknightI have no choice on the public
21:04.10hypknightit's in a university, the entire subnet is public
21:04.19PenguinMaybe I'm reading it wrong, but that's what it sounds like.
21:04.32hypknightthere are firewalls at the perimeter and whatnot so not to debate that.
21:04.39PenguinWhy can't you set up asterisk on a box with ONLY a private address?
21:04.39hypknightbut we're saying * can't multi-home?
21:04.44PenguinIt can't.
21:04.45[TK]D-Fenderpeng`You could look at it like that... but this is a nasty circunstance.
21:04.52[TK]D-Fenderhypknight: not like that.
21:05.02PenguinIt's terrible at trying.
21:05.02[TK]D-Fenderhypknight: There are solutions however
21:05.09PenguinPeople have been attempting it for years.
21:05.17hypknightwell crap.,
21:05.25hypknightIntegra is going to be the death of me....
21:05.26[TK]D-Fenderhypknight: Run another * instance as a gatekeeper
21:05.31hypknightok
21:05.42hypknightthen IAX them?
21:05.42PenguinMake sure it only has a private address.
21:05.45[TK]D-Fenderand go * > * internal
21:06.03[TK]D-Fenderprotocol's between the *'s don't matter.  They are all -local to each other in this plan
21:06.08hypknightcrap
21:06.17hypknightok
21:06.38PenguinBut if you can put asterisk on another box with only a private address, why would you need to keep the other one that is dual-homed?
21:06.39hypknightwell, actually, isn't that the same problem?
21:06.49hypknightbecuase it's an existing environment
21:07.05hypknightand it's a univeristy, running nearly 500 stations on asterisk
21:07.07PenguinCan yo uset up another one on the LAN?
21:07.07[TK]D-FenderPenguin: nope.  the relay * will just be local.
21:07.17[TK]D-FenderPenguin: it doesn't have to worry about how to forget the packets
21:07.26hypknightPenguin: yea, I can make that work
21:07.28[TK]D-Fenderforge*
21:07.51PenguinIf you can set up another box with private only, run all your media through it.
21:07.54[TK]D-Fenderhypknight: This provider of yours is a lot of trouble
21:07.55mjordanKatty: cool :-)  Glad the evil approach worked
21:08.09hypknight[TK]D-Fender: tell me about it... I don't like them, but it wasn't my choice
21:08.23hypknightI've been deploying * since 1.4, never had to jump through this may hoops...
21:08.32[TK]D-Fenderhypknight: You could try to sell them that you "ca't" (tm) do it... and source another...
21:08.36[TK]D-Fendercan't*
21:08.43hypknightcan't do that either
21:08.49hypknightthey just signed the contract
21:08.59PenguinAsterisk is fine on a LAN and also fine on the internet.  It's not so good when you try to do both at the same time.
21:09.02[TK]D-Fenderhypknight: bad order
21:09.06hypknightabove my pay-grade... I just get handed the pieces and told to make it work
21:09.18[TK]D-Fenderhypknight: Validate install.  Sign contract :)
21:09.30PenguinWelcome to network engineering!
21:09.47mjordanKatty: btw, I think the DB function would also work for you - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB
21:10.04hypknightSOOOOOO, we're saying: CPE Adtran -> * Gateway with 172.16.1.2 address -> Existing * box with an additoinal 172.16.1.X address right?
21:10.43PenguinIt's hard to see when you put it like that, but I think that's right.
21:10.57PenguinJust set up another asterisk box on the LAN with only a private address.
21:11.04[TK]D-Fenderhypknight: adtran -> (new * server on 172....) ---> (MAIN * server with PUBLIC IP)
21:11.13PenguinPeer your other asterisk to the new one.
21:11.33hypknightso you want me to NAT, the 172  one toward the public one? if not, the main one will still need to multihome
21:11.49PenguinDon't try to complicate it.
21:11.55hypknightI'm not complicating it
21:12.06hypknight172.16 addresses can't route to a public address
21:12.10PenguinAdtran has a private IP address on your LAN.
21:12.17[TK]D-Fenderhypknight: Shove a new one between.  This new intermediary gateway server will ALWAYS advertise the WAN IP.
21:12.34[TK]D-FenderWhen calling your main, the main will IGNORE that IP (nat=yes) and media will survive
21:12.52hypknightright, but how do I get it there?
21:13.01hypknightat a network level I'll have to NAT
21:13.04[TK]D-FenderStill via the local IP.
21:13.16hypknightso the main * is still multihomed
21:13.23[TK]D-Fenderthe interface will still route the LAN target on the LAN, but advertising the WAN IP which your main will ignore
21:13.38PenguinI still can't understand why your asterisk needs to be on that dual-homed box.
21:13.39[TK]D-Fenderhypknight: No, it's a question that your main won't have to know which IP to present <-
21:14.02PenguinWhat is the benefit of having it there?
21:14.16hypknightPenguin: it's an anchor and can't move right now... not an option
21:14.18PenguinIf it worked before, why do you suddenly need to set up this adtran device?
21:14.27hypknightbecuase they switched providers
21:14.34[TK]D-FenderSource is from a LAN (proxy), they ask for WAN for contact.. but you already consider them LOCAL, because if you don't then you ignore THEIR SDP and you're fucked in the other direction.
21:14.39[TK]D-Fenderthat's the nasty part of this
21:15.09hypknight[TK]D-Fender: at a network level is what I'm talking about... I can't have a 172.16.1.2 addressed server talk to a public one without nat
21:15.09[TK]D-FenderSo you need one box in the middle that doesn't know any better and ALWAYS lies ans says the WAN IP and that when sending calls to the main, the main can casually ignore that
21:15.21[TK]D-Fenderand that's FINE <-
21:15.25[TK]D-FenderSo do it
21:15.26PenguinIf they switched providers, why do you need to keep the public address on the asterisk box?
21:15.32[TK]D-Fenderand forward the apropriate ports
21:15.40hypknightPenguin: the WHOLE environment is public
21:15.44hypknight[TK]D-Fender: ew
21:15.54[TK]D-FenderPenguin: if they switched providers... we wouldn't have this invasive shit proxy to deal with :)
21:16.09hypknightOr, I can just demand that Integra pull their head out and conform to our IP scheme
21:16.23[TK]D-Fenderhypknight: That would be ideal
21:16.25PenguinI thought I understood the topology perfectly, but then we kept talking.
21:16.51[TK]D-Fenderhypknight: If they aren't VPN-ing through their gateway then there is no security or quality point for its existance
21:18.28*** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou)
21:18.33PenguinMaybe this is a stupid idea, but what if you configured another SIP proxy, such as kamailio, to deal with the addressing problem?
21:19.15[TK]D-FenderPenguin: If you want to suggest that as an alternative product,... sure, but it's another tool to learn and administer
21:19.41PenguinPoint taken.
21:19.45[TK]D-FenderPenguin: And I'm not sure what it can and can't do with this specific scenario
21:20.04[TK]D-FenderPenguin: Perhaps you have experience to that effect, but it's something I've never touched
21:20.32[TK]D-FenderPenguin: the one common thing I heard is that Kamailio and it's ilk are NAT-HELL.  Not sure how much is "truth" these days...
21:20.39PenguinNot enough to say "do it."
21:21.44[TK]D-FenderVery sad situation to have to deal with....
21:21.48[TK]D-FenderBRB
21:23.50*** join/#asterisk robl^ (robl@pdpc/supporter/active/robl)
21:24.29hypknightI think what I may do is use Freewswitch to solve the issue
21:24.36hypknightI think Integra is going to tell me to go to hell
21:26.08PenguinDo you know how to admin freeswitch?
21:28.39hypknightyea
21:28.52hypknightI actually have code contributed to that project... very familiar with it
21:29.14PenguinThen perhaps you can make it do exactly what you need done.
21:29.16hypknightI think I can use it as a B2BUA
21:36.05[TK]D-Fenderhypknight: possibly an option as that gateway
21:36.24[TK]D-Fenderhypknight: I'd like to think yo could do that much with it.
21:42.16*** join/#asterisk kpettit (~kpettit@99-116-144-119.lightspeed.hstntx.sbcglobal.net)
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22:06.22*** part/#asterisk mjordan (~mjordan@nat/digium/x-mdcsnvqeqzsyqsrm)
22:06.40snadgeim starting a job today that heavily involves asterisk
22:06.52snadgewhat does asterisk do? :P
22:12.30newtonrsnadge, http://www.asterisk.org/get-started
22:13.27navaismols *
22:14.18snadgethe what does it do was kind of a joke.. but that get started link is very informative, thanks newtonr
22:18.00snadgeimpressed by the number of asterisk courses there are too
22:18.35snadgenone in australia.. oh well ;)
22:18.48snadgeexcuse to travel to las vegas maybe?
22:21.32*** join/#asterisk hypknight (~hypknight@unaffiliated/hypknight)
22:22.04*** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug)
22:22.45hypknightok, so I got Integra to conform to our public IP address scheme... but it's doing the same thing: http://pastebin.com/TV8QtWdB
22:23.03*** join/#asterisk TimeRider (~steve@timerider.plus.com)
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22:27.19Kattylooks in
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22:43.38volga629Hello Everyone, Is asterisk should determine which H.264 Payload Type "codec identifier" to use ? If in SDP get something different ?
22:51.09*** join/#asterisk Blashyrkh (~chatzilla@188-23-196-169.adsl.highway.telekom.at)
22:51.51Blashyrkhi have a question regarding sippy b2bua and rtpproxy taht should act as border control for my asterisk
22:52.42Blashyrkhif i start the b2bua with the -r flag, do i need to start an extra rtpproxy instance for it to work that listens on the same ip:port?
23:03.19*** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone)
23:05.29hypknight[TK]D-Fender: ping?
23:32.12*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2)
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