IRC log for #asterisk on 20120908

00:08.59WIMPyDid someone say it's a bad idea to use hardware that's not supported by Linux?
00:10.09*** join/#asterisk YoMomma (~YoMomma@mail.jkrsi.com)
00:11.51WIMPyFirst I thought my kernel might be too new, but it happens on 3.1.7 as well.
00:12.02WIMPyIn function 'copy_from_user', inlined from 'dahdi_chan_write' at /usr/src/telephony/dahdi/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/dahdi-base.c:2447:23:
00:12.23WIMPy/usr/src/linux-3.1.7/arch/x86/include/asm/uaccess_32.h:211:26: error: call to 'copy_from_user_overflow' declared with attribute error: copy_from_user() buffer size is not provably correct
00:12.28WIMPySomeone know what causes that?
00:16.47*** join/#asterisk chris_n (~Chris@nc-63-162-197-228.sta.embarqhsd.net)
00:21.36kresp0sorry WIMPy, I dont know what could be
00:21.52kresp0good night all, I'm going to bed
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00:27.32chris_nwill a dsl signal cause fxotune to mis-tune the line impedance?
00:34.11WIMPy"strict copy size checks"
00:37.05sruffellWIMPy: yes, that's because the size of the copy is a calculated at run time….generally not preferred.
00:40.38sruffellI probably should just fix that.....
00:41.32WIMPyThat would be best. Or else an pinter to that option would be good sor a start.
00:41.54WIMPya pointer
00:42.17sruffellyeah…I'll just fix it here.  Every minute someone scratches there head about that is a wasted minute.
00:42.20chris_nfwiw, dsl does appear to cause problems
00:47.35carrarindeed
00:47.45carrareveryone should have FTTH
00:48.01WIMPy+1
00:48.19sruffellFTTH?
00:48.25carrarFiber
00:48.26carrarTo
00:48.27carrarThe
00:48.28carrarHome
00:48.33sruffellahh….I'm jelly
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00:51.46chris_nwill be replacing dsl w/metro-e in a couple of weeks
00:51.59carrarI'm 70' away from a 10gig circuit, but Century link will only support 3 mbps DSL here at my house
00:53.01carrarSo I have two of em bonded together
00:53.04carrarand a T1
00:53.10carrarseems to work ok
00:53.20carrarSLOW though
00:58.54chris_nI'm 18K feet away from one co and 23k feet from the other; I have 2 3M dsls along with numbers from two different exchanges
00:59.16chris_nI scraping it all in favor of a pure voip solution over the metro-e
00:59.37chris_nhopes to loose a lot of headaches
00:59.43WIMPyNo number portability?
00:59.59chris_nwe plan to port over the two published numbers
01:02.22chris_ninteresting; at 11.999KHz fxotune says: echo ratio = 0.0077 (84.4 / 10988.4)
01:02.31chris_ntoo bad we can't hear there
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01:11.50carrar11khz
01:11.53carrarYou can't hear that?
01:12.48WIMPyOn an analogue phone?
01:12.53carrarthats like smack dab in the middle of the audible range
01:13.01carraroh
01:13.46carrarPOTS is what, 180 Hz to 3.2 kHz
01:15.19WIMPySomething in that range, yes.
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01:20.25chris_nsorry MHz
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01:21.12chris_nPOTS is 300-3500Hz iirc
01:23.28jpsharpYep.  Nyquist says top end is 4Khz with an 8Khz sampling rate
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01:30.42WIMPyOk, so more than an hour later libpri won'tcompile voluntaily.
01:30.49WIMPy+r
01:31.45sruffell:/   what?  Today has been the day of looking at compiling issues.
01:32.37WIMPyNo return in non void returning function pri_bridge.
01:32.45sruffelllooks
01:33.42sruffellO.O sure enough…no compile error on my main system…let me try on some of the others...
01:36.08WIMPyOk, finally getting some reaction.
01:36.37sruffellwow…on my new Ubuntu server install..I hit that error.  That function is mostly old….
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01:36.44sruffellhttp://svnview.digium.com/svn/libpri?view=revision&revision=19  <— from 10 years ago
01:37.12sruffellprogress!
01:37.14sruffell:)
01:37.40WIMPyHmm
01:38.00sruffellI'll get a patch for that as well…..still carefully testing the earlier one.
01:41.19sruffellAlthough, I guess it's all a good sign.  I like it seems there are alot of people updating their systems, presumably to test out the beta…so it's all good.
01:46.04WIMPyNow I just need to find the channels...
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02:00.27sruffellWIMPy, for these…..what do you want me to put in the commit message for reporter?  Reported-by: <>@<> ?
02:02.40WIMPyHmm, what's the format jira uses?
02:03.19sruffellIt's all messed up now I believe…..Jira usernames don't match handles, etc..  In DAHDI I've been following the kernel convention of using a real identity.  But I'll put whatever you want.
02:03.20WIMPyWhile looking at that, I wonder if ASTERISK-16570 is still true.
02:04.10sruffelllol  2 years
02:04.35WIMPyI don't really care. I used Birger "WIMPy" Harzenetter on Jira so you can make the connection between real name an nick.
02:05.08WIMPyAt them moment I wonder why I get only one port.
02:05.28WIMPytzafrir: Are you around?
02:05.32sruffellbtw….I don't see any movement on ASTERISK-16570 internally here.
02:06.23sruffella planning issue was created…but it's still waiting to be scheduled since it's not affecting too many people and there are other show stoppres.
02:06.25sruffellstoppers
02:09.27WIMPyNot sure. It doesn't seem that unlikely.
02:10.32WIMPyIf you start a box and lines have been down for a while they might not come up immediately, and who knows in what order.
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02:29.00tzafrirWIMPy, yes
02:29.51WIMPytzafrir: Any idea why I see only one of the four PRIs on an Astibank?
02:30.13WIMPyOr is there a single port version that has four ports anyway?
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02:30.34tzafrirThat's set through the license
02:31.13WIMPyAh, so it's probably a one port mudule? But no way to find out except by software?
02:35.20WIMPytzafrir: Or is the license stored on the PC?
02:35.49tzafrirno. On the Astribank
02:36.04WIMPyOk. Thanks.
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03:43.21*** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net)
03:44.49x1userHi, all. I want to migrate from 1.4 to 10 latest. Before I do that. What might go wrong... I mean anyone who had messed with this before. And how much different is 10, because I only have experinece with 1.6?
03:49.06sruffellx1user: Why are you updating?  If you don't have a specific need, I would recommend trying out 11-beta since it will be the next long term release.
03:50.26sruffellBut in general, I would recommend starting by reading the various UPGRADE.txt files: http://svnview.digium.com/svn/asterisk/branches/11/UPGRADE.txt?view=markup
03:52.27x1userWe have working 1.4.2 but we want to provide more services to the clients so 10 would offer "all" the feautres. This is why i want to update.
03:53.13sruffell10 is not a long term support branch.  11 will be though.
03:56.11x1userthanks, for that link. It is very usefull, i ma just wondreing if i can make it straight to 11 from 1.4.2 without much pain?
03:56.58sruffelldepends on what parts of Asterisk you're using and how you're using them. I don't think anyone will be able to say directly.
03:58.00x1userIt is very basic seup with just sip trunks and connectic with few other asterisk via iax.
04:01.13sruffellIf really basic, then I doubt you'll have too many problems….but don't hold me to that since I don't know what "basic" means to you.
04:02.22sruffellBut, I still think you should just join the 11 beta.
04:03.06sruffellyour customers might start asking for http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ support in the near future and you'll have that covered. ;)
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04:34.00*** join/#asterisk ledoktre (~chatzilla@omni-01-161.cable.netins.net)
04:34.58ledoktreanyone here have any experience with making parkinglot and asterisk 1.8.x work together?  I've been trying for 2 days to figure it out, and I can't find anything info to steer me straight :-(
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04:45.07ChannelZyeah
04:45.31ChannelZconfig in features.conf
04:46.34ChannelZThen in your extensions.conf, in whatever context you want the parking lot to be accessable from, do    include => parkedcalls   where parkedcalls is the context configured in features.conf
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04:46.46ChannelZwhoops
04:50.49ledoktreChannelZ,  whats weird is that I have tried using the default parkinglot and also creating another.  Default has a context in it of 'parkedcalls'.  I include this in my context of my internal lan (as I am just testing it for now by dialling between phones).  It complains on reload about can't find the context.
04:51.23ledoktreAlso complains when I include the context for the custom parkinglot too.    I am guessing since it won't include, that explains why it won't let me park a call
04:52.42ledoktrewhen I use 'features show', it lists out stuff, but never any custom parking lot.  Just  Call parking (Parking lot: default), etc settings, enabled : yes
04:53.57ledoktreMore specifically when reloading on CLI ->   pbx.c:10512 ast_context_verify_includes: Context 'internal' tries to include nonexistent context 'parkedcalls'
04:56.42ChannelZwell that's your clue right there
04:57.19ChannelZWhat do you get if you do    module load app_parkandannounce
04:59.26youjellyHi guys
04:59.55ChannelZhey lady
05:00.00ledoktreChannelZ : WARNING[1397]: loader.c:846 load_resource: Module 'app_parkandannounce' already exists.
05:00.26ledoktreChannelZ : Also, I just did a 'core reload'.  It showed a lot of stuff, but I searched, it did NOT parse features.conf.      ???
05:00.37ledoktreDid not even list it in the reload
05:00.38ChannelZyeah it won't on a reload
05:00.55youjellylol ChannelZ, if by lady you meant laddy then OK :)
05:01.12youjellyor simply lad
05:01.15youjelly:)
05:01.20ledoktreChallenZ : ok, how can I see the output from a restart then?
05:01.29ChannelZpastebin your  features show
05:01.47Corydon76-homeIt won't read files on a reload if the file has not changed
05:01.52ChannelZyou can run it non-detached from the shell, like just 'asterisk -cvvv'
05:02.07Corydon76-homeIt cached the modification time of each file
05:02.40youjellycan you do a dialplan reload and see you're not getting any warnings
05:03.53ledoktreChannelZ : http://pastebin.com/dZhzd2Y6   Am trying out the restart.
05:04.34ChannelZwell it looks fine.  Do a 'dialplan show' and see if the context 'parkedcalls' is listed
05:04.53ChannelZ(assuming that's what you still have include => parkedcalls in your extensions)
05:05.15youjelly:)
05:05.30ChannelZand you should see the include also listed in the dialplan show under the appropriate context
05:05.49ledoktreChannelZ :  This should be a clue I think.  Restart yielded lots of stuff, but in particular ---   [Sep  7 23:50:39] WARNING[1304] features.c: Could not load features.conf
05:05.56ChannelZIf that checks out, then you need to define what "not working" means
05:06.24ledoktredialplan show does not show parkedcalls
05:06.29ChannelZodd that the lot seemed to be configured anyway (defaults maybe? I dunno)
05:06.51ChannelZwell check your syntax, file permissions of features.conf...
05:07.10ledoktreand by not working, i mean no matter what context i try to include, it complains about trying to include a nonexistant context.  and then when trying to park calls, invalid extension
05:07.17ChannelZthat it's not still called .sample
05:10.36ledoktreChannelZ :  I think I found it.   I have not tested yet, but you got me on the right track I think.     The file was installed from a deb.  Somehow / reason, features.conf was owned by root.   Chowning it to asterisk, and it shows everything with features and dialplan as above
05:10.39ledoktrejust need to test it out
05:13.01ledoktreChannelZ : Appears to be working -- thanks for that help.   Can't believe I didn't find/see that ownership issue.
05:13.14ChannelZIt's always the last place you look
05:14.10ChannelZnow have fun putting people on hold!
05:14.22ledoktreHeheh, will do !  One other thing?
05:14.50ledoktreAny ideas on any websites / etc that talk about pimping out Polycom phones specifically on * 1.8?
05:15.22ledoktreI've got park (heheh!) working now, and intercom, transferring calls, but got a few things to try and sort out like BLF, etc.
05:15.27ledoktresoft keys, etc
05:15.38ledoktreMaybe visual voicemail?
05:17.20ChannelZI'm not a polycomer sorry. Someone else here maybe will chime in
05:18.25ledoktreI understand.  Thanks anyway!
05:25.51learathanyone played with the cisco 9951?
05:25.59learathI've got one mostly working, but nothing fancy
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05:44.17Sprockshow do I set up asterisk so that I can make calls using google voice?
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05:52.06Sprocksdoes anyone know how I would set that up?
05:52.56ChannelZhttps://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
05:53.28ChannelZoh.. looks like they moved some stuff around.  You probably want to hit the 'old page' link at the top unless you're playing with the Asterisk 11 beta
05:54.30Sprocksi was just using the 1.8.10 that is available in ubuntu's apt-get
05:55.20ChannelZin that case you'll probably need to look for the gtalk and jabber modules if they split those up separately
05:56.12Sprocksso go through the old page?
05:56.24ChannelZyah
05:57.15Sprocksare the gtalk and jabber modules easy to find?
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06:00.52ChannelZwell presumably in your repository
06:01.01ChannelZor they might have already been included in the base package, I don't know
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06:01.27ChannelZyou can do a 'module show like gtalk' in the console and see if it's there
06:02.42davemha_Update to my DTMF tone problem: added "options wctxx4p vpmdtmfsupport=0" to /etc/modprobe.d/dahdi.conf and rebooted
06:03.08davemha_Problem solved - DTMF tones from the PRI can now be processed by Asterisk.
06:03.33davemha_Of course, it might have been the reboot, not the module option.  :)
06:04.23ChannelZstrange
06:04.57Sprockshmm I don't see anything in /etc/asterisk maybe it put things somewhere else
06:05.35ChannelZthis is why I hate packages sometimes
06:05.58Sprocksya I'm gonna source install
06:06.05Sprocksmuch easier and betetr
06:08.50ChannelZIn that case you'll also need libiksemel-dev
06:09.19ChannelZand libssl-dev
06:11.00Sprocksyep I have those
06:13.15Sprockswhats the diff between certified, standard w/ digium, LTS and standard?
06:15.16ChannelZcertified is if you have digium hardware phones, skip it (well unless you do..)  LTS is 1.8 and is LTS much like Ubuntu in that they plan to bug support it for longer than the 'standard' release
06:15.30ChannelZ(which is v10)
06:17.51Sprocksis there anything bad if I just do the cert over LTS?
06:19.15ChannelZthere's no reason to.  If you're building a new system from scratch, just get 10.7.1 unless you're doing this for some business or something who wants to not upgrade in a timely fashion and wants LTS
06:20.10ChannelZcertified is the same as regular except they disable a bunch of modules you'll probably wind up re-enabling.  Unless you have Digium phones there's no real reason to use it.
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06:22.33ChannelZbrb my graphics card is acting whack, rebooting..
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06:56.19kannanhello all. how can we do a "press 0 at any time to transfer to <somenumber>" in an IVR. only when we have a read , or waitexten or background, I can see a way to do a transfer. But a a Didit that can interrupt the whole ivr at anytime?
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08:06.22kaldemarkannan: see features.conf
08:07.15kaldemarthen again, if your IVR is all Background and goto's, you won't need anything special.
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08:28.34Smirkerhowdy
08:30.32Smirkerre: SIP CALLID. how would i retrieve the callid of an outgoing sip session initiated by Dial()?
08:31.20kaldemarSmirker: when and from where?
08:31.30Smirkeri try to access ${SIPCALLID} from the h context after I dial, but as expected it is returning the id from the SIP session that the dialplan was initiated from - not the actual session created from Dial()
08:33.17kaldemarh extension you mean?
08:33.27Smirkerthat's what i mean
08:33.41Smirkercall comes in from registered client (SIPCALLID 1) --> my context (which initiates the Dial command) --> another call initiated (SIPCALLID 2). i want to find SIPCALLID 2
08:34.01kaldemaryour channel at that point is the originating channel, so you can't access variables of a channel that doesn't even exist anymore.
08:36.20kaldemaryou could maybe use option U() for Dial to store the id somewhere to be available for the originating channel.
08:38.30kaldemaryou could feed the originating channel (${CHANNEL(name)}) as an argument and then set the var in the channel's shared namespace with Set(SHARED(callid,${ARG1})=${SIPCALLID}) in a gosub/macro that is executed on the called channel due to U() or M().
08:39.53Smirkerthat looks like a nice solution. let me test that out.
08:39.58kaldemarbut what's really the point of all this trickery? maybe there's a simpler solution for what you are aiming at?
08:40.12Sprockswow finally it installed correctly, ChannelZ how do I load/get gtalk module if its not there?
08:42.42Smirkerkaldemar: it is for a voip service. we send the call to a wholesaler, who gives us CDRs with the callid for matching purposes. i want to match those calls (which have billing information) to individual clients.
08:43.09Smirkerkaldemar: typically, i match the records based on callerid. but in some scenarios this doesn't work.
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08:47.14ChannelZSprocks: it should have been build if the prereqs were met
08:47.32ChannelZIn the source do a 'make menuconfig' and see if they are marked
08:47.41ChannelZchan_gtalk and res_jabber
08:48.54Smirkerkaldemar: just testing the U() solution now.
08:48.56kaldemarSmirker: you might not even need to use the shared variable space if you have func MASTER_CHANNEL. that allows to set channel variables on the master channel directly.
08:49.31SprocksChannelZ, i had the .conf files for them
08:49.48kaldemarSmirker: that would drop the need of feeding the channel as an argument and shrink the gosub/macro to just Set(MASTER_CHANNEL(var)=${SIPCALLID})
08:50.25Smirkerkaldemar: -erases some code-
08:51.11ChannelZSprocks: I don't really know what your question is
08:52.19Sprocksyou said earlier use "module show like gtalk" from console, is that asterisk console? I did that and it said 0 modules loaded
08:52.57ChannelZYes
08:53.20ChannelZso it probably didn't build then.. re:  make menuconfig and see if it's selected and if not, why
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08:54.14Smirkerkaldemar: using U(), a subroutine and MASTER_CHANNEL works perfectly! thanks heaps!
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08:56.03Sprocksdoes the XXX mean selected?
08:57.13kaldemarit means dependencies are not met.
08:58.29ChannelZdo you reeeealy have libiksemel-dev installed?
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08:59.49Sprocksill double check but I swear I installed it
09:00.10ChannelZ-dev, not just the shared lib
09:02.11gavimobilefolks, when I try to call myself I get dead air. I am trying to make an inbound route. here is my output with sip debug http://pastebin.com/7P1RkhFD
09:03.12gavimobilemy trunk is registered.. and I don't see any problem with dialplan
09:04.48Sprocksi guess I have to run configure again
09:05.32ChannelZyes
09:05.42ChannelZgavimobile: firewall issues?
09:08.33gavimobileChannelZ: is that what it looks like from the log?
09:09.01gavimobilewhen I call another number other than myself it works
09:09.55gavimobilealso, everything was working on a previous server with freepbx, I just changed the forward ip's in my router to the new server
09:10.56ChannelZI'm having a hard time telling what's going on in the log.. 3 different IPs.  but they're all external so I assume they're correct
09:11.26ChannelZIs sip.conf configured to know your correct internal and external IPs?
09:11.53ChannelZAnd do you get audio in either direction?
09:12.13gavimobileexternip=81.218.xxx.xx <--the ip of my server
09:12.14gavimobilelocalnet=192.168.0.0/255.255.255.0 <----my router is 192.168.0.1
09:12.55gavimobileI don't have audio problems when caling my cellphone for example. I can't tell if there is audio in the dead air. the dialplan is supose to answer and play a test wav file
09:14.15gavimobilethe weirdest thing is when I called, my old pbx sent me an email whith a voicemail...
09:14.57gavimobilemeaning I forwarded all ports to server b which were all set previously on server a. I called server b and server a sends me a voicemail
09:15.21ChannelZyou've got something screwed up
09:15.31gavimobileChannelZ: well thank you :-p
09:15.45ChannelZDNS or I don't know what you're talking about server A and B
09:16.10kaldemargavimobile: first mistake is forwarding. if the old box still registers to your provider first, it probably gets the calls.
09:16.34kaldemargavimobile: you're using freepbx, right?
09:16.46gavimobilekaldemar: not on me new server
09:16.51gavimobilemy old server is freepbx
09:17.17gavimobileI didn't get a voicemail since yesterday with all the many tests, so it may be inaccurate with that diagnosis
09:17.36kaldemarthen you can drop the "inbound route" etc. terminology and start speaking asterisk.
09:17.52kaldemaralso, look at CLI when you make a call, you'll know if the call even comes to your box.
09:18.06gavimobilekaldemar: :-p im triyng to make my dialplan pickup calls.. is that more proper?
09:18.23gavimobilekaldemar: the cli with or without sip debug?
09:18.34gavimobileI've checked the cli using both
09:19.18ChannelZit looks like you're dialing from 79.* (an iPhone running some SIP client?) and calling an external number yes?
09:19.39gavimobileChannelZ: yes, im using my iphone with a softphone
09:19.44gavimobileremotly
09:20.02gavimobilethe external number is myself. I just don't know how to intercept the call
09:20.05ChannelZso test half of it first.  Have you made a test extension that plays a sound and then does Echo and called that?
09:20.19gavimobilethe definitive guide shows such a simple example, I don't know why I cannot get it to work
09:20.31gavimobileChannelZ: yes
09:20.40ChannelZand it does?
09:20.45gavimobileI can call an ext than does the same work
09:20.47gavimobileyes
09:20.51gavimobileaudio no problem
09:21.02gavimobilealso I can call other extentions and hear both ways
09:21.21gavimobileI've also make an extention record which works but haven't tested that in a while
09:21.21ChannelZAnd do you have inbound calls from your ITSP working similarly?
09:21.50gavimobileChannelZ: I didn't understand that question
09:21.55kaldemargavimobile: pickup means a specific thing. what you need to configure is a matching peer for the provider, a context that the calls land in and extensions that match dialed numbers.
09:22.12gavimobilekaldemar: I made a context
09:22.26gavimobilecalled myInbound in my sip.conf for my trunks
09:22.54gavimobilemy dialplan has the matching context [myInbound] which answers and plays a test sound
09:23.10gavimobilethe log without sip debug shows me making the call but after that it does nothing
09:23.30kaldemara context does not match, extensions do.
09:23.32gavimobileI think it even says that the server answered the call
09:23.46gavimobilekaldemar: the context need to match
09:24.00ChannelZthe peer in sip.conf is set to a context in extensions.conf
09:24.02gavimobileas explained in the examples
09:24.14gavimobileChannelZ: that is correct
09:24.20ChannelZthe name of the peer has nothing to do with the dialplan context however
09:24.49gavimobilesip.conf [my-trunk] context=myInbound   extentions.conf [myInbound] exten => ...
09:25.02ChannelZIn any event if you get no audio calling in through your ITSP then that's the half to start looking at
09:25.47gavimobileChannelZ: wouldn't the best thing be to get a better understanding of the debug log?
09:26.03gavimobilecause we are all making assumptions
09:26.30ChannelZThe output you showed was of two calls (and the paste expired so I don't have it anymore)
09:26.48gavimobileChannelZ: should only have been one call
09:26.53ChannelZiPhone via SIP <-> Asterisk  and  Asterisk <-> ITSP
09:26.54gavimobileI can repost
09:27.21ChannelZYou say the iPhone <-> Asterisk half works, but don't know if the Asterisk <-> ITSP part works
09:27.44ChannelZso leave the iphone out of the picture and figure out what is going on between you and your ITSP
09:28.04gavimobilehttp://pastebin.com/N7FBKY43
09:28.06ChannelZlet's see THAT half of it.
09:28.18gavimobileChannelZ: I can call from another softphone
09:28.42gavimobileeven a softphone on the same gateway as my server
09:29.32ChannelZI'm saying get your inbound working.  Call your DID with your cell phone (as a phone, not SIP through Asterisk) and see if you have one or two way audio
09:30.16kannankaldemar , thanks. the features.conf should work
09:30.26ChannelZor you can make a junk extension that plays a sound and does echo and then do an originate on the console to let Asterisk alone dial out through your ITSP to your cell phone or something
09:31.50gavimobilethis is what [tk-defender] tolk me yesterday. didn't understand it maybe you guys can http://pastebin.com/zXG2nM7J
09:32.54gavimobileChannelZ: I don't know how to build that from the top of my head
09:33.06gavimobileand I cannot call from my cellphone
09:33.34gavimobileI don't have a phone to call from
09:33.52gavimobileat this point in time
09:34.46ChannelZhttp://pastebin.com/zryxzWmP
09:35.06ChannelZas in you don't have cell service?
09:35.29gavimobileright, it's a pay as you go plan
09:35.32gavimobiledon't ask
09:35.47gavimobileanyways I, adding the echo application but I didn't understand the bottom
09:35.52ChannelZwell that's going to make it harder to test
09:36.06gavimobileI can use 1 call with jabber
09:36.10gavimobileor whatever its called
09:36.23ChannelZnow you're just adding more variables
09:37.41ChannelZfrom your original paste about all I can say is you and your ITSP seem to be communicating fine
09:38.00gavimobileI can confirm that as well
09:38.01ChannelZAssuming your IP really is 81.218.196.30
09:38.17ChannelZWhy you have no audio is more than likely a networking issue
09:38.28gavimobileok echo test works fine
09:38.41ChannelZwhat does that mean.. echo test of what
09:38.47gavimobileChannelZ: that's the ip of my server
09:39.00gavimobileChannelZ: I dialed 666 and heard myself
09:39.12ChannelZfrom what
09:39.20gavimobilefrom my softphone
09:39.24ChannelZyeah.. you already said that half worked a long time ago
09:39.40gavimobileso what was the point of building this application
09:40.03ChannelZto see if your iphone was working between asterisk
09:40.35ChannelZIs Asterisk behind a firewall (or is the same machine as)?
09:40.59gavimobileyes, behind the router firewall
09:41.26gavimobilehrm...
09:41.33gavimobilemaybe I have selinux installed
09:41.34ChannelZand are you port-forwarding on it back to your Asterisk machine some range of ports?
09:41.42gavimobiledidn't think of that
09:41.43ChannelZ(as seen in rtp.conf)
09:42.18gavimobileChannelZ: my router is forwarding all necessary ports to my NEW pbx
09:42.21ChannelZwell that can muck things up as well
09:43.09gavimobilemy router is forwarding properly
09:43.20ChannelZI meant selinux
09:43.29gavimobiledo you know off the top of your head how to check if selinux is enforcing?
09:43.33ChannelZthough your iphone was external as well
09:43.39gavimobilesysetmlevel
09:44.26gavimobilesystem-securitylevel
09:44.28ChannelZNo idea, I do not dabble in the selinux
09:46.18*** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com)
09:46.23ChannelZcould just shut it off for testing purposes to see if that's it
09:46.31gavimobileI don't think selinux is installed
09:46.38gavimobilethere is no iptables service
09:47.06gavimobileand the connand for selinux isn't working system-config-securitylevel
09:47.14gavimobileso apparently I didn't install it on this server
09:47.23gavimobileI use it sometimes on web servers
09:47.34gavimobileso I was confused for a sec... but its not here
09:48.19ChannelZso this 'old server' of yours, is it still running?
09:48.35ChannelZIs your router perhaps still sending traffic from it's NAT table to the wrong place?
09:49.17gavimobilethis time I called and I got Got SIP response 503 "Service Unavailable" back from 178.63.143.236:5060
09:49.33gavimobileChannelZ: I can reboot my router
09:49.46gavimobileChannelZ: old server is still running, yes
09:50.08ChannelZas kaldemar suggested, is it still registering to your ITSP?
09:50.22gavimobileI wouldn't know how to test
09:50.37ChannelZshut it off
09:50.46ChannelZor at minimum kill asterisk...
09:50.53gavimobileamportal stop?
09:50.55ChannelZIs your new one registering?
09:51.15gavimobileyup.. done
09:51.31gavimobileChannelZ: is my new server registering with whom? my trunks
09:51.32gavimobile?
09:51.38ChannelZyour itsp
09:51.41*** join/#asterisk mathi (~Matthew@109.133.242.136)
09:51.42gavimobileyes
09:51.43mathihi
09:51.44ChannelZIE you have a register => line in your sip.conf
09:52.14gavimobileChannelZ: I do have register => for the trunk who makes the calls. the trunk for receiving calls doesn't require me to register
09:52.27ChannelZof course not
09:52.28*** join/#asterisk timahvo1 (~rogue@196.200.32.36)
09:52.30mathihow can I do something like: if ${lang} is 1, set lang to "fr", if ${lang} is 2, set lang to "nl", etcetera for all the languages, ... ?
09:52.48gavimobileChannelZ: ?
09:53.07gavimobileof course not what
09:53.40ChannelZforget it for now
09:54.07mathiwhat application should I use for this ?
09:54.10ChannelZso now that your old server is hopefully out of the picture, restart (asterisk) your new one and make sure it registers properly to your ITSP
09:55.04gavimobilesip show peers show it as ok though
09:55.13gavimobileill reboot just in case
09:55.24ChannelZyou can pull the plug on your router too for good measure
09:55.40gavimobileChannelZ: im not at my office
09:55.54gavimobileand I don't want to shut it off perminently
09:56.03gavimobilebut I can assure you asterisk is stopped on the old one
09:56.07gavimobileand I just rebooted the new one
09:56.09ChannelZwell then just wait a few minutes to make sure it's NAT table expires
09:56.39gavimobileI waited all of last night! I changed the forward ip yesterday
09:56.40ChannelZthough hopefully it'll be smart enough to figure it out
09:56.57gavimobileI think we should reboot the router for safe measures
09:57.00*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
09:57.11ChannelZI thought you just said you couldn't
09:57.34gavimobileI said I didn't want to shutoff my old server which asterisk is currently stopped
09:57.45gavimobilethe rotuer I can reboot
09:57.56ChannelZ...nevermind.. yes do that
09:58.22gavimobileChannelZ: ok I will be disconnected for a few minutes because im connected remotly to irc on one of my desktops
09:58.23gavimobilebrb
09:59.57ChannelZmathi: you could probably use ExecIf
10:00.15mathithank you i'll check
10:00.48mathiyep looks nice thx :)
10:03.11ChannelZExecIf($[${lang}=1]?Set(LANGUAGE=fr))  and whatnot
10:03.17mathiChannelZ, ExecIf($[${lang} = 1]?Set(lang="fr"))
10:03.26mathiChannelZ, I thought that string needs quotes ?
10:03.51ChannelZwell it's complicated
10:04.36mathii missed that in the docs:/
10:04.43*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
10:04.49gavimobileChannelZ: and were back
10:05.07*** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net)
10:05.13ChannelZThe parser tries to guess what you meant sometimes, quotes don't always help you out.
10:05.44gavimobilemy trunk -  netvision                  82.166.66.43                                 N             5060     OK (26 ms)
10:05.49ChannelZin this case if you did Set(lang="fr") you'll literally get  "fr"
10:06.00mathiomg :)
10:06.05ChannelZgavimobile: ok so that's good
10:06.11gavimobileit was like that before
10:06.46ChannelZThat's fine, but if your old system was re-registering you'd never see that on the new.  They'd basically have been fighting each other and your ITSP could be very confused.
10:07.26gavimobileChannelZ: will this work? exten => start,1,Playback(echo-test)
10:07.27gavimobile<PROTECTED>
10:07.32ChannelZSo you said earlier you have no actual phone service at the moment... so what was the test call you were trying to make earlier to?
10:07.35mathiChannelZ, then, you have used a global variable (as you wrote LANGUAGE all in upper case), but I think it makes more sense to make it a channel variable? As I clearly take input of user, and set his language, thus for that particular call
10:08.04gavimobileChannelZ: I don't have cellular server, but I do have working trunks
10:08.05ChannelZmathi: I don't really know what you're doing.  I thought you were trying to change the channel's language, as in what sounds it plays, etc.
10:08.16gavimobilecellular service*
10:08.44gavimobilefrom my pbx I called my cellphone (I have free inbound on my cell) and that works
10:08.59mathiChannelZ, it'zs an IVR and I first ask user's language, I wouldn't use a global variable here, I think that it will be shared among all calls otherwise
10:09.00*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
10:09.01gavimobilecalling other extentions work
10:10.20ChannelZmathi: but what I mean is, does this language variable you're trying to turn from a number into text mean something to YOU elsewhere (a script, or other dialplan, etc.) or are you trying to set the language Asterisk uses so that it automatically plays sounds out of the right prefixed sound folder?
10:11.14mathiChannelZ, indeed:) I have folders with sound, e.g.  fr/welcome, nl/welcome, ...
10:11.49mathithis language variable is to choose the right folder as you guessed
10:12.34ChannelZok... well I thought it was LANGUAGE but that's wrong anyway.. I'm trying to remember how you change the channel's language
10:13.22mathiChannelZ, what does it mean, "change the channel's language" ?
10:13.38gavimobilewith channel
10:13.45ChannelZuhm.. just what it says
10:14.13ChannelZif you do Playback(hello) Asterisk picks the hello file from en/hello or fr/hello or whatever, depending on what language the current channel is set to (whatever is doing Playback)
10:14.20mathiahhhh
10:14.28mathiso I am coding things that have already been done for me
10:14.49ChannelZsort of
10:14.50gavimobilesame => n,Set(CHANNEL(language)=en)
10:15.01ChannelZif you're wanting to give an IVR so people can choose, you still have to hook that part up yourself
10:15.06ChannelZthanks gavimobile yes
10:15.35gavimobilewasn't for you though :-p
10:15.36ChannelZso it'd be more like    ExecIf($[${lang}=1]?Set(CHANNEL(language)=fr))  and whatnot
10:16.01ChannelZassuming ${lang} was set to something interesting.
10:16.13mathiah yes I still need my ExecIf cause I have numbers as input, the only advantage  i see is instead of Playback(${lang}/welcome), I can skip the ${lang} part
10:16.29ChannelZthough depending on how your IVR is setup, '1' could just be an extension that calls the Set(...) directly, but I don't know how you have things programmed.
10:17.14ChannelZYes, that's the idea -- rather than having to hack every Playback, Asterisk can handle it for you.  It also will make things like voicemail prompts or other sounds that other applications play that you don't necessarily have direct control of in the dialplan
10:17.36ChannelZgavimobile: so have you made another test call and gotten any different results?
10:17.39mathiChannelZ, well that's a good question, when is it a good idea to start an extension. I could just have some ExecIf, or I could ad extensions as well
10:18.40gavimobileChannelZ: same thing but its still different from yesterday which is a good thing
10:19.02ChannelZmathi: it's more of a design question.  Were you using Read() or something to let people select a language?
10:19.19ChannelZgavimobile: what does that mean? same but different...
10:19.32mathiChannelZ, same => n,Read(lang,select-language,1)
10:20.00gavimobileI didn't say same but differ.. yesterday it was dead air. today I get a reply from the telco
10:20.05ChannelZthat's fine
10:20.38ChannelZso then just do the ExecIfs as posted above and youre done
10:20.52gavimobileChannelZ: an error message from the telco saying cannot connect my to the number
10:21.20gavimobileto the "requested" number ... so this should mean a problem maybe with my dialplan I hope
10:21.21ChannelZgavimobile: not sure how that's better but OK
10:21.30ChannelZyou were getting a reply before, it was placing the call...
10:21.43gavimobileChannelZ: I was not getting any reply before
10:21.48ChannelZat least I think, I closed that window again
10:22.05gavimobilelet me post the latest debug
10:22.41ChannelZoops no you're right, the SIP was going nowhere.
10:23.13ChannelZalthough it was initially replying with the UNAUTHORIZED
10:24.04gavimobilehttp://pastebin.com/pUx48w4f
10:24.27ChannelZwell no, it actually did work afterwards: -- SIP/didlogic-trunk-0000003c answered SIP/0000FFFF0001-0000003b
10:24.47gavimobileso is my dialplan screwed up?
10:25.10ChannelZmaybe
10:27.18gavimobileChannelZ: what can you tell me about this
10:27.34gavimobilehttp://pastebin.com/Y8ruh6yh   my trunk doesn't require register and im sure that's the correct trunk details
10:27.46ChannelZI'm still trying to sort through it, these logs are a mess with all the double spacing and stuff
10:28.16gavimobileChannelZ: yes, putty screws them up. I don't know how to save the output directly on the server
10:28.17ChannelZSomething is still screwy with your network with these retransmissions, but other than that, it's saying 'not found' which is probably that it doesn't like the number you are dialing
10:28.28ChannelZ- Called SIP/didlogic-trunk/972773354673@sip.didlogic.net
10:28.33gavimobileChannelZ: omg
10:28.36gavimobileI think I know what it is
10:28.41ChannelZThat number format is maybe wrong to them
10:28.56gavimobilehrmmmm.
10:28.58ChannelZcountry codes and all that I assume, it don't look like no US phone number
10:29.19gavimobiledidlogic is making the call, right?
10:29.22ChannelZbut for instance my ITSP makes me dial 1 before the area code/number, even on local calls
10:29.24*** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke)
10:29.40ChannelZWell, you're asking them to make a call.
10:30.04ChannelZIf you tell them to call 3948 and they have no idea what that means....
10:30.07gavimobileChannelZ: yes I am, so why is the error message coming back in the language from my country
10:30.28ChannelZX-Asterisk-HangupCause: Unallocated (unassigned) number
10:30.39gavimobilethe number im calling is also sip trunk purchased locally
10:30.42ChannelZhuh?
10:31.16gavimobilenevermind.. I don't know what im talking about
10:31.38*** join/#asterisk PbxMan (~user1@213.37.35.26.dyn.user.ono.com)
10:31.40PbxManhello
10:32.25ChannelZactually wait this is all mixed up
10:32.38ChannelZdo this, this is too confusing
10:32.40ChannelZsip set debuf off
10:32.47ChannelZs/debuf/debug/
10:32.50PbxManis anybody familiar with the Java AGI libraries for asterisk?  Is net.sf.asterisk.fastagi obsolete?
10:33.30ChannelZsip set debug peer didlogic-trunk
10:33.51gavimobiles/debuf/debug/ ?
10:33.53ChannelZthen make the test call again so I can just see the traffic between you and didlogic
10:34.02gavimobilefrom cli?
10:34.04gavimobiledoesn't work
10:34.16ChannelZcan you be more vague?
10:34.36gavimobileno such command
10:34.37ChannelZon the * console: sip set debug off
10:34.46gavimobilesip debug I disabled already
10:34.51ChannelZthen: sip set debug peer didlogic-trunk
10:36.24gavimobileChannelZ: this is all I could capture http://pastebin.com/rLYV9YYE
10:37.35ChannelZwell this is even stranger. The call made progress this time but then you sent a CANCEL
10:37.56ChannelZbut again something is still messed up that packets are randomly disappearing it seems with these retransmissions.  I don't know what the heck is going on
10:37.57gavimobileI hung up after it told me that my call cannot be completed as dialed
10:38.13ChannelZok so you are successfully making a call through your ITSP.  You're just not dialing a valid number.
10:38.15ChannelZThat audio came from THEM
10:38.30gavimobileI don't think that's correct
10:38.43gavimobileand lets test by trying to dial someone who  has a 077 number like me
10:39.58gavimobileI can dial another 077 number with no problem
10:40.15gavimobilejust dialed this number 077-5254912
10:40.24ChannelZso what exactly is the problem here
10:40.37gavimobileI cannot call my own 077 number
10:40.48ChannelZyou don't appear to be trying
10:41.05gavimobilehow is that?
10:41.06ChannelZ- Called SIP/didlogic-trunk/972773354673@sip.didlogic.net
10:41.16ChannelZwhat is 972773354673 ?
10:41.22ChannelZdoesn't look like 077 anything to me
10:41.24gavimobile972 is the country code
10:41.38gavimobileand we need to remove the leading zero if called from outside the us
10:41.42gavimobiledidlogic is from us
10:41.48gavimobileso they need to dial 972 77 etc
10:42.18ChannelZyou need to figure that out for sure then because your getting a response which is the number you're dialing is bogus
10:42.20gavimobilefor my locally I need to dial 077. so my dialplan actually dials 972 77 which I dial 077
10:42.40gavimobileim forsure
10:42.51gavimobileI just dialed a number identical to mine with the same trunk
10:42.56gavimobileI can show you the output
10:43.16ChannelZthen you need to take it up with them why you can't dial that one specific number
10:43.29gavimobilemy trunk provider doesn't need to know the local ip for the server?
10:43.33gavimobileno right?
10:43.41gavimobilethey just need my external ip address
10:43.48ChannelZyeah
10:43.48mathiChannelZ, once user has selected his language, i try:  same => n,Background(select-action)
10:43.54mathisame => n,WaitExten(4)
10:44.04mathibut nothing is said, strange
10:44.22gavimobileChannelZ: the probl is there is no support for my local trunk provider.. they are as helpless as talking to a stone wall
10:44.40gavimobileis my dialplan correct?
10:44.49ChannelZmathi: turn on some verbose  (core set verbose 3) and then make a whole call through your IVR selecting the language, etc. and pastebin the output
10:45.03ChannelZgavimobile: I have no idea, I've never seen it.
10:45.10mathiI know only asterisk -rv command
10:45.22mathihow can I set the verbose mode to 3 ?
10:45.28gavimobileChannelZ: I posted it earlier... ill go ahead and post it again
10:45.39mathiahhh
10:45.40mathisorry
10:45.44mathiyou wrote it :)
10:45.45ChannelZYou are sending an receiving SIP, and apparently audio when you dial a number that actually works.  Your number that doesn't work is not because of a problem I can necessarily diagnose.
10:46.20ChannelZmathi: same as -rvvv (three v's is verbose 3)
10:46.38ChannelZbut either way
10:46.41mathivery useful thx! I'll try and come back
10:48.43ChannelZI gotta go to bed soon, it's almost 5am
10:48.52gavimobilewow
10:48.55gavimobileso go to bed
10:48.57gavimobilelol
10:49.30gavimobilemaybe just before you go, take a look at this and tell me if you see something wron http://pastebin.com/nRn8W1Rd
10:49.50ChannelZI was going to try to not leave mathi hanging
10:50.15gavimobileoh, I thought he was taken care of
10:50.24gavimobilefor me its almost 2 pm :-p
10:50.27ChannelZno, I need to see his console output
10:50.36mathiChannelZ, http://pastebin.com/SKDdVD5m
10:50.52ChannelZAs for you, that dialplan does basically nothing
10:51.05gavimobileChannelZ: it should answer and play a tune
10:51.13ChannelZor rather it's totally unrelated to these external calls you're trying to make
10:51.22gavimobileChannelZ: well maybe this is my problem
10:52.06gavimobileI thought that the value of the context in the trunk is like a goto in the dialplan
10:52.31gavimobilewhat would be a valid catch all incomming phone calls dialplan
10:53.21ChannelZthis dialplan did not come from whatever it is you have been pasting the sip debugs from
10:53.35ChannelZI mean unless there is a ton more in it that you cut off
10:53.46ChannelZor you're also using AEL or something
10:54.08gavimobileI can send you whatever you request
10:54.10ChannelZmathi: looking, sec...
10:54.27mathiChannelZ, I didn't know it was 5 am for you, I can try by my own it was already very kind to help me
10:54.37ChannelZgavimobile: pastebin "dialplan show"
10:54.38gavimobilebut you said I am able to send and receive calls which your right. so this is probably a dialplan issue.
10:55.45gavimobilehttp://pastebin.com/wWVad7Ve
10:56.32gavimobileI have a good feeling about this
10:57.31ChannelZoh you have a bunch of includes, that's why
10:57.56gavimobilemy includes are working to my knowledge
11:00.26*** join/#asterisk jaxon007_ (~jay@123.252.144.92)
11:00.33gavimobileshouldn't there be like one line of code to intercent all calls send to my pbx?
11:00.39ChannelZmathi: so I'm not really sure what you're doing -- it looks like your ExecIfs are probably working.. but then you do a Background(fr) ??
11:00.59mathiChannelZ, same => n,Background(select-action)
11:01.00gavimobileintercept*
11:01.32ChannelZthen you might have a typeo
11:01.54ChannelZpastebin your extensions.conf
11:02.02mathiI have this file:  select-action.wav
11:02.08mathiin fr directory (and I choosed fr)
11:02.11mathio
11:02.14mathiok
11:02.14gavimobileChannelZ: you want my extentions?
11:02.51ChannelZgavimobile: yes but what is it I'm supposed to be figuring out again?
11:04.04gavimobileChannelZ: how to get some sleep
11:04.05gavimobilelol
11:04.12gavimobileI just want to receive calls
11:04.14gavimobilethat's all
11:04.25gavimobilereceive the call and play a sound
11:04.28gavimobileanything
11:05.00mathiChannelZ, I think I know the problem, I have put a directory with my site name, in /sounds, e.g. /sounds/my-site/fr/select-action...    so I cannot benefit of the CHANNEL(language) variable :-(
11:05.01ChannelZso is that what you were trying to show me?  is 'netvision' a totally different provider your DID is at?
11:05.25gavimobilehttp://pastebin.com/QRAKVWWv
11:05.39mathiI would like to put the base directory of sounds, for /sounds/my-site
11:05.44ChannelZmathi: you literally have everything in /sounds  ???  not /var/lib/asterisk/sounds/....
11:06.07gavimobilenetvision is one of my few did providers
11:06.10mathiChannelZ, yes, and I created a subdirectory my-site (well, my actual site name)
11:06.19gavimobileI want to use them now only for incomming using my new server
11:06.45ChannelZmathi: do you have ALL your sounds there, like even the standard asterisk ones?  Or just your custom ones?
11:06.57*** part/#asterisk jaxon007_ (~jay@123.252.144.92)
11:07.15mathiChannelZ, I have put the sounds that were there in a directory "default", oops i shouldn't have done that?
11:07.32mathiby default there were some sounds there already
11:08.08ChannelZwell... is any of this actually working!? :)
11:08.16mathiobviosuly not
11:08.17mathi:p
11:08.21mathibut
11:08.22ChannelZLet me explain
11:08.26mathii want to have more stucture there
11:08.32mathiI may have several projects
11:08.50mathiI want in sounds to have the project names, and then the sounds for each project in these folders
11:10.10ChannelZwell if you want to break the structure like that then you're better off doing it yourself and having to specify the pathnames in your Playback/Background commands and such
11:10.24ChannelZbut moving everything around like that makes it difficult
11:10.40ChannelZBecause what of things like voicemail prompts?  None of that is probably working right
11:11.24ChannelZgavimobile: first, 'start' is an arbitrary extension name, it doesn't actually have any meaning UNLESS your ITSP supports extension registration and that's where you told them to send your calls.
11:11.34ChannelZTypically they send it to your DID.
11:11.48mathiChannelZ, it would be great if I can specify, asterisk default files should be fetched in /sounds/default, and my own files should fetched in /sounds/my-project
11:11.51gavimobileChannelZ: I didn't understand that
11:11.59ChannelZSo if your number is 5551112222 you catch those calls with exten => 5551112222,1,Answer   etc
11:12.28ChannelZmathi: well you can..
11:12.46gavimobileChannelZ: I used start cause I don't know how to make an inbound call dial say extention 301
11:13.09gavimobileI don't know the logic to intercept the inbound call
11:13.36mathiChannelZ, by specifying in each background, read, playback, ... or is there a way to set a new path from where to fetch automatically ?
11:13.36gavimobilecan't I just create 1 line of code to Ansewer all inbound calls from all my trunks?
11:13.48gavimobileor at least from one of my trunks?
11:13.57ChannelZmathi: sort-of anyway... hang on I can only do one conversation at once here
11:14.26ChannelZgavimobile: no
11:14.46gavimobilewell netvision forwards all calls to my number to my ip addresses
11:14.56ChannelZgavimobile: it varies from provider to provider but almost 100% of the time a SIP provider sends you the call to an extension if your DID
11:14.58gavimobilehow can I have asterisk actually forward me the call
11:15.22ChannelZgavimobile: To "intercept" that call you have to have a matching extension in that context
11:15.56gavimobileso I create an extention mathing the number? meaning if my number is 077xxxxxxx I create an extention 077xxxxxxx
11:15.58ChannelZWhether that's a pattern match or an exact numerical match is up to you but "start" has no meaning
11:16.13ChannelZ(well it has meaning, but they are not sending you calls to "start")
11:16.21ChannelZYes.
11:16.28ChannelZAnd that extension can in turn Dial() something else
11:16.55gavimobileChannelZ: could you be kind and give me a very simple example of how you would "intercept" the call
11:16.56ChannelZOr do any number of things.. like say hello first...
11:17.04ChannelZI did up above
11:17.13ChannelZexten => 12345,1,Answer
11:17.26gavimobilewhat does 12345 represent?
11:17.26ChannelZexten => 12345,n,Playback(tt-monkeys)
11:17.28ChannelZwhatever
11:17.32gavimobilemy 077 phone number?
11:17.35ChannelZ12345 is whatever your DID is
11:17.37gavimobileexter => _077
11:17.39gavimobileahhhhh
11:17.49gavimobilelet me try that
11:18.01ChannelZassuming that's what they send you.  If you call yourself you should see it come in on the console with some verbose turned on and know what exactly they're sending you
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11:18.48gavimobileWORKS
11:18.58gavimobilea day in a half im wokring on this
11:19.04gavimobilefor removing the word start
11:19.13gavimobileChannelZ: thanks man
11:19.21ChannelZmathi: the normal asterisk structure is to take the VarLib or data directory (I forget which but for me they are both the same) as configured in asterisk.conf, add 'sounds' to the path, and then add the language code, followed by what you told it to play.
11:19.43ChannelZSo for instance Playback(barf)  for me will attempt to look for a file in /var/lib/asterisk/sounds/en/barf.*
11:20.22ChannelZSo in other words when you do Playback or Background, it's a relative path.
11:20.42ChannelZPlayback(myproject/barf) would in turn look for /var/lib/asterisk/sounds/en/myproject/barf.*
11:22.28ChannelZYou can also specify an absolute path, like Playback(/sounds/myproject/barf)
11:23.31mathiin asterisk.conf I see the following:  astvarlibdir => /var/lib/asterisk
11:23.36ChannelZSo if you really want to put your custom sounds in that specific structure, then you should probably do 2 things.. 1. install all the core asterisk sounds into the normal locations in whatever languages you need to support them in
11:23.41mathibut the soudns are already fetched automatically in /sounds
11:24.58ChannelZyes but if you made an extension that did Playback(vm-password) exactly like that, you probably get nothing right now.
11:25.15ChannelZWhich is what you're going to get if you try to use Asterisk's voicemail
11:25.35mathiChannelZ, maybe I'll just mix all the sounds ....
11:26.03ChannelZwell I understand what you're trying to do, and you can do it, but you're making it very difficult on yourself.
11:26.13ChannelZIf you can live with re-thinking it a little, here's what I'd do.
11:26.57ChannelZMake your "project" folders inside the normal language folders.. so you'd have /var/lib/asterisk/sounds/fr/project1/... and /var/lib/asterisk/sounds/nl/project1/...
11:27.35ChannelZThis way you can still keep your concept of projects and have their sounds separate, they just happen to exist inside each language you want to support
11:27.50mathiright
11:28.00ChannelZThen when you playback your custom sounds from your dialplan, all you need to do is Playback(project1/somesound)
11:28.15mathiyes indeed that's better
11:28.26ChannelZand Asterisk will take care of getting the right language based on the previous work we did.  AND all of the system sounds will work.
11:28.32mathiwhat about the sounds that come with asterisk ?
11:28.34mathiah
11:28.53mathiso I don't create a folder "default" ine ach language with asterisk sounds
11:28.58ChannelZthey will just exist in /var/lib/asterisk/sounds/fr/....
11:29.00ChannelZetc
11:29.03mathiI wanted to seperate them as well
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11:29.12ChannelZThey need to be unique?
11:29.22mathilike project-1 project-2 default-sounds-of-asterisk
11:29.53mathinevermind, I'll just create the project folders then
11:30.35ChannelZBut there's no great way to hotwire the system sounds playback to a different structure.
11:31.08ChannelZI mean you can make a copy of them however you want and play them yourself, but for things like voicemail or SayDigits etc. they are expecting them to be in the normal structure.
11:31.11mathiChannelZ, the thing is that in sounds/en/ for example, it contains already a lot of directories and sounds by default, so my folders project1 and project2 will be among all these files and folders, I like when it's clean ...
11:31.32ChannelZso make symbolic links somewhere else to those project folders
11:31.39ChannelZlike /sounds
11:32.18ChannelZthen you can   ls /sounds   and only see your pretty folders :)
11:32.27mathibut I will post a suggestion for asterisk team because I think this can be improved if you think multi-project
11:32.49ChannelZwell I don't know what you consider a "project"
11:33.00ChannelZlike what that even means to asterisk
11:33.19mathiyea, I don't use the appropriate terms... yet...
11:33.45ChannelZplus that unless those sounds need to be unique for each 'project', why would you want to duplicate all the system sounds 10 times or whatever
11:35.23mathiwell let's say I record with my voice some prompts for a particular company's IVR, and another set of prompts using another person's voice for another company that uses the same Asterisk server
11:35.39mathiit would make sense to seperate these sounds, as they are different persons speaking
11:36.01ChannelZok so you're hosting many people's ivrs on the same system
11:36.13mathino it was just an example :P
11:36.21ChannelZyeah
11:36.25mathiI lied
11:36.26mathi:p
11:37.02mathithe thing is that I can find sound easier for a particular IVR system
11:37.03ChannelZwell either way the sounds used by core applications (like voicemail, conferencing, etc.) really want to use the common path.
11:37.15mathiif they are all mixed up, i'm afraid to be lost one day
11:37.36mathiok
11:39.41mathibusy changing the structure, it should work now)
11:43.09ChannelZGood luck - I'm off
11:44.24mathiChannelZ, good night)
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12:33.32PbxManhello I have this AGI call exten => 999,n,Agi(agi://localhost/hello.agi) does anybody know why it is not returning to the next line after the call?
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13:59.26gavimobileright now member=SIP/MEMBER1 rings MEMBER1 when someone calls the queue. how can I make the queue ring a cellphone number?
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14:21.21mathithere are no booleans in asterisk ?
14:22.10pabelangerIn where, dialplan?
14:25.04mathiyes
14:25.39mathipabelanger,  it seems there are no booleans, should I use number 1 and 0 then?
14:26.50pabelangermathi: yes
14:27.00mathioki thanks:)
14:30.23gavimobileanyone?
14:32.00pabelangergavimobile: repost your issue
14:32.04pabelangernot in scroll back
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14:33.21gavimobilepabelanger: right now member=SIP/MEMBER1 rings MEMBER1 when someone calls the queue. how can I make the queue ring a cellphone number?
14:34.26pabelangergavimobile: use a local channel to dial both SIP/MEMBER1 and cellphone
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14:36.23gavimobilepabelanger: does this mean I create a new user which when he's called it dials my cellphone?
14:37.05pabelangeryes
14:37.14gavimobileill try that
14:37.16gavimobilethanks
14:37.17pabelangeror, use an existing user and dial both
14:40.17mathiit seems I have a problem with this dialplan:
14:40.20mathisame => n,Gotoif(($[${time}>=7 & ${time}<=21) | ${time}="#"]?valid-time:invalid-time)
14:40.27mathiit accepts 5
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14:43.47mathican anyone see a problem with my dialplan?)
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15:50.51mathianyone knows why this accepts 5 as a valid time ?
15:50.54mathisame => n,Gotoif(($[${time}>=7 & ${time}<=21) | ${time}="#"]?valid-time:invalid-time)
15:53.47pabelangeryou expression is not right
15:53.56pabelangeryour brackets look to be wrong
15:54.40mathiit doesn"t seem wrong to me
15:54.58pabelanger($[${time}>=7 & ${time}<=21)
15:55.00pabelangeris not right
15:55.04pabelangerlook at the round brackets
15:56.03pabelangerAlso, if you are checking time why not us GotoIfTime?
15:56.10pabelangers/us/use/
16:01.01mathipabelanger, thanks it works when I fix the brackets
16:01.13mathipabelanger, but when I enter "#", it repeats the message
16:01.20mathiinstead of going to valid-time as specified
16:02.50kresp0when you enter "#" where?
16:03.27mathihere:   same => n(select-time),Read(time,dokagenda/${ARG1},2)
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16:11.37mathiI think by default pressing "#" it repeats my prompt??
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16:38.25mathiIf I press "#", the logs are saying "User entered nothing"
16:38.48mathiif I would type "*" for instance, then it would say "User entered "*""
16:39.04mathiI am using Zoiper
16:39.14mathianyone had this problem before ?
16:42.38pabelangermathi: *CLI> core show application Read
16:43.01pabelanger# is a termination for read
16:47.15mathipabelanger, is there any way I can use hash # as a usual input
16:48.14pabelangermathi: not with Read(), use WaitExten()
16:49.03pabelangerYou could always patch Read() to allow it, maybe create a new option for it
16:49.13mathipabelanger, in that case, which application do you suggest me to use with WaitExten to get the suer input? Unfortunately background redirects immediately to an extension
16:51.15pabelangerwhy do you need #?
16:52.20pabelangerBesure you could do with BackGround() is jump to a exten => ._,1,NoOp() then use logic to check the input
16:52.30pabelangerhowever, this will not work
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16:52.44pabelangerHow many digits are you expecting?
16:53.06pabelangerexten => XXXX,1,NoOp() for 1-9999
16:53.13pabelangerthen use dialplan logic
16:53.21mathipabelanger, ok I think I should just not use "#"
16:56.00carrarGreat idea
16:56.38mathiaha))
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16:58.03mathiso i replaced my dialplan with:  same => n,Gotoif(($[${time}>=7 & ${time}<=21) | ${time}="*"]?valid-time:invalid-time)
16:58.16mathibut if I type *, it goes to invalid-time
16:58.30mathiI also tried "${time}"="*", but didn't work
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17:10.51mathipabelanger, do you know why pressing * forwards me to invalid-time instead of valid-time priority ?
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17:17.47carrarWhy not use the if time commands?
17:19.00carraror GotoIfTime
17:19.31docgnomeso I'm trying to setup asterisk with google voice and... i think i have it sort of working? but i think it is at least connected to gtalk as i've gotten notices when buddies connect in asterisk -rvvvvvvv
17:19.41mathiGotoiftime:  "If the current time matches the specified time, ....."
17:19.50mathiI am not testing against the current time
17:20.01docgnomethough apparently i don't always get those
17:20.29mathicarrar, ^
17:20.44docgnomegetting super frustrated, and really have no idea what i'm doing.
17:21.04mathithe user enters an hour, and I want to check if it's valid
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17:21.32mathiand "*" to skip... but it doesn't work :(
17:22.30mathibeen googling for some hours now, just can't make a comparison with *  ....
17:24.29docgnomemy extensions.conf http://pastebin.com/mBuLs8eC my gtalk.conf http://pastebin.com/X0pnCkqn my jabber.conf http://pastebin.com/X50z6G7g
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17:30.56mathiso this works:
17:30.58mathiGotoif($["${time}"="*"]?valid-time:invalid-time)
17:31.06mathiif I press *, it goes to valid-time
17:31.17mathibut this doesn't:
17:31.18mathiGotoif($[(${time}>=7 & ${time}<=21) | "${time}"="*"]?valid-time:invalid-time)
17:31.35mathiso is it a problem with the & and | operators ?
17:32.01[TK]D-Fender&& <---
17:32.05[TK]D-Fender||
17:32.09[TK]D-Fenderdouble each
17:32.15mathithat's not what the docs says
17:32.48carrardon't do it then
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17:32.56mathidon't do what?
17:33.03[TK]D-FenderWhat the cods say
17:33.07[TK]D-Fenderdocs even...
17:33.14mathiahah) ok I'll try to use double
17:34.09mathiomg! works with double && and ||
17:34.33mathidocs need to be updated, lost too much time on this :|
17:34.34mathihttp://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-1.html#asterisk-CHP-6-SECT-1.2
17:35.21[TK]D-Fenderblitzrage: Copy Room stat!
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17:36.05[TK]D-Fendermathihttp://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-1.html#asterisk-CHP-6-SECT-1.2 <--- operator SNAFU, | and & should be doubled "&&" , "||"
17:36.11[TK]D-Fenderblitzrage: mathihttp://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-1.html#asterisk-CHP-6-SECT-1.2 <--- operator SNAFU, | and & should be doubled "&&" , "||"
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17:39.42[TK]D-Fendermathi: And ... you're welcome.
17:40.18mathi[TK]D-Fender, thanks for your help :-)
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18:26.02docgnomeok, so I now have outbound calling working
18:26.05docgnomehowever
18:26.41docgnomewhen i call in to my google voice account, it now rings and i pick it up but i never get connected. my physical phone i'm calling from just keeps ringing and goes through to voice mail
18:29.01docgnomehttp://pastebin.com/3YaeEgRp
18:29.04docgnomethis is the output i get
18:37.10docgnomeany ideas?
18:37.46docgnomeit's like the sip stuff doesn't get through to the uh
18:37.51docgnomegoogle voice
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19:43.36ChannelZdocgnome: are you doing the whole 'answer, wait a second, send DTMF 1' in your dialplan?
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19:57.52docgnomeChannelZ: it happens i can see it with asterisk -rvvvvvvv but i don't have that set anywhere i can see
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20:05.43ChannelZnot sure what you mean by you don't have it set...
20:06.14ChannelZin any event probably your DTMF isn't working or something.
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20:11.21docgnomei mean, i can't find SendDTMF anywhere in my configs
20:11.27docgnomebut it happens
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20:11.52CaplainContact: "matt" <sip:matt-iphone@192.168.1.64:5060>
20:11.58Caplaini dont even have that ip on my subnet
20:12.01Caplainwhere did i go wrong?
20:13.10Caplainalso i get audio when i dial locally but not when i dial out via gtalk from my iphone
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20:17.48ChannelZdocgnome: well it has to happen *somewhere* - extensions.conf
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20:23.02WIMPypabelanger: The patch for read to accept # is on jira.
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20:41.48ChannelZdocgnome: it might be missing the digit doing it in Dial. Make a different test; Answer(1000) the channel, then SendDTMF(1), then Dial(SIP/101)
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21:24.12artyxDoes anyone here build their own asterisk boxes?
21:26.15[TK]D-FenderOnly the vast majority of us
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21:43.13danfromukHi all. I've been trying to get Shared Line Appearance and BLF working and given up. I'm looking at alternative solutions to satisfy the client's requirements. Is it possible to have incoming calls placed in a RINGING state, allow staff to select which call they wish to answer, and using AMI and some software route the calls through to the staff's sip peer?
21:45.08artyx[TK]D-Fender, do you use a rackmount chassis/micro/itx type motherboard, or is it in a tower case
21:45.51[TK]D-Fenderartyx: Yes
21:46.05artyx[TK]D-Fendert: So yes, rackmount chassis/micro/itx type motherboard.. which type:?
21:46.18artyxFor example, i just built a box based on the D2700 mud (dualcore atom)
21:46.58[TK]D-Fenderartyx: I've used MATX, no mini.  I've down full ATX.  Arch is arch....
21:47.32danfromuk[TK]D-Fender: evening. Just thought i'd let you know what the Panasonic say that it will take them 12 months to support Shared Line Appearance with BLF. So giving up with them.
21:47.47artyxMOBO type? Arch is arch...  but could you be more specific, do you use amd ,intel, low wattage atom.. etc
21:48.04sruffellscratches head
21:48.39sruffellis curious where artyx is going with this line of questioning
21:48.49[TK]D-Fenderartyx: The problem is YOU are vague.  Just because something is in a Mini-ITX form-factor says NOTHING about the CPU sitting on it, ram, etc.  You may as well ask "has anyone made a server with a RED case?  I hear they're faster...."
21:48.57artyxDual xeon motherboard? The possibilities are endless of course.. Just wondering what other people use, your inherent refusal to come out with a straight answer only eggs me to pursue a real response
21:49.25[TK]D-FenderSo far your only real question is on SIZE, and nothing about what makes that special.
21:49.48artyxI even contributed an example. .the D2700 MUD (a dualcore atom) .. IF you looked it up, you would note that the D2700mud is the cutest little itx/micro atx motherboard and cpu bundle
21:50.00artyxSupports a whopping single pci slot...
21:50.07artyxbut perfect for a 1.5u rackmount chassis.
21:50.08[TK]D-Fender... ok, so "cute" is not the important factor....
21:50.13[TK]D-Fendernow*
21:50.14sruffellok…I'll bite…I've used tons….but one that's always fun is a Powermac B&W
21:50.31[TK]D-Fendersruffell: MONOCHROME NAZI!!!
21:50.42artyxsruffell: SIP only?
21:50.55[TK]D-FenderSometimes GULP too...
21:51.26[TK]D-FenderWhat does an Astribank count as?
21:51.52sruffellartyx: no…SIP, IAX, DAHDI
21:52.04artyxsruffell: You found a dahdi card for a powermac (g3/g4?)
21:52.05sruffellbut, I'm still wondering *why* you're asking these questions.
21:52.12sruffellartyx: yes…..
21:52.16artyxIt beats chopping wood
21:52.27sruffellwell…I'm not sure answering does then. ;)
21:52.57[TK]D-FenderWood > Answering pointless questions
21:53.10artyxtoday has been uhh lesse... chop wood,  drag the power-washer up onto the roof and wash the 43 solar panels off (under jet path, so they get dirty quick)  trim the roses, order gravel and sand for making retaining wall blocks (a project to piss off hte neighbor)
21:53.30sruffellI mean, honestly…if they sell it and it has transistors on it…someone is probably running Asterisk on it.
21:53.37artyxfinding out what people like to use for asterisk hardware is cakewalk comparatively
21:53.38artyx=)
21:53.50jpsharpsruffell: And/or NetBSD
21:54.09sruffelllol..true…now *that* is something I've not installed personally.
21:55.12jpsharpTis what I use if I don't have to install Asterisk.
21:55.22[TK]D-FenderAsterisk doesn't have a native plug-in for Cakewalk.  Take a look at Garageband instead.....
21:55.29sruffellalthough…to be accurate…I know of no installations of Asterisk on 8/16-bit micros…..but do know of versions that run on mmu-less 32-bit micros.
21:55.41sruffell[TK]D-Fender: lol
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22:25.59mathi[TK]D-Fender, hi
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22:34.36[TK]D-Fendermat'sup?
22:39.02mathi[TK]D-Fender, I am sorry for earlier, the right operators are "|" and "&", I didn't test well enough, so better cancel the doc change...
22:39.19mathiI am thus still stuck with my problem
22:40.58[TK]D-FenderThey should still be || and &&
22:41.35mathiit works well with | and &
22:41.47[TK]D-FenderNoOp($[ 1 || 1])
22:41.56[TK]D-Fenderthat should return 1
22:42.32mathiwether I use && or & ... my dialplan still doesn't work :(
22:42.33mathisame => n,Gotoif($[(${time}>=7 & ${time}<=21) | "${time}"="*"]?valid-time:invalid-time)
22:44.31mathithe testing for numbers works fine... if I enter 22 it goes to invalid-time, 21 is forwarding to valid-time, ... but * forwards to invalid time ... for some obscure reason
22:46.16[TK]D-FenderOh.... forget the hell out of brackets :)
22:46.17mathiit must be common though to test against "*" as it's one of the few special characters on a phone...
22:46.42mathi[TK]D-Fender, something wrong with the brackets ?
22:46.52[TK]D-Fenderyes, * = D-U-M :)
22:46.57[TK]D-Fender(no B)
22:47.02[TK]D-FenderB )
22:47.21mathiI don't follow sorry
22:47.22mathi:p
22:47.34[TK]D-Fender* parser = DUMB.  No bracket support
22:49.05mathi[TK]D-Fender, you mean { and } ?
22:52.01[TK]D-Fender() ,----------
22:52.19[TK]D-Fender(${time}>=7 & ${time}<=21)       <--- THESE brackets
22:52.33[TK]D-Fender{} = braces, not brackets
22:52.59mathiCurly brackets { }  from wikipedia
22:53.20mathi( )  <-  parentheses
22:53.22[TK]D-Fenderadd Citation required"
22:53.30mathihttp://en.wikipedia.org/wiki/Bracket#Square_brackets_.5B_.5D
22:53.45[TK]D-FenderYeah yeah... let's move on now ;)
22:53.59mathi:p
22:54.02mathiok fixing my code
22:54.09mathithank you for your help
22:54.19[TK]D-Fendernp
22:54.36mathi[TK]D-Fender, now i'll help you
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23:24.03artyxya?
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23:30.53ChannelZyyyup yup yup yup yupyupyupyupy
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