IRC log for #asterisk on 20140510

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00:09.57jameswf${MONITOR} * Set to "TRUE" if the channel is/has been monitored (app monitor())
00:12.58Mango45Thanks!
00:18.26Mango45Curiously, that doesn't work for me.  The variable is empty.
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00:48.10mushroomedCan someone help me creating a simple ConfBridge? I don't want to type PINCODES, I just want the functionality
00:48.58mushroomedI've already read voip-info.org ConfBridge cmd but it's too extense, I wan't to keep it simple
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01:51.11WIMPymushroomed: There isn't much to do. You just need a default template for users and conferences in confbridge.conf and tehn just call it.
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02:57.22ChannelZIt's so easy even a caveman could do it
02:58.03MaliutaLapChannelZ: That's the case with most * stuff though ... if people would read the damn docs
02:58.40WIMPyThere are plenty if issue without docs explaining what's going on.
03:01.17MaliutaLapWIMPy: for dumb people ;)
03:01.46MaliutaLapI really should get that "I see dumb people" shirt
03:01.46WIMPymust ver very dumb then.
03:02.42MaliutaLapWIMPy: you seem pretty clued in from what I have seen
03:03.03WIMPyDefinitely not enough.
03:04.02MaliutaLapWIMPy: one can never be clued in "enough", there is always something more to learn ... it's one of the reasons I like IT work
03:04.15MaliutaLapnot that I like the people I do the work for
03:04.37WIMPyOnly if there is a solution in sight.
03:04.49MaliutaLapbut that's just me and "people". Doing tech support for a couple of years will do that to one
03:05.12MaliutaLapWIMPy: bend the tech to your will and there is almost always a solution :)
03:05.49WIMPyThe trouble is finding the right spot to bend.
03:06.13WIMPyAnd My Asterisk regularly fails after upgrading because I forgot one of the patches.
03:06.27MaliutaLapwhat are you patching for?
03:07.06WIMPyA few little things, but the one I tend to forget is the one that makes outbound calls possible.
03:07.30MaliutaLappatch for some odd kind of card?
03:07.44WIMPyNo, SIP.
03:08.15WIMPyMy ITSP kills calls if it gets a NAK for AOC messages.
03:09.58MaliutaLapso you're patching for a fecked up ITSP ... great
03:10.21WIMPyI consider SIP fucked up by design.
03:10.26MaliutaLapwhat have they done, forked SIP and created their own custom version? ... or is just crappy
03:10.40WIMPyEveryone does so.
03:11.08MaliutaLapI've been through a few providers and haven't found anything too bad
03:11.08WIMPyYou know what's great about standards?
03:11.16WIMPyEveryone has his own.
03:11.28WIMPyThat's nowere as true as for SIP.
03:11.56WIMPySo I consider SIP interoperability pure luck.
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03:13.06WIMPyI mean, pure SIP is rather useless. So it comes down to what extensions are implemented and in what way.
03:13.34MaliutaLappffft SMTP interoperability is a freakin' miracle
03:13.52MaliutaLapand SMTP has been around a lot longer than SIP
03:14.02WIMPyCompared to SIP that ALWAYS works.
03:14.21MaliutaLaplaughs
03:14.25MaliutaLapnot always
03:14.46MaliutaLapthere are some pretty screwed up SMTP implementations out there
03:14.47WIMPySeriousely, what kind of troble do you have with SMTP? I can't remember anythign I came across.
03:16.00MaliutaLapOh, you see issues with people not responding to greylisting properly, and most of the code in stuff like postfix is actually to detect and deal with borked implementations of a [simple] standard
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03:16.50MaliutaLappeople not responding to "temporarily unavailable" codes is becoming more common
03:17.01WIMPyI have seen implementations that allow obvious spam while filtering legit mail for bogus reasons, but that's not part of SMTP.
03:18.36MaliutaLapthat has more to do with peoples configurations
03:18.49WIMPyIndeed
03:18.51MaliutaLapand the spam filtering software they use ontop of that
03:19.35MaliutaLapbut things like exchange ignore parts of the protocol, at least in terms of what comes before the '@' symbol
03:20.19WIMPyYou don't seriousely expect anythign form Microsoft to be compatible to known standards, do you?
03:20.42MaliutaLapno, that's why I don't touch any of their shit
03:22.58MaliutaLapI like to say: "Windows are for opening up and jumping out of, 1929 stock broker style, as a last resort"
03:25.58WIMPywonders what Asterisk would be...
03:26.31MaliutaLapa pixie in sky? ;)
03:27.12WIMPyProbably the sledge in a time when the wheel not available.
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07:15.54bhavikpatel6842Hello everyone
07:16.16bhavikpatel6842I am facing one issue regarding  webrtc using ASterisk
07:16.34bhavikpatel6842I am trying to integrate SIPml5 demo.
07:16.42bhavikpatel6842my Asterisk version is 11.9
07:17.20bhavikpatel6842I am not able to sip to sip calls using two chrome browser which is latest version.
07:17.39bhavikpatel6842I use this link for reference   p, li { white-space: pre-wrap; }  https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
07:17.58bhavikpatel6842configuration and settings
07:17.58bhavikpatel6842http://pastebin.com/7KCvtcNf
07:18.25bhavikpatel6842Sip dialogs for
07:18.25bhavikpatel6842incoming calls : http://pastebin.com/ykeS0JCi
07:18.25bhavikpatel6842and outbound call log :
07:18.25bhavikpatel6842<PROTECTED>
07:18.35bhavikpatel6842Let me know if I am doing wrong.
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08:00.50songoku1610hi
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10:18.38cuscohi
10:21.33cuscoI must be missing something to add a member to a queue with hint'ed state interface
10:22.02cuscoqueue show <queue> shows: 150 (Local/150@agents) (dynamic) (Invalid) has taken no calls yet
10:23.26cuscois there a queue option needed?
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11:11.09margeashello ppl, I need to set-up the SIP caller-Id in asterisk11 in e164 format, and the service provider is expecting a SIP header like this "From: "447937947990" <sip:APIKEY@sip.nexmo.com>". I'm using the SipAddHeader() dialplan function but it seems I end up with *two* From: header lines in the sip dialogs, one added automatically by asterisk, the second it's the SipAddHeader one.How can I replace the one's added by asterisk,
11:11.10margeaswithout adding another one?
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12:12.23margeasDoes anybody know how I can "replace" the "From:" sip header before calling a peer? thank you
12:21.01fileyou can't replace it but you can configure in sip.conf aspects of it
12:21.17filefromuser to control the user portion, fromdomain to control the domain part
12:27.26margeasfile: could you please look at https://docs.nexmo.com/index.php/voice-api/sip, the callerid portion...
12:28.28filethey've given you the portion for domain and APIKEY, looks like they are using the name portion to convey number
12:28.41fileso you can set that using Set(CALLERID(name)=447937947990
12:28.46filein the dialplan before dialing
12:29.05margeasfile: ok let me try..
12:34.12cuscotaking the oportunity that there is som activity here...
12:34.29cuscoqueue show <queue> does not show member with hint
12:34.38cuscoI don't quite understand why
12:34.43cuscoand shows as Invalid..
12:34.53cuscoif I set state_interface to SIP/bla i'ts ok
12:34.58cuscobut to hint:ext@context
12:35.04cuscodoesn't show in the queue
12:35.06cusco....
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12:37.30margeasfile: after adding this to the dialplan: same => n,Set(CALLERID(name)=447937947990) the From header is From: "Anonymous"<sip:74f2ecf3@anonymous.invalid>;tag=as383edf3d back from nexmo and I get a SIP/2.0 401 Unauthorized
12:39.20margeasit's like if the callerid name get not set...while if I add the number in the name portion of my phone set, I get authenticated and the call is successful
12:40.29fileadd Set(CALLERID(name-pres)=allowed)
12:42.39margeasfile: THANK YOU! It's finally working...what is that name-pres? I haven't found nothing related in the docs...
12:42.46fileit's presentation
12:43.29filecallerid can be marked with different presentation values, so for example if someone blocks their callerid it may traverse the PSTN with their callerid until it reaches the end... where it is then supposed to not send it to the end device
12:45.17margeasso I guess the default is to *not* present the callerid which is why my number was not passing through?
12:45.45fileit's not, without knowing more about the setup I couldn't say why
12:46.22margeasfile: well It's all there in that link from nexmo
12:46.40filethat's only outgoing
12:46.44filethat's not inbound
12:46.58filecusco, what is your queues.conf and the output of queue show <name>?
12:47.37margeasfile: I don't have an inbound number with them so my configuration is set-up to use nexmo only as peer....
12:47.54margeastype=peer
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12:47.59filemargeas, what is initiating the call?
13:02.13margeasfile: just a moment...
13:03.26cuscofile: queue is configured in realtime database
13:03.41cusco999 (Local/999@agents) with penalty 2 (dynamic) (Invalid) has taken no calls yet
13:04.07cuscowould queues.conf still be important?
13:04.31fileno, but it just means I can't reproduce it without a lot of effort
13:04.39filefrom config file I can say that looks different and does work
13:04.50cuscoI see but I can select its configuration from sql
13:04.56filehint:1000@users (Local/1000@users from Local/1000@users) (ringinuse disabled) (Not in use) has taken no calls yet
13:05.24cuscoyes I don't know why it doesn't show hint in queue show <name>
13:06.07cuscoalso queue members are configured in realtime sql
13:06.14cuscoI tried adding it dynamically
13:06.20cuscobut same
13:06.30filewhat version of Asterisk?
13:06.38cuscowell yea.. 1.6.2
13:06.39cusco:(
13:06.45fileuhhhh
13:06.51cuscohides
13:07.16file1.6.2 app_queue does not have support for using hints like that
13:07.40cuscobut I found on the interwebs people using hints in 1.6.2
13:07.49cuscowell would there be anotehr way?
13:07.51filehints exist
13:07.59filebut app_queue does not support using them
13:08.03file(in that version)
13:08.04cuscoat the moment we use group_count in dialplan ...
13:08.48cuscowe tried migrating to 11
13:08.55cuscobut had to rollback
13:09.02cusconeed to try that again :/
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13:35.03RicmacasHello everybody, I'm trying to send a sip MESSAGE through a trunk proxy
13:35.21Ricmacasand since I'm a noobie, I have no idea, can anyone give me a hand?
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14:05.07RicmacasUhm, should I repeat my question?
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16:48.48Kobazhmm
16:49.00Kobazis there an easy way to play generic comfort noise just all by itself
16:49.18Kobazof course you can find some and stick it in music on hold
16:49.28Kobazbut is there a way to generate it
16:50.11[TK]D-Fender* doesn't generate it
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16:54.07Kobazdo de do
16:54.13Kobazbuilding an outbound dialer
16:54.33Kobazi have until july
16:54.52Kobazshould be able to make a kick-butt dialer by then
16:57.16Kobazhttp://soundbible.com/suggest.php?q=background+noise&x=0&y=0
16:57.18Kobazkind of close
17:01.33[TK]D-Fendertotally different.
17:01.44Kobazyeah but i can play with it
17:01.46[TK]D-FenderCNG is generated by the client to cover missing audio
17:01.54[TK]D-Fenderthat isn't really "background"
17:01.57Kobazi'm not using it for that
17:02.20[TK]D-FenderYou really haven't been too clear on what you actually intend
17:02.33Kobazlike, an agent is going to dial into a bridge basically
17:02.48Kobazand then be connected with calls that the dialer finds
17:02.59Kobazso i want some sort of subtle indication that the agent is connected to the system
17:03.28Kobazand it's not going to be MusicOnHold(default)
17:03.35Kobazthat will drive people insane
17:05.10[TK]D-FenderSounds like you just want non-music..... music
17:05.16Kobazyeah
17:05.18Kobazexactly
17:05.22[TK]D-Fenderthat it's a thing... it's your set-list
17:05.22Kobazlike comfort noise
17:05.39[TK]D-FenderGo by some cricets, record them and feed a local iguana later
17:05.46Kobazhehe
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17:09.31Kobazwhere the heck is the option to convert from stereo to mono in audacity
17:10.03Kobazoo
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17:10.04Kobazthere it is
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17:40.06haydenanyone using pjsip in production already?
17:42.47fileI have heard of people using it.
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18:07.29Ricmacas[14:35] <Ricmacas> Hello everybody, I'm trying to send a sip MESSAGE through a trunk proxy [14:35] <Ricmacas> and since I'm a noobie, I have no idea, can anyone give me a hand?
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18:10.49[TK]D-FenderRicmacas: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_MessageSend
18:11.12Ricmacas[TK]D-Fender: That's what I'm using
18:11.28Ricmacasthough when I attempt to send the message, it sends it "outside" the proxy
18:12.20[TK]D-FenderPASTBIN is your friend.... show us all of the related bits & debug
18:12.22[TK]D-Fender~pb
18:12.22infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
18:12.24[TK]D-Fender^^^
18:12.42RicmacasOh, I'm actually debugging with Wireshark
18:12.54RicmacasWhich I'm not sure if its fine
18:13.15Ricmacasbut what happens is quite simple, the host sms.sip.provider.com doesnt exist
18:13.32Ricmacasbut if i launch the provider's sip client, it communicates with it through the proxy
18:13.50[TK]D-Fender* SIP debug is all you should be using
18:14.09Ricmacasso, i cant use MessageSend(number@sms.sip.provider.com) because that will attempt to resolve sms.sip.provider.com
18:14.26Ricmacaswhich isnt even a public domain, its only accessible through the trunk's proxy
18:15.33Ricmacaswhich requires authentication, of course.
18:15.54[TK]D-FenderDon't specify the host then, use a peer
18:16.05RicmacasOk, how so?
18:16.17[TK]D-Fenderchange what you put after the @
18:16.34[TK]D-Fenderand make a peer for it
18:17.53RicmacasUhm...
18:18.32RicmacasYou mean that I create a trunk for that domain?
18:18.46[TK]D-Fender[14:16][TK]D-Fenderand make a peer for it
18:19.12[TK]D-Fendersip.conf.  [thisisapeer] type=peer ...
18:19.17RicmacasI have a sip.provider.com peer
18:19.24[TK]D-Fenderthat is a HSOTNAME
18:19.36[TK]D-Fendernot an asterisk SIP peer entry in sip.conf
18:19.43Ricmacaslet me be more specific
18:19.49[TK]D-Fender[14:19][TK]D-Fendersip.conf. [thisisapeer] type=peer ...
18:19.55[TK]D-FenderMAKE THIS
18:19.57RicmacasI have a [provider] type = peer
18:20.00Ricmacasok no need for caps
18:20.22[TK]D-Fenderand then use it instead of a hostname on your MEssage()
18:20.33Ricmacasso i'd use
18:20.42Ricmacasnumber@peer?
18:20.51[TK]D-Fenderyes.
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18:31.08mic_hello
18:31.17mic_the pain of dtmf length when a call is bridged
18:31.30mic_when asterisk intercepts dtmfs and generates new ones
18:31.46mic_should I jump off the bridge before attempting to fix that/
18:31.47mic_?
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18:38.27puzzledhi
18:40.00puzzledAnyone have an idea why Set(SAFE_EXTEN=${FILTER(0-9,${EXTEN})}) would give this error: ast_yyerror():  syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: s !=
18:40.31puzzledI'm using asterisk 11.9.0
18:40.49[TK]D-FenderShow us the actual call.
18:41.31puzzled[TK]D-Fender: hi, log or dialplan?
18:41.41[TK]D-Fenderactual call
18:42.31puzzledok, just a sec
18:44.48[TK]D-FenderWhile you're at it.. why not the full dialplan around it as well...
18:45.03puzzled[TK]D-Fender: http://pastebin.com/eZ6KQWbK
18:45.53[TK]D-FenderSet(SAFE_EXTEN=${FILTER(0-9,${EXTEN})}) <- this is certainly pointless given where you are
18:47.31[TK]D-FenderAnd that is an unexpected end... because it returns nothing
18:48.09puzzled[TK]D-Fender: dialplan snippet http://pastebin.com/xTyLXeCJ
18:48.37[TK]D-Fenderdidn't end up needing for the above assessment...
18:50.12puzzled[TK]D-Fender: I'm afraid I don't get your assessment. Why pointless? EXTEN is 8099 and I want to deduct SAFE_EXTEN. Just like in the README.seriously.
18:50.35[TK]D-Fender"EXTEN is 8099" ... no ... no it isn't
18:51.23puzzledyou mean because of the hand-off via DSTEXTEN to the database context?
18:51.47[TK]D-FenderBecause of the dialplan extension
18:52.45puzzledsorry I don't get what you mean
18:53.33[TK]D-Fender<PROTECTED>
18:54.43[TK]D-Fenderhint : it's immediately after the first [
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18:57.48puzzled[TK]D-Fender: got it, thanks
18:58.43[TK]D-FenderNever forget what ${EXTEN} is.....
18:59.49[TK]D-Fendersame => n,Set(EXTEN=${DSTEXTEN}) <- and that it's READ-ONLY ... which makes this pointless.
19:00.48puzzledgot it, the read-only part escaped me
19:00.54[TK]D-FenderAnd you have a better variable you could be using...
19:01.17[TK]D-Fenderexcept that if your coding is expected to be consistant, you never needed to set it like that at all
19:01.27[TK]D-FenderYou should learn to pass args to your Gosubs....
19:03.41[TK]D-Fenderhttp://pastebin.com/y8w3KUR6
19:04.19[TK]D-FenderBut then another idea arises... look where you're using it in the first place... from a pattern that can never fail the filter.  Do you have it from one that should?
19:04.25[TK]D-FenderSomewhere else....
19:05.06[TK]D-FenderAKA "where am I calling this from that can actaully fail and validates the filter at all?"
19:05.27puzzlednope I was just playing with new stuff in 11. Thanks for the Gosub snippet. That's much easier
19:05.44puzzledif I allow guests than that's going to be at the top :)
19:05.45[TK]D-FenderGotta read your apps instructions :)
19:06.00puzzledvery much so. mea culpa
19:07.13[TK]D-FenderIt's a question of validating whether any of your patterns necessitate a filter at all.  "guest" doesn't imply much by itself
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19:10.19puzzledhmm the core show do suggests Gosub(database,s,1(${EXTEN})) unless I'm reading that wrong
19:10.43puzzledGosub([[context,]exten,]priority[(arg1[,...][,argN])])
19:11.03[TK]D-FenderYes, bracket was called for there, not commas
19:11.07[TK]D-Fendermy bad on that one
19:11.29puzzledno problem. it made me read stuff and question it so that's a good thing :)
19:11.37[TK]D-FenderI mirrored Macro syntax reflexively
19:11.59[TK]D-FenderCrying shame they couldn't have actually done it tht way...
19:12.47puzzledI haven't seen that notation before. not very consistent especially coming from 1.4, 1.2 and ancient versions
19:15.00[TK]D-Fenderone of the 1.6 branches changed  Gosub's syntax to includ args and that's hot they chose to do it.
19:15.03[TK]D-Fenderbad call.
19:17.41puzzledyup, guess I'll be doing a lot of core show application ...
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