IRC log for #asterisk on 20140406

00:00.22lvlinuxok, i was under the impression (from how it was worded in the rather short section in the docs) that the SLA app auto-did that and then the phone would be dialing on the PSTN circuit directly.
00:00.57lvlinuxi'll try to mess with that context and see what I can get.
00:01.09[TK]D-Fenderapp = dialplan = you have to have it there
00:01.59lvlinuxhehe yep that's the way I'm used to doing it---then I got into this key system nightmare and I'm all confused lol
00:02.57[TK]D-FenderThe nightmare is only beginning...
00:03.11lvlinuxhehe that's what I'm afraid of...
00:03.37lvlinuxthankfully the end result is only a simple three PSTN line key system with three phones.
00:04.04lvlinuxnot that there's anything "simple" about it lol
00:09.23lvlinuxok i'm not getting it or something
00:10.10WIMPycore set verbose 3
00:10.15lvlinuxin my sla_stations context, on the extension that the line key speedials, i added a second priority that goes straight to VoicemailMain(101@default) and it doesn't work.
00:10.18lvlinuxok
00:10.21WIMPyand look at what's happening.
00:13.03lvlinuxi already had it set verbose---it says that its executing [station1@sla_stations:1] SLAStation("SIP/b101-0000000b", "station1")
00:13.11lvlinuxand then gives Auto fallthrough, channel 'SIP/b101-0000000b' status is 'UNKNOWN'
00:13.50WIMPySo that application exits and your dialplan doesn;t continue.
00:14.24lvlinuxyes---so obviously that's why it's not going on to the next priority
00:14.35lvlinux(which is the voicemailmain)
00:14.50WIMPyNo, there is no next priority.
00:15.02lvlinuxso how can i diagnose why it's exiting---there's no "sla set debug on"
00:15.15[TK]D-Fenderyou ran out of dialplan
00:15.23WIMPyNFI
00:15.27[TK]D-Fender[19:55][TK]D-FenderWhich * throws out if a call is accepted to the dialplan but nothing answers it or passes on any other status
00:15.37lvlinuxwhat's NFI?
00:15.45[TK]D-FenderNo &^#$ing Idea
00:15.59lvlinuxoh oh
00:16.00[TK]D-Fenderlvlinux: You aren't showing us anything useful.
00:16.25[TK]D-FenderIf you want an autopsy, give us a body
00:16.30lvlinuxok i'll pastebin some stuff
00:16.32lvlinux:-)
00:17.09lvlinuxi guess i figured i'd ask questions first to keep from wasting your time---but now i realize that that's wasting your time lol hehe
00:28.04lvlinuxok here's sla.conf: http://pastebin.com/LuXLnQVR
00:29.06lvlinuxsip.conf: http://pastebin.com/Q591bVek
00:29.35[TK]D-FenderStill means nothing
00:29.46lvlinuxextension.conf: http://pastebin.com/r86iM4Vx
00:29.49[TK]D-FenderYou don't seem to understand the very basics
00:29.56[TK]D-FenderYou've been told you have a DIALPLAN ERROR
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00:30.02[TK]D-Fenderthat is EXTENSIONS.CONF
00:30.14[TK]D-FenderAnd you aren't showing us the CALL
00:30.20lvlinuxyes well there's that---i just put the other two just in case you wanted to verify the contexts were correct.
00:30.36[TK]D-FenderShowing us raw configs also proves precisely nothing about what the phone is actually DIALING
00:30.49lvlinuxk i'll do the * output
00:31.53lvlinux* output when I press the line 1 button on the phone: http://pastebin.com/60a0XnjZ
00:32.08lvlinuxdo you want a sip debug too?
00:32.31[TK]D-Fender[20:14]lvlinuxyes---so obviously that's why it's not going on to the next priority
00:32.32[TK]D-Fender[20:14]lvlinux(which is the voicemailmain)
00:32.52[TK]D-Fenderhttp://pastebin.com/r86iM4Vx <- there is no "voicemailmain" in here at all
00:33.01[TK]D-FenderNor is there a 2nd priority on anything
00:33.03lvlinuxwell i got rid of that and went down to basics
00:33.22lvlinuxnow the extensions.conf i'm using is exactly what you see
00:33.49[TK]D-FenderYou're creating a moving target by the time you are showing us anything.
00:37.33lvlinuxhere's where I had the Voicemail line: http://pastebin.com/jdKJttGL
00:38.53lvlinuxand here's the output: http://pastebin.com/Z3f17ynt
00:40.06[TK]D-Fendervoicemail looks a little messed up for aborting like that
00:40.27[TK]D-FenderAnd senseless to use there, but is a bit besides the point
00:41.05lvlinuxyes i know it's senseless---just using it for testing
00:43.10[TK]D-FenderWhat ver of * are you running?
00:43.29lvlinux11.7
00:45.13[TK]D-FenderIs dahdi configured and running from CLI?
00:45.29lvlinuxyes
00:45.30[TK]D-Fender"dahdi show status", "dahdi show channels"
00:45.49lvlinuxstatus is good
00:45.57lvlinuxchannels show the three lines in service
00:46.08lvlinuxand the contexts line1, line2, and line3
00:49.00[TK]D-FenderHrm, the guide I'm seeing say that the dialplan is"generated"
00:49.49lvlinuxthat's if you put the autocontext=line1 line at the trunks and stations in sla.conf
00:50.09lvlinuxor autocontext=lineX rather
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00:50.23[TK]D-Fenderexten => station2_line1,1,SLAStation(station2_line1)
00:50.47[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Basic+SLA+Configuration+Example
00:51.14[TK]D-FenderLooks like they've changed a thing or two since I saw it in 1.4
00:51.40lvlinux:-)
00:51.56[TK]D-Fenderthat dialplan line looks like it uses a hybrid name to refer to the sla.conf sections to pick the device
00:52.18lvlinuxyes
00:52.42lvlinuxyou mean with the underscore---ie station1_line2
00:52.51[TK]D-Fenderyes
00:53.33[TK]D-Fender<PROTECTED>
00:54.05[TK]D-Fenderexten => station1_line1,1,SLAStation(station1_line1)
00:54.07[TK]D-Fenderlike this one
00:54.16[TK]D-FenderSo it looks like you made the wrong choiice for your speed-dial
00:54.37[TK]D-FenderWhich should ahve been to "station1_line1"
00:54.47lvlinuxok i'll give that a shot.
00:55.16lvlinuxBTW when I dial in on the DAHDI lines, the phones don't ring, but the line lights on them do blink correctly for the DAHDI line
00:56.40lvlinuxat least i'm not using polycom phones---these take long enough to reboot but not like the polys lol
00:59.08[TK]D-FenderYou shouldn't have to be rebooting them all the time
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00:59.30[TK]D-FenderAnd that's a terrible basis for choosing a phone
00:59.35lvlinuxyou have to when you change a SIP parameter
00:59.43[TK]D-FenderWhich, don't worry... will likely manifest later
00:59.46lvlinuxlol i didn't choose them for that---i like the polycoms a lot better
01:01.46lvlinuxok * gave the same output as before---should i try something besides Voicemail to test this?
01:02.08lvlinuxi should say same effect as before not same output let me PB it.
01:02.40lvlinuxhttp://pastebin.com/L6ppFgJc
01:06.26[TK]D-Fender[20:45][TK]D-Fender"dahdi show status", "dahdi show channels"
01:06.28[TK]D-FenderPB it
01:06.31lvlinuxk
01:07.13lvlinuxhttp://pastebin.com/UxZBqS1E
01:07.42lvlinuxthe DAHDI ports are hooked to an ATA on the other side if that makes any diff.
01:09.29[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration <- I'm seeing some more odd context's required for this in this guide page
01:09.57[TK]D-Fenderthat handles dialing, etc.... mind you this is with a non-live SIP provider where you have to colelct the # first...
01:10.05[TK]D-FenderThis is really a mess...
01:11.59lvlinuxhehe
01:12.47lvlinuxyeah i tried it with that type of setup w the [lineX_outbound] contexts and it didn't work so that's why i tried to simplify it back down just to get it working first.
01:13.35lvlinuxi don't need SIP dialing though on this system since they only have 3 PSTN lines.
01:15.02[TK]D-FenderNo, I see that.... it looks like it maybe should be working at this point (since you're dialing what it seemed to imply you shave have)
01:17.41lvlinuxif i replace the VoicemailMain call with a call out on DAHDI (same =>   n,Dial(DAHDI/1/18887779999) I get:
01:18.01lvlinuxhttp://pastebin.com/edFVXtdz
01:18.10lvlinuxcongested/busy
01:19.08[TK]D-Fenderthat would imply it is already in use...
01:19.18[TK]D-Fender"core show channels"
01:19.29[TK]D-Fender"dahdi show channel 1"
01:19.32lvlinuxit's not0 active channels
01:19.32lvlinux0 active calls
01:20.16lvlinuxhttp://pastebin.com/nyA69vUd
01:20.25lvlinuxthats' dahdi show cahnnel 1
01:20.37lvlinuxsry "dahdi show channel 1"
01:22.02lvlinuxwait a minute it says it's off hook
01:22.07lvlinuxhmmm
01:24.35lvlinuxwhen i unplug it it says onhook
01:24.49lvlinuxbut when i plug it back in it immediately says offhook
01:24.59lvlinuxsame thing for channels 2 and 3
01:27.07lvlinuxsame behavior if i plug it into the actual PSTN line instead of my ATA
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01:38.17[TK]D-FenderI'd check into that...
01:38.42[TK]D-FenderOk, I've drilled about as much into this as I can go....
01:38.50lvlinuxwhat do you mean check into what specifically?
01:38.56lvlinuxthe TDM card?
01:39.22[TK]D-FenderFor all of the downsides (not being able to use "redial", killing your CDR's, etc... this really seems like a waste to try to fake out SLA
01:40.01lvlinuxyou're telling me!!!! you know that saying "the customer is always right" lol well that's what I've gotten myself into here :-(
01:40.35lvlinuxsmall pizza place---ancient analog PBX switcher key system thing--static, dropped calls, unpredictable.
01:41.14lvlinuxi go and install a nice little * box with 3 IP phones, and they say they can't stand not being able to say "pick up line 3 bob..."
01:41.47[TK]D-FenderThat's what parking is for
01:41.54lvlinuxand they think a blind transfer is "good for an office but not efficient for here..."
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01:43.11lvlinuxhmmm i had thought about parking but didn't look into it too much.
01:43.38lvlinuxcan you have the BLF indicators showing the status of parked calls maybe?
01:43.55[TK]D-FenderPark call.  Get lot #.  Hang up.  Yell "Bob, 702!"
01:44.15[TK]D-FenderYou can get presence on your lots as well
01:44.23lvlinuxnope they won't go for that---- it's the "702" that they'll complain about.
01:44.32lvlinuxthey want to press one button on a blinking light
01:44.48[TK]D-Fenderwith the lots having BLF they CAN just hit that buttone <-
01:45.15lvlinuxhmm, ok and then I could do a speed dial to the parked lot?
01:45.30[TK]D-Fenderdepending on the phone, sure
01:45.45lvlinuxdepending on the phone---yeah i was afraid of that.
01:45.53lvlinuxi don't think i can w these phones
01:46.18lvlinuxit only lets me set BLF for shared lines I think :-(
01:46.51[TK]D-FenderObviosly not.
01:47.06[TK]D-FenderBecause what the phones consider a "shared line" does not exist in Asterisk
01:47.20[TK]D-FenderThe only thing you were using them for WAS presence + speed-dial
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01:47.30lvlinuxoh?
01:47.33lvlinuxhmmmmmm
01:47.43[TK]D-Fenderyour "SLA" WAS a speed-dial.
01:47.53lvlinuxso was i not supposed to set the phones to shared line mode---ooops
01:47.56lvlinux?
01:47.57[TK]D-FenderYou missed the big-print somehow...
01:48.10[TK]D-FenderOh dear God....
01:48.20lvlinuxlol i'm going to wet myself
01:48.21[TK]D-Fender* does NOT support SLA.
01:48.44[TK]D-Fenderthose apps = cheap-hack to fake it out.
01:48.52lvlinuxahhhhhhhhhh
01:49.03[TK]D-FenderTHE CAKE IS A LIE
01:49.20lvlinuxno wonder it wasn't working right
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01:49.53[TK]D-Fenderwell I saw it dialing those "text" extensions... I don't really get it at this point....
01:50.41[TK]D-Fenderthat LOOKED like it was just doing a speed-dial across your primary (appearance
01:51.01lvlinuxi think the only way to get BLF on these phones is to set it as DSS/BLF/shared line mode,
01:51.20[TK]D-Fenderthat doesn't make sense as a combo.
01:51.29[TK]D-FenderBut... no time to dig further.  Got to head out...
01:51.40[TK]D-FenderSeriously look at parking and save yourself a world of pain.
01:51.59lvlinuxif i can fake it out enough where they think it's like the old system then that's what I'll do.
01:52.00[TK]D-Fenderheads out....
01:52.02lvlinuxThanks so much
01:52.05lvlinuxas always
01:52.18lvlinuxbeats head on wall
01:56.30lvlinuxwell i've got to try to get BLF working on a private line with these phones then.
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08:02.40SconkHi is there any way of logging in the debug log what password a phone is trying to register whit ?
08:10.04jeffspeffSconk, are you monitoring the * console?
08:10.14jeffspeffwhen the phone tries to regiser?
08:10.25Sconkyes
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08:11.49Sconki keep getting wrong password..
08:11.59Sconkand its a sip client
08:12.35jeffspeffdid you verify the password in the sip.conf file and the password in the phone's config?
08:14.04Sconkthe freepbx gui shows right password budt i will try looking in the files
08:15.18Sconkits the right password
08:15.36Sconkbudt its a old ACN iris 300  i think its playing some tricks with me
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14:24.17YannikSHi, when originating calls using AMI no CallerID is sent to the internal phone specified in 'channel'. It should be set to the external number which is being called. How can I fix this?
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15:53.20logical2Good morning, I built asterisk from source 11.5.1 back in Sept of last year.  I would like to recompile to the latest version..   is there a tutorial anyone is aware of..  my Google searches produced little results..
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16:33.39filewobbles in
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17:01.18logical2found answer to earlier question, solved. narrative: "wget src, tar, configure, make menuselect, make install"
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20:32.29filesilence!
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20:54.36slav3_kittenhuh i wonder
20:54.38slav3_kitteni should google to see if i can have inbound numbers give my cisco phones a different ring tone
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21:49.22bsdiceyo
21:55.25ChannelZyoyo to the mofo
21:59.06bsdicetalking to me?
21:59.09bsdiceno mofo :)
21:59.23bsdicebut I have good news... still running asterisk and my phone still works ;)
22:03.02ChannelZWell that's a good thing
22:04.32bsdiceyeah, really liking Asterisk
22:04.38bsdicecan't believe it's free
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22:29.03filefeel free to contribute back in any way you can to help the project ;) documentation fixes, helping on here or the -users list, etc
22:33.09bsdiceI will
22:55.20[TK]D-Fenderfile: IIRC there is a new option in sip device entries in sip.conf to use that peer's settings for registrations which is needed in the case of having to go out a specific proxy.  Do you recall it offhand?  I can't see it in the v11 sample config
22:55.49fileI think in the register line instead of specify a host you can specify the name of the peer entry
22:57.40[TK]D-FenderThat would be uniform to how Dial handles "@"
22:57.54[TK]D-FenderAlmost too clean ;)
22:58.03fileit's common code.
22:59.14[TK]D-FenderDo you know when this got put in?
22:59.29fileoh geez, probably 1.2 or 1.4 days
23:00.54filethere was talk long ago about putting the registration itself into a peer, or having a new type, but I don't think anything happened
23:02.17bsdiceyou mean like callbackextension= ?
23:02.50bsdicevisited a friend in Germany and tried to get that going for Deutsche Telekom IP VDSL, failed
23:03.04fileah yup
23:03.11bsdicehad to go with register= in general section
23:03.13filethe mess that is chan_sip
23:03.41bsdicehrhrhrhr
23:03.59bsdicehey, haven't tried 12 yet, because 11 trunk is so damn stable here
23:04.22bsdiceread somewhere calling SIP "extensible hell" and sure it is
23:04.35bsdicenever fell in love with H.323 though
23:04.54bsdicethat always smelled proprietary Cisco, Siemens and whatnot
23:05.11BludSuckingFiendh.323, yuck... ASN1 and tons of vendor extensions
23:05.23bsdicesorry... TRIGGER WARNING
23:05.25bsdicehehehe :D
23:08.09BludSuckingFiendGet to try putting h.323 through NAT this week and not looking forward to it
23:08.22filerealizes the sound he is hearing is rain
23:08.39BludSuckingFiendnot that it is much better with SIP. Makes me wish IAX caught on across vendors
23:09.19filethe methods of doing it with SIP are understood these days, better than it was
23:09.37*** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta)
23:10.32BludSuckingFiendYeah, though it is one of the reasons I've always hated the FTP protocol.
23:10.54BludSuckingFiendDifferent data/control channels
23:14.47bsdiceenter SFTP ;)
23:14.57bsdiceSIP over NAT is well-controlled these days indeed
23:15.36bsdiceeven though the hoops pjsip stack jumps through to get it going is crazy
23:15.56BludSuckingFiendI end up using TDM trunking anyway though
23:17.29bsdicehow many phones you got under management?
23:17.44BludSuckingFiend~200
23:18.09BludSuckingFiendon asterisk anyway
23:18.33bsdiceI'm just doing it for friends, with encryption. maybe 10-20 SIP accounts
23:19.34BludSuckingFiendYeah, I've found SIP offerings from a lot of enterprise companies lacking.
23:19.51bsdiceever played with autoframing?
23:20.11BludSuckingFiendUnfortunately high density TDM is fairly opaque on asterisk
23:20.16bsdicetried to get big frames going with speex to limit overhead on mobile, not working
23:20.22bsdiceprobably too stupid
23:20.29BludSuckingFiendnope, never
23:20.51bsdicethrew out the last TDM card about half a year ago
23:20.56bsdicehaven't looked back
23:21.09BludSuckingFiendJust bought a quad span board
23:21.17bsdiceeicon?
23:21.19BludSuckingFiendand an R515 box to put it in
23:21.25BludSuckingFiendSangoma
23:22.52BludSuckingFiendThe company's been running 200 phones and 3 ISDN PRIs off a desktop box for awhile
23:23.05BludSuckingFiendfinally convinced them to actually buy decent hardware for it
23:23.42bsdiceI had been running ISDN BRI for a while, sadly driver was not stable, memleak in newer kernels
23:23.45bsdice>3.0
23:24.13bsdicehad worries that fax would no longer work
23:24.24bsdicebut with a-law no problemo
23:24.46bsdice(via iaxmodem and hylaxfax behind that)
23:25.06BludSuckingFiendah, yeah
23:25.13BludSuckingFiendUsing that for a fax server too
23:25.20WIMPybsdice: Which driver?
23:25.38bsdiceugh
23:25.42bsdicehang on 1 sec
23:26.18bsdicezaphfc
23:27.26WIMPyAh, ok.
23:28.04bsdiceWIMPy if you compare the D-channel hardware irq handler with other HFC-style drivers, it is no wonder. "kaputtgefummelt"
23:28.30bsdiceshame but then, at least in Germany where the box is, ISDN is on its way out
23:29.59WIMPyI prefer to use LCR over DAHDI. Has worked very well since I finally made the change.
23:30.36BludSuckingFiendit certainly is old. Hopefully providers will be able to get their act together supporting VoIP properly
23:30.59BludSuckingFiendfor now I just use PRIs for any critical traffic
23:31.03WIMPyI wouldn't expect that in the near future.
23:31.10bsdiceGerman Telekom works, if you know what you are doing. Deeply.
23:31.32bsdiceI think I cursed for 10 hrs until I had it going and this is a very big provider in Germany
23:31.39WIMPyWhat works? Do they already have full SIP-I?
23:31.51bsdicewhat is SIP-I
23:32.05WIMPyAn attempt to make SIP usable.
23:32.20WIMPySome horrible mishmash of SIP and H.323.
23:32.21bsdicewell we registered 10 numbers and been using it successfully
23:33.08WIMPyOr ASN.1 with SIP compatibility or whatever.
23:33.18bsdiceno aware of them supporting that
23:33.28bsdiceat least they appear to have fixed session-timers
23:33.56bsdiceP-Asserted-Identity or whats it called is still a problem
23:34.08WIMPyIn what way?
23:34.11bsdiceneed to tweak Asterisk pretty good to get it to do what you want
23:34.17bsdicehang on looking for snipped
23:34.48bsdiceif you set ;trustrpid=yes then your caller ID will be off
23:35.09bsdiceoff in the sense, that asterisk displays RPID instead of From
23:35.27bsdicewhich means you get the -0 instead of an extension
23:35.32bsdiceconfusing
23:35.43bsdiceSet to yes causes P-Asserted-Identity override
23:36.22bsdiceanother real nutbreaker is their login casino
23:36.47bsdiceneed to use the email address, not the VDSL dialup login/pass and also not their customer center's login/pass
23:36.52bsdicewhich all is different
23:36.58bsdicenailbiting...
23:37.26bsdiceand finally iptables with stateful block of incoming connections
23:37.37bsdicei.e. block 5060 from external sources
23:37.58bsdicewhich I solved by using qualifyfreq=50
23:38.32bsdiceto keep the state valid towards tel.t-online.de
23:38.41bsdiceso external calls reach the box still
23:40.43WIMPyDoesn't Asterisk also support keep-alives?
23:41.50bsdiceI am abusing qualify
23:42.38bsdicedon't know
23:45.18*** join/#asterisk Dovid (~Dovid@ool-2f113725.dyn.optonline.net)

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