IRC log for #asterisk on 20100518

00:00.53puckett_jwHowever, I am NoOp'ing some stuff…. Executing [s@subStandardExt:5] NoOp("IAX2/pbx-3221", "["incoming" != "spinen"] is 1") in new stack Executing [s@subStandardExt:6] NoOp("IAX2/pbx-3221", "{CUT(ARG1,:,2-)} is 10") in new stack
00:01.13puckett_jwWhich shows that they are not null
00:04.04puckett_jwI am using the new BETA Adium, can someone acknowledge that my posts are coming through?
00:04.34SaiSoma|AtHomepuckett_jw: i'm looking, but no expert here.  i'm guessing a syntax error, still looking
00:04.46SaiSoma|AtHomecan you post the NoOp, raw, no the output?
00:04.48puckett_jwAwesome, thanks
00:05.20puckett_jwHere are the lines...
00:05.24puckett_jwexten => s,n,NoOp(Call started in ${CDR(dcontext)} and is going to ${CDR(accountcode)})
00:05.24puckett_jwexten => s,n,NoOp(["${CDR(dcontext)}" != "${CDR(accountcode)}"] is $["${CDR(dcontext)}" != "${CDR(accountcode)}"])
00:05.24puckett_jwexten => s,n,NoOp({CUT(ARG1,:,2-)} is ${CUT(ARG1,:,2-)})
00:05.24puckett_jwexten => s,n,Set(LOCAL(nextSet)=${IF($["${CDR(dcontext)}" != "${CDR(accountcode)}"])?${CUT(ARG1,:,2-)}})
00:05.40SaiSoma|AtHome*nod*  just FYI . .before fender does it
00:05.42SaiSoma|AtHome~pastebin
00:05.43infobot[~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
00:05.44SaiSoma|AtHome:)
00:07.42paulcpucket_jw: I'm still not clear what your problem is - can we have the bullet point problem definition statement?
00:07.48puckett_jwBTW - I am running Asterisk 1.6.2.7
00:08.40puckett_jwI am trying to set a varable to Null if the dcontext & accountcode are the same
00:09.01puckett_jwI set the accountcode the the context of the extension elsewhere in the call
00:09.27puckett_jwI am checking to see if the call is an internal to same context
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00:10.13puckett_jwI am trying to use exten => s,n,Set(LOCAL(nextSet)=${IF($["${CDR(dcontext)}" != "${CDR(accountcode)}"])?${CUT(ARG1,:,2-)}}), but I am getting the error… Syntax IF(<expr>?[<true>][:<false>])  (expr must be non-null, and either <true> or <false> must be non-null)
00:11.16puckett_jwI know my expression is not null… NoOp("IAX2/pbx-3221", "["incoming" != "spinen"] is 1")
00:11.45puckett_jwAlso the true is not null… NoOp("IAX2/pbx-3221", "{CUT(ARG1,:,2-)} is 10")
00:12.26puckett_jwTherefore, I cannot figure out what I have done wrong.  I am sure that I have probably left a character out somewhere, but I cannot find it
00:13.10puckett_jwI was also woundering if the ":" in the CUT function could be throughing off the IF function?
00:13.21paulcWhat's your CUT line all about? Describe its use?
00:14.36puckett_jwSo I am wanting to make the subStandardExt sub get called in a way that it would be recursive, where ${ARG1} could be… .<extension>[|dialtimeout][|device][:<extension>[|dialtimeout][|device]:...]
00:15.31puckett_jwMy plan is to have the sub call its self on certain DIALSTATUS' when additional extensions are passed
00:16.04puckett_jwThis way a call could go to a device, and the rollover to another device if not answered
00:17.52puckett_jwHere is the start of the sub… http://pastebin.com/BdG6VsQi
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00:18.59puckett_jwpaulc: Does that make sense?
00:26.41paulcMaybe it's late in the day and my brain is fried, but I still don't get it
00:27.00paulcmixing stuff passed in (arguments, multiple destinations?) with destinations vs contexts?
00:28.41paulcbacks away slowly - it's almost home time
00:29.11puckett_jwpaulc: OK.  Thanks
00:36.31puckett_jwI found it!  I had a ) in the wrong place
00:37.06puckett_jwIt should have been exten => s,n,Set(LOCAL(nextSet)=${IF(["${CDR(dcontext)}" != "${CDR(accountcode)}"]?${CUT(ARG1,:,2-)})})
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00:56.26pkecastillohello guys!
00:57.02pkecastilloI need the "exten number", with AMI commands... any help?
00:57.09pkecastillonot with the extension inbound, else with the anexo of ring group, while your state is ringing...
00:58.40pkecastilloI test with "ExtensionStatus", but nothing! :(
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03:13.34devdvdhi all
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03:16.44sprite--I'm running into a very weird issue. Moving my development environment over to a production environment. Asterisk does not detect hangup from app_conference, versions of everything are the same. No errors or warnings are being thrown. Also tried app_konference which has a bunch of updates and latest asterisk from 1.6 branch. Neither fixed the issue.
03:19.16[TK]D-Fendersprite--: Wheres the failure for us to examine?
03:19.58sprite--[TK]D-Fender: I can generate some logs real quick from working vs non-working. But like I said it's not throwing any errors. After I hang up both parties. They channels are still showing as active if I do sip show channels.
03:20.10sprite--Asterisk is never generating a hangup event.
03:22.26pabelangersprite--: indications loaded?
03:22.27[TK]D-Fendersprite--: So far that has precisely nothing to with with app_conference.
03:23.01[TK]D-Fendersprite--: If * is not told by the remote client "Hey I'm done with this call" then you have ANOTHER problem altogether
03:23.34sprite--pabelanger: Nope seems I am missing my indications.conf let me copy that over and see if it fixes the issue.
03:23.47[TK]D-Fenderindications also has nothing to do with SIP.
03:23.52sprite--Earlier when I tried with ast 1.6 I had indications.conf though and it didn't solve anything
03:23.58[TK]D-FenderSIP is afull-progress signalled protocol
03:24.45sprite--[TK]D-Fender: It detects hangup when it is not a conference call. It is only when a party is in a conference it does not work.
03:25.14sprite--I will get some logs real quick of events working vs non-working server.
03:25.38pabelanger[TK]D-Fender: :) didn't see the part about it being SIP
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03:26.36philippelhey all - does anyone recall the specific 1.4.X version that queues added the additional device state fields?
03:26.41[TK]D-Fenderpabelanger : People always miss the big print
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03:27.45[TK]D-Fenderphilippel: Thtought that should be 1.6.  branches aren't supposed to get new features mid-stream like that
03:28.18philippel[TK]D-Fender you actually don't want to get me 'started' on that one, but it was back ported into 1.4 somewhere aroudn 1.4.25ish
03:29.19[TK]D-Fenderphilippel: Must have taken a lot of arm-bending and favours
03:29.20philippeland the sad thing was that the 'fix' was not a proper fix and hasn't gotten proper until trunk (1.8) when they finally added  HINTS as an option (from a 1.4 patch I submited over a year ago)
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03:29.48[TK]D-Fenderphilippel: Hate to say it... hit the changelogs <-
03:29.50philippelI wasn't part of it happening, just part of trying to get it 'right' with hints
03:30.09philippelI am - just thought someowne might have known off the top of their head :)
03:31.18philippelr184980 | mmichelson | 2009-03-30 08:23:59 -0700 (Mon, 30 Mar 2009) | 22 lines
03:31.25philippelso now to translate that into a versoin number
03:33.06[TK]D-FenderShould be easy enough.
03:36.22[TK]D-Fenderphilippel: 1.4.25-rc1
03:36.34[TK]D-Fender2009-05-13
03:36.55[TK]D-Fenderphilippel: Or the full release following : 2009-05-21 - 1.4.25
03:37.08philippelI was just honing in on that :)
03:37.35philippelthanks
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03:39.02sprite--[TK]D-Fender: Ok the issue was somehow related to trying to start MixMonitor with a recording directory that did not exist. It seems to be working now. Very weird.
03:41.14[TK]D-Fendersprite--: Problems always disappear when you look at them.. they're like faeries!
03:42.13sprite--[TK]D-Fender: I don't see how MixMonitor failing to start should completely break hangup decection in asterisk though.
03:42.22sprite--detection
03:44.00sprite--but creating the missing directory for my mixmonitor fixed the hangup issue in conference and it now properly detects a hangup.
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03:58.42sprite--[TK]D-Fender: I lied. It randomly worked for some reason once. Now it's not working again. I will get a log real quick.
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04:06.25sprite--[TK]D-Fender: http://gist.github.com/404572 This time Asterisk detected one of the hangups.
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04:09.14shmaltzhi everyone
04:09.25shmaltz~sleep
04:10.13infobotsleep is probably overrated, and a poor substitute for caffeine.
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04:10.16shmaltzis infobot asleep?
04:11.33shmaltz~google asterisk
04:14.30shmaltzdoes anyone know what the market share of asterisk is?
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04:19.52vutamhoanI got this issue - https://issues.asterisk.org/view.php?id=15915  - Any suggestion is appreciated. (Static build fail)
04:22.05sprite--[TK]D-Fender: Hit me up when you get back. Updated the gist with a 3rd attempt where needed party is detected. Willing to pay if you can figure out a solution to get this sorted.
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05:08.30BugKhaMHi, what's controlling the order of cards loaded in /proc/zaptel? Mine keeps changing every time the system reboots
05:10.06BugKhaMtry changing the order of modules loaded in /etc/sysconfig/zaptel or changing priority of script loading in /etc/init.d/zaptel but the problem still persists
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05:15.01spenguin[work]TEST
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05:17.57k-mananyone ever tried to make a poe injector for a linksys phone?
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05:22.27p3nguinI made one for my Cisco desk phone.
05:23.34drmessanoI try to inject poe wherever I go
05:23.56drmessanoQUOTH THE RAVEN, NEVERMORE
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05:27.09k-manp3nguin: can you point me to some good instructions for doing it?
05:27.26k-mancan i use the power adaptor that came woth the phone?
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05:29.40p3nguink-man: http://pinouts.ru/Net/poe_pinout.shtml
05:30.14p3nguink-man: I used a regular power brick that I did have plugged into the phone.  Now it just plugs in at the jack rather than on the phone.
05:30.31p3nguink-man: http://www.interfacebus.com/Power_Over_Ethernet.html
05:30.41k-manp3nguin: yeah, thats what I want to do
05:31.51p3nguink-man: Does your cable come out of a wall jack already?
05:32.08k-manp3nguin: yes
05:32.32k-manbut i plan to move the ethernet jack to a new location, and add poe at the same time
05:32.49k-mancurrently the voip phone is next to the TV which is useless as I'm always on the couch when it rings
05:32.52k-mani need it next to the couch
05:32.54k-manhehe
05:34.18p3nguink-man: I used a Radio Shack 274-1576 dc power jack, mounted it in the wall plate, disconnected the brown and blue pairs from the back of the rj-45, and wired the power from the dc jack to those free pins on the rj-45.
05:35.00p3nguinI did it so I only have one cable for the phone instead of two.
05:36.35k-manyeah, i could do something like that
05:36.37p3nguinYou need to make sure your device is able to be powered by PoE mode B, which powers the unused pairs rather than powering the data pairs.
05:36.52k-manah, its a linksys SPA942
05:36.57k-mannot sure which mode of POE it uses
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05:38.39k-manhmm.. i could just buy a PoE switch for about $50 off ebay
05:38.45k-manthat might be a much easier option
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05:41.02spenguin[work]k-man: should be easy
05:41.07spenguin[work]if you arent following standards
05:41.08k-manyeah
05:41.16spenguin[work]only 4 wires are normally used
05:41.21spenguin[work]for a 100mbit line
05:41.24k-manhmm... ok
05:41.29k-manill look more into it
05:41.47spenguin[work]leave out 1236
05:41.56spenguin[work]the rest are for you to play with
05:43.43k-manhmm.. i need to buy a roll of networking wire so i can run this new network cable under the house
05:44.21k-mancan you harm a non poe device if you plut it into a powered port?
05:46.58k-mans/plut/plug
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05:48.07p3nguinNot much easier, really.  It should take you a couple minutes to drill a hole in the wall plate and mount the power jack, then another couple minutes to solder and wire the power to the ethernet jack.
05:53.21p3nguink-man: I guess it depends on if you want to spend $50 or $3.
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05:55.58p3nguinI'm seeing info saying the phone supports 802.3af mode A, but nothing saying it doesn't support mode B as well.
05:58.03k-manwhats the difference? any idea?
05:58.16p3nguinYes.  I thought I already mentioned it.
05:58.26k-manoh, maybe i missed it
05:58.31p3nguinMode A powers over the data pair, mode B powers over the unused pair.
05:59.13k-manah, its just the pins
05:59.15p3nguinFor wiring up a dumb injector like I did, I wouldn't advise trying to power over the data pair.
05:59.24k-manok
05:59.26k-manfair enough
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06:00.11p3nguinMode B uses the pairs not needed for data, so it's okay to cut them from the jack to disconnect it from the switch.
06:00.46k-manah,  isee
06:01.27p3nguinSo what I did was disconnect the blue and brown pairs from the back of the rj-45.
06:01.48p3nguinThat allows the data pairs to still be connected from the switch to the jack.
06:02.10p3nguinThen mount the power jack in the wall plate and wire the power to the unused pairs.
06:02.33p3nguinNow power is only on the jack and not the switch.
06:03.08k-mani was thinking to put the power plug in the cupboard where i have my switch, and inject the power there to the relevant ethernet cable
06:03.23k-manbut same diff, i can do the same thing in the cupboard
06:03.31p3nguinI don't know if the switch ports can handle power being applied when it isn't supposed to have it, so I just disconnect it entirely.
06:03.44k-manyeah, i see what you mean
06:04.32p3nguinI'll have to get some pictures of mine.
06:04.40p3nguin(tomorrow)
06:05.31Jumpiep3nguin...was scrolling up on your convo
06:05.42Jumpieso basically you're having him split away the power pins
06:05.45Jumpieand just connect data?
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06:07.03p3nguinPull the wall plate off and look at the back of the jack.  You've got 8 wires, but only 4 are used.  Take off the unused pair from the back of the jack and move them out of the way.  Now wire the dc power jack to those unused pins on the back of the rj-45.
06:07.17p3nguinunused pairs, rather.
06:08.05p3nguinLeave the two data pairs connected, power the unused pins.  This makes a "dumb" mode B power injector.
06:08.38Jumpiebut i thought the whole idea was, him wanting to use a poe switch to a non poe device
06:08.44Jumpiewhy even bother with the powered wires?
06:08.53Jumpiejust so safety's sake?
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06:09.04p3nguina single Ethernet cable as opposed to an Ethernet cable and a power cable
06:09.17p3nguinHe wanted it for mobility.
06:09.18Jumpieoooooooh
06:09.23Jumpieou're talking ghetto riggin it
06:09.49p3nguinIf that's what you call it, but it's not really rigging at all.
06:10.08p3nguinIt's the same technology that all those $25 PoE devices use.
06:10.09Jumpiewell you are changing the way it was originally intended
06:10.10Jumpie:D
06:10.12florzJumpie: you do have at least a basic understanding of electrical engineering, don't you?
06:10.23Jumpieim a telcom guy, and yes
06:10.30Jumpiei just hvent done a seperaton of a poe device/line like that
06:10.31Jumpiehavent had a need
06:10.41p3nguinIf the device supports mode B, then it was intended to be used that way.
06:10.52Jumpieim just saying, most people use poe devices with poe switches
06:11.03Jumpieyour explanation is great, i was just sayin it was a rig
06:11.04Jumpiehhe
06:11.29Jumpiewhen i meant 'intended' i guess i meant, anything other than using a single cable for power/data
06:11.43p3nguinYou aren't making sense to me.
06:11.52drmessanoHe IS using a single cable
06:12.03Jumpieer well, sorry
06:12.16Jumpiei guess what i meant was, instead of just plugging the end right into the phone, as a poe device normally
06:12.16drmessanoJust injecting the power at the wall
06:12.21Jumpieyea...
06:12.24k-manno, the SPA942 is a poe device
06:12.27Jumpiethast not what the manuf intended
06:12.33Jumpieim not sayin its bad lol it was just a comment
06:12.33p3nguinSure it is.
06:12.40k-mani have a non POE switch, so i want to inject power between the switch and the spa942
06:12.47drmessanoWho cares where the power is injected?
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06:12.51p3nguinIf they built it to use mode B PoE, then that was how the mfg intended for it to be used.
06:12.56Jumpiek-man i could have sworn you said you had a poe switch going to a non poe device
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06:13.02drmessanoAt the wall plate, in an inline injector, or at the switch
06:13.15Jumpieok, sorry for confusion
06:13.22p3nguinIt's still a midspan PoE injector, so what's the difference?
06:13.32k-manJumpie: i was saying that sometimes PoE switches go cheapish on ebay, it might be worth trying to pick one of those up instead of fiddling around
06:13.35drmessanoIf he has a poe switch and a non-poe device, wouldn't he just make a sandwich?
06:14.07Jumpiep3nguin, i guess i meant, most people use poe devices without the intention of having to do any punching, soldering
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06:14.19p3nguinWhy does it have be manufactured rather than fabricated?
06:14.21Jumpieagain, i wasnt bagging on your explanation
06:15.02drmessanoMost people run Windows too.  Doesn't make it right
06:15.15p3nguinIt took me 5 minutes and $3 to inject PoE.  I think that's pretty frugal.
06:15.51Jumpieheeh
06:15.52Jumpieok ok
06:16.03Jumpiei think you're gettin overvly defensive about something i wasnt saying was bad at all
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06:16.17Jumpiei was simply saying, most people buy poe devices/switches to just plug and play
06:16.17drmessanoI kinda like the idea of cutting the end off of a wall wart, punching down the ends on the unused pins inside the plate, and running to the nearest receptacle
06:16.18p3nguinYou called it a ghetto rig.
06:16.19Jumpieand youre method was great
06:16.35drmessanoIts actually very clean
06:16.50k-manyes, its a good plan, i'll have a good look at doing it when i get home
06:17.42drmessanoand you could grab any 5v 1A+ power supply from ebay.. especially one with a zip cord and not the round cable on the linksys supplies
06:17.49drmessanoMake it easy to cut and punch
06:18.32p3nguinProbably ought to get a 48V one instead of 5V.
06:19.00Jumpiedrmessano that somethin chepa you could get at like, radio shack too?
06:19.03drmessanoDo the Linksys phones use 48V standard PoE?
06:19.05Jumpieor graybar, grainger
06:19.06BugKhaMHi, how to change the order of the zaptel's channel drivers loaded in the kernel
06:19.07p3nguinI guess it might depend on what you're powering, though.
06:19.18drmessanoJumpie: If you want to pay 5x as much
06:19.49p3nguinI have to assume that since they are 802.3af compliant, they'll want 48V over Ethernet.
06:21.24p3nguinI did learn that my older (legacy) Cisco phone is designed with Cisco's old standards (non-802.3af) and mode B is reverse polarity.
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06:22.48p3nguinI'm not brave enough to piggy-back power onto the data pairs, so everything worked out just fine.
06:23.20p3nguinI potentially could have blown up my phone and switch port.
06:23.51Jumpiehehe
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06:38.41maxagazcan I include a directory in asterisk to manage users.conf and extensions.conf ?
06:39.07p3nguin#include ?
06:39.39p3nguinI think you're supposed to include files, not directories.
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06:41.18maxagazp3nguin, directories would be much convenient
06:41.49p3nguinI don't know if directories are able to be included since I have never done it.
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07:28.27eject_ckHi all
07:29.43*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
07:29.47eject_ckI have 3 sip clients (SPA3102, Nokia E52, X-lite) behind NAT. Asterisk is in internet :). When I'm trying to register 3rd device - then one of already connected peers goes unreachable. WHat the best practice to deal with this ?
07:30.21Ziaeoni dont know if its good practice
07:30.29Ziaeonbut you could try changing registration port
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07:30.52ZiaeonI've seen certain ISP's/modem combos dislike various instances of 5060
07:31.38kaldemar~sipnat
07:31.39infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
07:32.00eject_ckkaldemar: tahnks!
07:32.53kaldemarso tell asterisk that the clients are behind a nat and turn on qualify.
07:33.21eject_ckdid it already
07:33.29eject_ckno any additional configs on clients ?
07:35.26kaldemarnot really.
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07:40.03ksnany idea of this error? http://pastebin.org/246635 should be kernel module related, but the modules are loading fine
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07:42.21*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.7, 1.6.1.19, 1.6.0.27 (2010/05/04), 1.4.31 (2010/05/04), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs
07:42.36Ziaeoncustomers are tired of hearing "you need a better network"
07:48.19kaldemarksn: which modules?
07:49.13ksnkaldemar, wctdm, now it loads in asterisk too, but continue to tell me "channel 1,2 ignored" and i don't receive calls :\
07:49.36ksnseems i had just to rmmod and modprobe again a couple of time
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07:51.29visioconfGood morning everybody
07:51.47eject_ckkaldemar: what do you mean ?
07:53.31kaldemareject_ck: that you shouldn't need any additional configuration on the clients.
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07:54.57kaldemarksn: seems that dahdi is not properly configured or 1-2 are not valid channels.
07:55.32eject_ck^)
07:55.44eject_ckkaldemar: I mean NAT specific settings
07:55.49eject_ckok, got it working :)
07:56.46ksnkaldemar, where can i debug more the problem? the config generated seems fine (i just had to change country code)
07:57.10eject_ckbtw, I have SPA-3102 as VoIP gateway to PSTN and vice versa on FXS line. Does it supports to use 2nd Fxs port simultaneously (make to simultaneously calls)  as well ?
07:57.39visioconfany ideas how to be a voip minutes seller ?
07:58.00visioconfI mean what must I have
07:58.51kaldemarvisioconf: a clear vision of what you want your business to be.
07:59.45Xt0fGood Morning, Good Morning. I have implemented TLS with AMI but unfortunately I got the msg tcptls.c:218 handle_tcptls_connection: FILE * open failed! when trying to connect to port 5039.  I am using CentOS 5.5 and Asterisk 1.6. Any ideas ?
07:59.51kaldemarksn: what does your /etc/dahdi/system.conf look like?
08:00.39ksnhttp://pastebin.org/246705 <- :p
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08:01.02visioconfyeah, I want to sell voip minutes to other reselling companies. But i'm serching for the tech side
08:01.30kaldemarksn: and the output of lsdahdi (or cat /proc/dahdi/*)?
08:01.39visioconfkaldemar: I need any docs, cases studies , requirements docs...etc
08:01.45BANSALcan anybody help me .. I am unable to install asterisk ...
08:01.48BANSALhttp://pastebin.org/246706
08:02.04BANSALI tried and got these errors ..
08:02.19BANSALcan anybody comment on this ?
08:02.21ksnkaldemar, http://pastebin.org/246708
08:02.27kaldemarvisioconf: selling voip minutes is quite vague. you must be more specific.
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08:03.59kaldemarksn: looks like the channels are configured and even in use. how did you configure the channels in /etc/asterisk/chan_dahdi.conf ?
08:04.37visioconfKaldemar, OK I'll try to specify it. I need to have an idea about the needed technology and infrastructure to have a voip server and a billing system in order to sale minutes on that server to a specific user
08:05.36ksnkaldemar, pretty much standard stuff, want a pastebin?
08:05.41kaldemarvisioconf: voip only? no pstn interfacing?
08:05.51kaldemarksn: sure, i'll take a look.
08:06.32visioconfkaldemar, as a first step only VOIP
08:07.09ksnkaldemar, http://pastebin.org/246718
08:07.26visioconfkaldemar: however f there is always pssibility to interface with pstn it would be excellent
08:09.00kaldemarvisioconf: well, you obviously need something to do the VoIP part for you. what is the most suitable, depends on your needs. it might be a B2BUAS like asterisk or a proxy like kamailio or opensips. the billing part is done by some other software. for asterisk, a2billing seems to be used quite widely for an example. many people also write their own interfaces to asterisk's call data records.
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08:10.18BANSALplease help me...
08:11.03visioconfkaldemar, and how could I specify that such user have such quota via such gateway, is it managed via asterisk ?
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08:11.46kaldemarksn: have you tried configuring the channels under [channels]? with "channel => 1-2"?
08:12.19ksnmh actually not
08:13.48ksnmh still ignoring
08:15.23kaldemarcheck dahdi-channels.conf for some conflicting configuration.
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08:15.58Janosgot a small question, is ael recommended for use with 1.4 ? or should i upgrade to 1.6 if i want to use it ?
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08:16.48kaldemarJanos: you can use it just as well with 1.4.
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08:17.28kaarlohi
08:17.32Janoskaldemar, sweet, then i think i'm going ael, thanks a lot
08:17.43kaarlois there any example how to use lua dialplan on asterisk?
08:18.26kaldemarkaarlo: configs/extensions.lua in the source package
08:18.56kaarloah, okay
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08:19.16kaarlothank you, nvm that there is a example lua configuration ;)
08:19.53kaldemarextensions.lua.sample is the correct filename..
08:20.16kaarloyep, found it at /usr/share/asterisk/conf/extensions.lua.sample
08:20.42kaldemarBANSAL: have you tried a newer version of libpri?
08:21.16BANSALkaldemar:  I am using 1.2.2
08:22.24Xt0fHello, any ideas relating to my TLS + AMI problem ? Thanks in advance.
08:23.04kaldemarBANSAL: any particular reason for not using a newer version?
08:23.53BANSAL<PROTECTED>
08:24.26BANSAL<PROTECTED>
08:24.31kaldemarBANSAL: http://downloads.asterisk.org/pub/telephony/libpri/
08:24.32Janosanother question, having a static queue(no agents, just sip channels) how does asterisk determines if the queue has someone to answer on it ?
08:25.15Janosi have this simple queue, and i have to add joinifemtpy=yes otherwise sip channels won't ring
08:25.42ksnkaldemar, dunno seems fine to me, it can be a kernel/hardware problem?
08:25.53BANSALkaldemar: should I use libpri-1.2-current.tar.gz
08:26.37kaldemarksn: really looks strange to me, can't see a reason for that, sorry.
08:26.47kaldemarBANSAL: what version of asterisk are you using?
08:27.14BANSAL<PROTECTED>
08:27.16ksnkaldemar, okaay thank anyway :)
08:27.36BANSALnewer one is supposed to be unstable I think ...
08:29.12kaldemarBANSAL: use 1.4.10.2
08:29.50BANSAL<PROTECTED>
08:30.09drmessanoLibpri 1.2.2 is 4 years old
08:30.23drmessanoYou need to be on 1.4.10.2
08:31.00Xt0fNobody for my TLS + AMI issue....? ;-(. ...thanks in advance...
08:32.13kaldemarBANSAL: i'd suggest that you move to the newest 1.4 release of asterisk and the newest dahdi (zaptel was renamed to dahdi ages ago). if you insist on using zaptel and asterisk 1.4.20, use the latest release which is 1.4.12.1.
08:33.47visioconfkaldemar, I need any doocuments about haw to setup an IP callshop
08:33.55BANSALkaldemar: ok .. so the errors where just because of libpri version ...
08:33.59kaldemarBANSAL: if this is your first install, use one of the 1.6 branches. 1.6.0 and 1.6.1 branches will soon be switched to securify fixes only. 1.6.2 is the latest release branch.
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08:34.58kaldemarvisioconf: you probably won't find any before you specify what you need in more detail.
08:35.53visioconfKaldemar, I'm newbie in such techs so could you please tell me what do I need to sepecify ?
08:35.59BANSALkaldemar: well since I am totally new to asterisk so I am less concern about version ... if you say asterisk 1.4.20 zaptel 1.4.12.1 and libpri 1.4.10.2 will work fine .. then I can proceed with it.
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08:38.58kaldemarvisioconf: the service you want to provide. what exactly you're looking to sell and to whom and how.
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08:40.30kaldemarBANSAL: if you're new to asterisk, don't take old versions. you'll be more likely to run into trouble with them. take the latest releases.
08:41.02BANSALkaldemar: ok .. I'll use the latest one ... :)
08:41.15BANSALkaldemar: thanks for helping me ..
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08:47.00Xt0fkaldemar : could you help me with Open SSL and AMI ? ;-)
08:49.55BANSAL<PROTECTED>
08:50.27BANSALkaldemar: make: *** [makeopts] Error 1
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09:29.00carrarYAWN
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09:41.06af_I am connnecting a patton 4110 (vesion 5.x firmware) to an asterisk box. do you know of any example? (obly 4.x, that is different)
09:41.56Chainsawaf_: I have a 4118 connected to Asterisk 1.6.2.6
09:42.00Chainsawaf_: Let me know what you need.
09:42.20Chainsawaf_: Mine only has FXS ports, so I can't help you with FXO matters.
09:42.31af_well just a working example. it looks like there isnt' of firmware 5.x
09:42.39af_plenty of firmware 4.x
09:42.52Chainsawaf_: I'm glad I'm not the only one who found that.
09:42.53af_syntax differs. no comments on that
09:43.08Chainsawaf_: And the 400-page software configuration guide is mostly "aren't we great" waffle. Yes.
09:43.09af_it looks it's impossible to downgrade, also
09:43.39Chainsawaf_: Tell you what; send me an e-mail and I'll mail you my 4118 config back. I have 4634 config as well if that's helpful.
09:43.46af_and, the reset to factory defaults button does not work
09:43.48Chainsawaf_: I just need some time to scrub passwords etc.
09:44.03af_you are really kind
09:44.04Chainsawaf_: tony@linx.net
09:45.31af_sent
09:45.45af_just going to eat something
09:46.13af_I will send you back my working conf, if I will have success
09:46.18af_see ya
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10:13.24bonbonhello all, I'd like to know if Asterisk is the right product I should be evaluating for a system that would tie a phone number with a customer. The customer would enter the phone number on our website and would be issued a 6-digit code. He/she will then call a toll-free number from the registered phone and enter the code. An application will then match the incoming phone number and the entered code with an entry in a backend data
10:13.51carrarsure
10:13.56carraryou'll need to write that
10:14.01carrarbut isn't very hard
10:14.24bonbonthat's fine. but for call handling, tracking the entered code and the incoming phone number etc - all right up asterisk's alley?
10:14.32carraryup
10:14.37bonboncool, thanks
10:15.11carrar& dinner
10:15.48bonbonissues a six-digit code for dinner
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10:49.35oktayHello guys. Is there a VOIP phone/router that has OpenVPN builtin?
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10:51.23florzoktay: for at least some snoms there is firmware with openvpn available from snom
10:52.43WIMPySnom 370 and 8xx AFAIK
10:53.43WIMPySome Draytec Vigors also have openvpn support, but I don't know if you can route your voip traffic that way.
10:55.40oktayi was looking at snom too .. their official forums made it sound like it might not work.
10:55.56oktayWIMPy: 370 is the minimum that supports openVPN?
10:56.10WIMPyYes
10:56.30oktaythis is for Dubai by the way. so I am open to suggestions against blocking.
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11:08.20oktaylet me see if i can buy snom here..
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11:16.18oktaya cheapo .. zycoo zp302 seems to support it
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11:22.23oktayanybody know at what level Dubai blocks VOIP ?
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11:28.39eject_ckoktay: OSI7 I guess ;)
11:29.21oktayi hope
11:29.47eject_ckoktay: and even on OSI 7 for RTP
11:30.16oktayyou know how they block Youtube here ?
11:30.24eject_cks/OSI 7/OSI 5/g
11:30.33carrarJust VPN out
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11:30.44carrarthen you can pass whatever you want
11:30.46oktayyes. i am looking for a phone that supports it
11:30.49eject_ckoktay: I have no idea :)
11:30.53oktayeject_ck: DNS
11:30.56carrarsetup a VPN
11:31.01oktaythey point to IP to a page that says it's blocked
11:31.04carrarso everything on your nework is open
11:31.12eject_ckLOL :(
11:31.23*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
11:31.35oktayif you use another DNS server, you're good to go.
11:31.42carraror dump your nazi blocking ISP
11:32.05oktaycarrar: it's not nazi. it's dubai. it think it's a sheik
11:32.18oktayand that is where the client is. no blocking on my side.
11:32.37oktayi want to make it simple. so a hardware phone is probably the best choice.
11:32.48carrarif the client is worth it, setup a site to site vpn
11:32.51*** join/#asterisk mikkel (~mikkel@130.226.36.170)
11:33.03oktaycarrar: he's my brother :)
11:33.14oktaywe have vpn
11:33.15carrardefintely more trouble hen it's worth :)
11:33.37eject_ckGuys I'm playing around features.conf and can't understand why it not works for me :). I'm able to get tt-monkeys via Dynamic Feature (*6) in my case, but can't get any of  builtin features like Attended Transfer, One Touch Monitor working :(. What I missed ? I see them on features show but my keypress ignored
11:33.37carrarIf you have a VPN, why not use it?
11:33.40oktaywe have openvpn and I'm looking for a phone that can use it out of the box
11:34.03carrarpass your SIP through your existing open VPN tunnel
11:34.07carrarSIP & RTP
11:34.21oktaywhat would the phone have to have for that to work?
11:34.21eject_ckoktay: add youtube hosts to u r local hosts file :)
11:34.32carrarphone wouldn't need to know
11:34.33oktayeject_ck: i use opendns. No problem.
11:34.43oktaycarrar: i'm interested
11:35.01eject_ckoktay: I can recommend symbian (I have Nokia with VPN support)
11:35.18carrarPhone just things it's connecting to another IP address, just happens to go over your exsiting openvpn tunnel you already have established between your two sites
11:35.24carrarthinks
11:35.34oktayah. need a server on the other end too
11:35.40*** part/#asterisk Xt0f (~d54c8bc2@gateway/web/freenode/x-hioushtxffwxjpdo)
11:35.46oktayor.. perhaps and extra nic on his box
11:35.57oktayor just ocnfigure the gateway..
11:36.09oktaybut he's running windows.. forwarding would suck
11:36.33carrarbuild a cheap firewall/openvpn server
11:36.41carrarand set it to him
11:36.42carrarsend
11:37.14eject_ckCan someone help me ?
11:37.15oktaythat is not a bad idea.
11:37.15carrarthis way all your lan to lan traffic is encrypted and secure
11:37.48carrarand do the same for anyone else in your family
11:37.54carrarlinik them all together
11:38.37carrarYou'll find other uses for this secure family VPN network I'm sure
11:38.39oktayi don't know if i want to have a remote server to look after.. i have to think about that.
11:38.50carrarthen build it on unix
11:39.03oktayunix?
11:39.33carraruse small little atom board
11:39.58oktayhm.. just an extra pci..
11:42.27carrarnice little Intel ZOTAC mobos work great
11:43.15carrarand can be used for other things
11:44.19oktayyup
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12:01.27*** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net)
12:01.58WIMPyoktay: If I remember correctely, you can use the Snom 360 as router for the vpn.
12:02.40oktay360?
12:02.58WIMPyErr, 370, sorry.
12:03.01*** join/#asterisk iajrz (~irving@190.80.186.180)
12:03.01oktayah. ok.
12:03.06oktayi am trying to locate one to buy
12:03.09iajrzhey everyone
12:04.48iajrzneeone here running asterisk on centos?
12:05.07oktayblasphemy
12:05.24iajrzugh
12:05.35iajrzam I going to be burnt alive or something :P
12:05.37oktaywhat seems to be the problem specific to centos?
12:06.12iajrzbeen having issues for about a year, almost once a month. ofr some reason, the filesystem detects orphan inodes or some such
12:06.17iajrzand asks to do fsck manually.
12:06.51oktayduring a reboot?
12:06.55iajrzthe site this is set up, there's permanent electricity, and I've changed the hardware gradually...
12:06.55WIMPyiajrz: Doesn't belong here, but are you sure, your hardware is ok?
12:07.18iajrzafter reboot... its working, then stops working, so needs reboot.
12:07.39oktaywhat stops working that needs a reboot? asterisk?
12:07.41iajrzWIMPy: you can never be too sure, but I've gradually changed the whole PC
12:08.08iajrzoktay: the PC dies. not only asterisk, but everything stops working.
12:08.11oktaymy experience with linux says a reboot does NOT fix things :)
12:08.20iajrzI infer it might be a centos issue...
12:08.37oktayit's more probably a hardware issue
12:08.42iajrzreboot wont fix bug, alright. but when I reboot then fsck -y, it gets back to running
12:09.06iajrzI feared it might be so... just wanted to check if anyone had a simmilar issue
12:09.45iajrzso meanwhile, I've been trying to find out how to make the PC run fsck as if it was me when it goes into repair mode so I didn't have to come here once a month.
12:10.02iajrzjust tell em "reboot the bitch!" and get her working
12:10.19oktay<PROTECTED>
12:10.46WIMPyRemember the Simpsons episode with the wooden parrot or whatever it was?
12:11.10iajrzoktay: it dies. Asterisk dies, at the very least.
12:11.25oktaywhen you restart it?
12:11.29oktayasterisk alone
12:11.31iajrziirc, cuz lately the people here have been rebooting it by themselves
12:11.46iajrzby my recollection, everything from prompt to cron dies.
12:12.02iajrzso hard reset is needed : /
12:13.32iajrzanyways, just to get something to do, any pointers in what hardware might be faulty? any previous experience?
12:14.15oktaywhat have you swapped so far? board? memory?
12:14.36iajrzI swapped the whole box.
12:14.44iajrzfrom case to CPU, you name it, I changed it
12:14.53iajrzexcept... oh, god. Except for the T1 card.
12:15.20oktayyou don't see anything int he logs?
12:15.32*** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger)
12:16.25*** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163)
12:17.59iajrzuhm... I see a stop.
12:18.00iajrzliterally
12:18.14iajrzall these entries per second, then silence.
12:18.35oktaywhat does 'last' show?
12:19.00iajrzuhm... anything from a call going in to a call going out or plain old regular stuff
12:19.08oktayno no
12:19.11oktaynot asterisk
12:19.13oktaysystem logs
12:19.28oktayyour system is hanging.. you should not try to debug asterisk
12:19.46*** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net)
12:19.50iajrzI checked asterisk just in case... the rest looks good, too.
12:20.40dmastg'morning all
12:20.50oktayg afternoon
12:20.52iajrzmorning
12:21.48oktayiajrz: asterisk stops working. you look at the screen and you see? what?
12:21.56iajrza prompt.
12:22.07oktayand you can log in?
12:22.10iajrza dead prompt.
12:22.11iajrznope
12:22.20iajrzkeyboard wont answer either
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12:22.31oktaydead cold
12:22.32oktayweird
12:22.39oktayit might still be the T1 card driver
12:23.15iajrzI had that shivering idea. It's a Rhino R2T2 card
12:23.36iajrzdrivers were downloaded from the web.
12:23.41oktayfancy name. i've never used a T1 card
12:24.35iajrzugh
12:26.09iajrzwell, I'll just have to swap hardware.
12:26.17iajrzthanks a lot, oktay
12:26.36iajrzgoes home, feeling noobish
12:26.43oktayhope it works.
12:27.08iajrzthx :)
12:27.39*** part/#asterisk iajrz (~irving@190.80.186.180)
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12:36.27oktaysnom 370 is quite expensive here..
12:37.05oktay250 euros
12:41.02*** join/#asterisk smellynoser (~ashley@87-194-183-38.bethere.co.uk)
12:41.09smellynoserHas anybody ever used asterisk with a Swyx ISDN card?
12:41.11smellynoserIs it possible?
12:41.47smellynoserAlso, is there a good SIP client that's recommended for Windows?
12:41.56*** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek)
12:42.51[TK]D-Fendersmellynoser: We don't typically recommend soft-phones period
12:46.37oktayis there a voip gateway product wiht openVPN (< $200) ?
12:52.16*** join/#asterisk devyll (~paul@thpallady.net.hostway.ro)
12:53.15oktaywhich is which again ? FXS, FXO ? :)
12:53.19devyllhello. logfile:  -- Registered SIP '<sipusr>' at <sipclient_ip> port 6202  . does 6202 represent the source port or the destination port ?
12:53.28*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net)
12:53.30devyllI want to secure from firewall all sip registration
13:00.16oktayi think the ports should be well documented
13:00.18oktaywhat do you need?
13:01.13*** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at)
13:01.38devyllI need to restrict all connections to sip registration from all ips except two of them from iptables. and I wanted to make sure that udp:5060 is what I'm looking for
13:04.10oktayi have 5060 and 5061
13:04.24*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
13:04.41oktayhttp://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules
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13:10.32oktaylater guys
13:10.36*** part/#asterisk oktay (~oktay@81.215.202.193)
13:11.27[TK]D-Fenderdevyll: You could just use the permit/deny masks per-host....
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13:29.19waKKuhi folks.. a quick doubt
13:30.00waKKuam I able to "cancel" a blind transfer in SIP ? .. I mean, if I press # to transfer, but want to get back to the call.. is it possible?
13:30.48waKKuof course, if i no longer type the exten to transfer to
13:30.52WIMPyI think, you can just press # again.
13:31.14waKKuhm... seems no work here, using xlite
13:31.59ManxPowerreal phones have a transfer button
13:32.11WIMPyRight. It's the hangup code. *0 for me, but I think I changed that.
13:32.21WIMPyIndeed.
13:32.37*** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net)
13:32.44waKKuhm.. lemme try it
13:32.49WIMPyBut it only works on IP phones :-(
13:32.58ManxPowerhuh?
13:33.14*** join/#asterisk frk2 (~faraz@zivios/member/fkhan)
13:33.28WIMPyHave you ever tried a transfer on an ISDN phone?
13:33.29*** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net)
13:33.40WIMPyOr any other feature except hold for that matter.
13:33.44waKKunops.. i don't have one ;(
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13:36.33*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net)
13:37.12*** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com)
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13:40.12*** join/#asterisk [psy] (psy@lounge.datux.nl)
13:40.44[psy]is it possible to let asterisk call to its self by sip, without using any external servers? i want to do this to create a regression test for asterisk
13:43.00waKKuyou mean, call files?
13:44.31*** join/#asterisk hatoon (~zsggncb@187.111.253.180)
13:44.40[TK]D-Fender[09:29]<waKKu>am I able to "cancel" a blind transfer in SIP ? .. I mean, if I press # to transfer, but want to get back to the call.. is it possible? <- that is a DTMF Asterisk-based transfer, not a "SIP transfer", and no... blind is blind
13:44.42pabelanger[psy]: look into the Asterisk testsuite
13:44.53pabelanger~testsuite
13:45.10[psy]pabelanger k i will
13:45.13[TK]D-Fender[psy]: Sure
13:45.15pabelanger[psy]: http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/
13:45.36[psy]waKKu call files in combination with a good extensions.conf and sip.conf
13:45.40waKKu[TK]D-Fender, ok.. thanks :)
13:45.43[psy]but i'll look into the suite
13:46.08*** part/#asterisk hatoon (~zsggncb@187.111.253.180)
13:46.15*** part/#asterisk ManxPower (~manxpower@216.186.151.147)
13:50.47[psy]pabelanger what i wanna do is make lots of calls and playback a file, then record the file on the other end of the conversation, and after that compare the call quality
13:51.04*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
13:51.10[psy]you cant do that with sipp or pjsua, both of which we currently already use in our own asterisk test suie
13:56.02pabelanger[psy]: You can use Originate command
13:56.07*** join/#asterisk ltd (~z@pat.transact.net.au)
13:56.55*** join/#asterisk notjohn (~john@106.165.61.69.DED-DSL.fuse.net)
13:57.26*** join/#asterisk bsaxon (~bsaxon@66.76.242.154)
13:57.26notjohnwhere can i find detailed info on reading/querying CDR logs?
13:57.34*** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net)
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13:57.52[TK]D-Fendernotjohn: In the docs included in your source tarball
13:58.35*** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
13:58.40notjohn[TK]D-Fender: Does it address the actual data or just the field descriptions for the cdr database table?
13:59.07[TK]D-Fendernotjohn: Did you look?
13:59.47notjohnI've seen README that describes the cdr log table and it's lacking the details of what an "s" ... "h" etc means... how to determine pages/transfers from real calls
14:00.19[TK]D-Fendernotjohn: those are extensions in your dialplan... if you don't know what those are... time to read Chapter 5 of THE BOOK
14:00.21[TK]D-Fender~book
14:00.22infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
14:00.50[TK]D-Fendernotjohn: And CDR isn't transactional per-call.  Its a "lastapp" type record.
14:00.59[TK]D-Fendernotjohn: Single per-call
14:01.42notjohn[TK]D-Fender: meaning each action is logged.... not just each call right?
14:02.34[TK]D-Fendernotjohn:  "isn't transactional" <-
14:02.36[TK]D-FenderNOT
14:02.59[TK]D-Fender[10:01]<notjohn>[TK]D-Fender: meaning each action is logged.... not just each call right?    [10:00]<[TK]D-Fender>notjohn: Single per-call
14:03.08[psy]pabelanger thx, however, that does the same as using callfiles, correct?
14:03.32*** join/#asterisk sjb_gt (~sachajber@71-15-84-164.dhcp.gnvl.sc.charter.com)
14:03.34kaarlohmmm... how did i get the asterisk dialtone when i pick up my soft phone?
14:03.37kaarlolike the real phone
14:03.55*** part/#asterisk sjb_gt (~sachajber@71-15-84-164.dhcp.gnvl.sc.charter.com)
14:03.59*** join/#asterisk Guest28029 (~GBove@208-104-67-26.dyn.fttp.comporium.net)
14:04.01kaarloi want to make it people easier to change to voip
14:04.04pabelanger[psy]: yes
14:04.20*** join/#asterisk BANSAL (~BANSAL@117.199.120.56)
14:04.35[psy]so can i create to sip peers and let 1 register to asterisk itself?
14:05.18[TK]D-Fenderkaarlo: When you "pick up" your softphone... that isn't * providing a dialtone.
14:05.46kaarlois it only possible over FXO?
14:05.47*** join/#asterisk krion (~seb@unaffiliated/krion)
14:06.02[TK]D-Fenderkaarlo: Your device generates the tone.
14:06.24kaarlookay... over FXO asterisk would generate the tone, right?
14:06.39kaarloehm, i mean FXS
14:06.54[TK]D-Fenderkaarlo: FXS devices all generate dialtone.
14:07.01[TK]D-Fenderkaarlo: That is implicit
14:07.10kaarlookay
14:07.13[psy]so can i create two sip peers and let 1 register to asterisk itself?
14:07.22kaarlothanks for informations
14:08.09*** join/#asterisk devdvd (~myemail@173-31-160-214.client.mchsi.com)
14:08.16devdvdmorning all
14:10.20[TK]D-Fender[psy]: You don't need to register for your test.
14:10.31[TK]D-Fender[psy]: You jsut need to place calls, and yes, you can.
14:10.41[psy]oki
14:10.50[psy]i'll fiddle around a bit more
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14:11.18*** mode/#asterisk [+o jtodd] by ChanServ
14:11.59Faustovnathalie?
14:16.30devdvdim trying to dynamically add a member to a queue with addqueuemember that comes in over a sip trunk.  so the full line would be soemthing like SIP/trunk/1234567890.  When I have AddQueueMember(queueName) in my dialplan and dial in with a locally connected extension (SIP/100) it works fine, adds SIP/100 to the queue and moves on.  But when I call in off the sip trunk it only adds SIP/trunk.  I have verified Add QueueMember can add the full
14:16.40devdvdor is it even possible
14:17.27devdvdusing asterisk 1.6.2.7
14:17.33*** join/#asterisk Raden (~Raden@71.89.121.119)
14:19.29*** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ptzivwbihwuvpqse)
14:19.34notjohn[TK]D-Fender: I guess i'm confused as to what a "call" includes then
14:19.54[TK]D-Fendernotjohn: "includes"?
14:20.12pabelanger[psy]: If you are using the testsuite, look at the iax-call-basic test.  You can use it as a reference.
14:20.36[TK]D-Fenderdevdvd: Show us your actual attempt, and your actual failure
14:20.44[psy]pabelanger wont that just use the sipp program?
14:20.50*** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br)
14:21.22pabelanger[psy]: no, the iax-call-basic test uses the originate command from AMI
14:22.11notjohn[TK]D-Fender: I see data that comes in say for DID call and that's pretty easy to determine.. but if someone comes in off a menu or a transfer from reception it isn't logged as the initial call.  There will be the incoming call to the main number and then the transfer will show up as another row
14:22.12pabelanger[psy]: Actually, the originate command from the CLI
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14:22.31*** mode/#asterisk [+o putnopvut] by ChanServ
14:22.41[TK]D-Fendernotjohn: a transfer is another call.
14:22.53[TK]D-Fendernotjohn: * dialing another device in the dialplan is another call.
14:23.04[TK]D-Fendernotjohn: Every call gets a CDR (answerwed ones that is)
14:24.38notjohn[TK]D-Fender: k... I understand that now
14:27.48eject_ckI still cannot make some bulitin features working :( One touch monitor for example
14:27.56*** join/#asterisk jmkgreen (~chatzilla@fentech.gotadsl.co.uk)
14:28.07devdvdok so i think i see the why. I just dont know how to fix it.
14:28.09devdvd<PROTECTED>
14:28.09devdvd<PROTECTED>
14:28.10devdvd<PROTECTED>
14:28.12eject_ckCan someone help how to make it working ? My features sho One Touch Monitor                 *1
14:28.30eject_ckI make call and then press *1 and nothing happens
14:28.45[TK]D-Fendereject_ck: pastebin your complete failed attempt
14:28.48jmkgreenI'm getting 'File ... does not exist in any format' yet the file exists and appears to be the correct format. I'm in need of some inspiration of what to look at next
14:28.59devdvdlooks like asterisk isnt attaching the number to the interface
14:29.01[TK]D-Fenderjmkgreen: Show us the failure and your file.
14:29.12[psy]pabelanger thx i'll look into that :)
14:29.13pabelangerjmkgreen: load all format_ modules
14:29.21jmkgreen[2010-05-18 15:15:15] WARNING[6976] file.c: File voiceglue/tts/Please_confirm_that_this_is_you__Press_1_fJ5iYtBfsO-0E7LNnB56m_0 does not exist in any format
14:29.24jmkgreenthat's the error
14:29.25devdvdso instead of SIP/callwithus/123456789 im just getting SIP/callwithus-XXXXX
14:29.25[TK]D-Fenderdevdvd: No, YOU aren't telling it what device to add
14:29.29*** join/#asterisk Yon (~Yon@212.247.19.244)
14:29.36[TK]D-Fenderjmkgreen: COMPLETE....
14:29.55[TK]D-Fender[10:29]<devdvd>so instead of SIP/callwithus/123456789 im just getting SIP/callwithus-XXXXX <- NO
14:30.22jmkgreen[TK]D-Fender: what bits were you expecting, sorry?
14:30.29devdvdok? please educate me master
14:30.31[TK]D-Fender[10:28]<devdvd> -- Executing [s@agentlogin:1] AddQueueMember("SIP/callwithus-00000000",  "support") in new stack <-- this is saying that this channel is the one CALLING the app.. not a PARAMETER passed to it
14:30.40devdvdah
14:31.11[TK]D-Fender[10:28]<devdvd> -- Executing [901@default:1] Goto("SIP/callwithus-00000000",  "agentlogin,s,1") in new stack <- guess you aren't noticiing that channel is passed with EVERY app executed.
14:31.19devdvdyea
14:31.22devdvdactually i did notice that
14:31.30devdvdwhich is where i got my earlier statement from
14:31.31[TK]D-Fenderjmkgreen: COMPLETE failed call and ls dump of the file
14:31.49[TK]D-FenderdevWell again it comes down to "you aren't telling it what device to add" <-
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14:32.35devdvdi think the issue is that i expect it to work a way that it doesn't
14:33.14[psy]ah pabelanger you're of the testteam :)
14:33.59*** join/#asterisk d_preston215 (~chatzilla@static-72-86-159-138.phlapa.east.verizon.net)
14:34.03[TK]D-Fenderdevdvd: "core show application AddQueueMember" <- start by reading the instructions.
14:34.08d_preston215Morning
14:34.16jmkgreen[TK]D-Fender: http://pastebin.com/MBA5iKGh I've posted as much as I think is useful - it's bloody huge otherwise
14:34.17devdvdcuz on a trunk the interface actually is SIP/trunkname whereas with a direct attached extension (softphone, etc) SIP/exten is the interface name
14:34.37*** part/#asterisk bsaxon (~bsaxon@66.76.242.154)
14:34.42[TK]D-FenderdedvYOU did not pass the device to use as a parameter when YOU called AQM
14:35.20eject_ck<PROTECTED>
14:35.41devdvdyea, i know i didn't. thats what im getting at.  I was trying to make it dynamic. i know that i can do AddQueueMember(queuename) and that app will pick up the interface automatically
14:35.50[TK]D-Fendereject_ck: Never said you would see it normally.
14:35.51devdvdive seen it done with internal extensions
14:36.17devdvdbut i guess the way asterisk deals with trunks wont allow for that...which is fine, i just needed to know before i started trying to code around it
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14:36.26[TK]D-Fenderdevdvd: Ther is no "automatic".  YOU pass parameters to the app.
14:36.53[TK]D-Fenderdevdvd: "Deals with trunks?  Pardon?  You don't seem to understand the very basics of the dialplan.
14:37.25devdvdthen how do you explain when I do AddQueueMember(support) and then dial in with a locally connected extension that it adds that extension to the queue?
14:37.29[TK]D-Fenderdevdvd: Exten => 100,1,AddQueueMember(PUT FUCKING PARAMATER IN BETWEEN THESE BRACKETS TO TELL IT WHAT THE FUCK TO DO!)
14:37.41devdvdand support is the parameter im passing
14:37.44[TK]D-Fenderdevdvd: The the INSTRUCTIOSN to find out what you have to PASS IT
14:37.50pabelanger[TK]D-Fender: time for a beer
14:38.01[TK]D-Fenderdevdvd: That ISN'T SPECIFYING THE device
14:38.11jmkgreenpabelanger: or a loud hailer
14:38.55eject_ck[TK]D-Fender: http://pastebin.org/248198
14:40.13[TK]D-Fendereject_ck: Go prove that DTMF works from that deive with VoicemailMain or something
14:40.44geloeject_ck: you don't have w or W options in your Dial...
14:40.58[TK]D-Fendereject_ck: And it helps wo read Dial's instructions
14:41.10eject_ck[TK]D-Fender: sec
14:41.49jmkgreenSo can anyone spot the problem?
14:42.09jmkgreenI can't see it being a formatting issue as the wav files as been previously working
14:42.17eject_ck[TK]D-Fender: you are right! thank you a lot!
14:42.35[TK]D-Fenderjmkgreen: pastebin your asterisk.conf
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14:43.10pabelangerjmkgreen: And your modules.conf file too
14:43.11ManxPowerRemember everyone, buying a NexTone MSX is something you'll be regretting for the rest of your life.
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14:43.58jmkgreenok
14:44.02jmkgreenreload that pastebin entry
14:45.56[TK]D-Fenderjmkgreen: No, it gets a NEW number
14:46.39jmkgreen[TK]D-Fender: number? What number?
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14:46.50[TK]D-Fenderjmkgreen: PASTEBIN
14:46.56ManxPowerjmkgreen, the one it gives you when you save a pastebin
14:46.58[TK]D-Fenderjmkgreen: The old one doesn't get "updated"
14:47.02jmkgreenoh I'm sorry
14:47.04jmkgreenhttp://pastebin.com/fW1ttkp3
14:47.08jmkgreendidn't spot that bit
14:47.24ManxPower<PROTECTED>
14:47.40ManxPoweryou are dangerously close to being labeled a fool.
14:48.02jmkgreen<PROTECTED>
14:48.23pabelangerjmacz: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
14:48.32pabelangerconvert your .wav file
14:48.41[TK]D-Fenderjmkgreen: I saw the link... Ok, play another *-provided recording so we can verify the path
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14:50.53pabelangerjmkgreen: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql
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14:51.45jmkgreenpabelanger: The tts generator has not been changed - I'm struggling to understand how it might be spitting out the wrong format suddenly...
14:52.48[TK]D-Fenderjmkgreen: Copy an * stock recording in there and try it
14:53.09jmkgreenthat's easier said than done - this asterisk box is configured only to dial out using the tts recordings
14:53.22jmkgreenit doesn't actually have much inbound extensions :(
14:53.32[TK]D-Fenderjmkgreen: Just shocve a manual playback into your AGI
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15:00.51jmkgreenok it played vm-INBOX.gsm fine
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15:05.27[TK]D-Fenderjmkgreen: PASTEBIN
15:06.51WimmeH
15:06.55Wimmewoops :)
15:07.18jmkgreen[TK]D-Fender: http://pastebin.org/248339
15:08.21[TK]D-Fender[10:52]<[TK]D-Fender>jmkgreen: Copy an * stock recording in there and try it
15:08.26[TK]D-Fender[10:53]<[TK]D-Fender>jmkgreen: Just shocve a manual playback into your AGI
15:08.39jmkgreenoh you mean into the voiceglue/tts/ dir?
15:11.24[TK]D-FenderTHERE
15:11.43jmkgreenhttp://pastebin.org/248371
15:11.52jmkgreenit can't find it within that sub dir
15:12.20[TK]D-Fenderjmkgreen: Verify the fodlers owner as well
15:12.24[TK]D-Fenderfolder's
15:12.30jmkgreenyep it's all asterisk.asterisk
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15:12.50jmkgreendrwxr-xr-x 6 asterisk asterisk   4096 2010-05-18 10:22 voiceglue
15:13.04jmkgreenwithin that: drwxrwxr-x 2 asterisk asterisk 4096 2010-05-18 16:09 tts
15:13.16[TK]D-Fenderjmkgreen: dot eh call again with debug 10
15:15.13jmkgreenhttp://pastebin.org/248383
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15:19.01pabelangerjmkgreen: Convert your .wav files.  PCM Wav is not supported under Asterisk
15:19.01[TK]D-Fenderjmkgreen: go verify that another folder isn't being used like /usr/share
15:19.07[TK]D-Fenderpabelanger: thatr isn't it
15:19.15[TK]D-Fenderpabelanger: he copied a stock recording there totest
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15:23.08pabelangerjmkgreen: does vm-INBOX.ulaw exist?
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15:24.08slimaHi, I have some problem with dialplan, I would like to do this: When callerid isn't 100,101,102... go to: s,4: Gotoif($["${CALLERID(num)}" != "10x"]?s,4) isn't working, whats wrong?
15:24.34slimax is a problem a gues...
15:24.34paulcslima: I don't think you can use x as a wildcard character in that scenario
15:25.14paulcLook at the REGEX function
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15:28.06Corydon76-digAlso try ${CALLERID(num):0:2}
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15:39.17Warp4hi all
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15:42.50KNERDWarp4: Howdy
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16:01.50slimawhat's wrong with it: Gotoif($["${CALLERID(num)}" != "^10[0-9]^$"]?s,4)
16:01.52slima?
16:02.08pabelangerslima: not supported
16:02.28slimai give up
16:02.29pabelangerslima: look at the REGEX function
16:02.50slimahm
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16:05.50jmkgreenif you do 'core show settings' and look under Directories - do you see your sounds dir?
16:05.57jmkgreencause I don't see it here
16:06.16bent_screwdriveranyone know a good way to determine what area codes and prefixes are local for a given area?
16:07.27pabelangerjmkgreen: Because the logic does not exist in your version, it has been added to trunk
16:07.49jmkgreen[TK]D-Fender: interestingly the file works if I specify it with a full path.
16:08.26jmkgreenI'm trying to ascertain which path asterisk has for 'astsounddir'
16:08.55Knightfaljmkgreen: Vanilla install?
16:09.16jmkgreenKnightfal: close to
16:09.25jmkgreenit should be /var/lib/asterisk/sounds
16:09.52jmkgreenfiles in that dir play, files relative to that dir do not
16:10.08jmkgreenyet files within that dir, given as a full path, do play
16:10.31jmkgreensuggests the sounds dir is different, yet the "other" dir has the same files
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16:14.55jhirleyo/
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16:30.07eject_ckhow can I make delay before monitor() starts ?
16:30.38eject_ckThere is my dialpan
16:30.39eject_ckexten => _067XXXXXXX,1,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(number)}_${UNIQUEID}.wav)
16:30.39eject_ckexten => _067XXXXXXX,n,Dial(SIP/121/${EXTEN},30,tTw)
16:31.29eject_ckI'm route call to SPA-3102 and it starts recording even if called station not answer
16:31.35*** join/#asterisk RobH (~robh@wikimedia/RobH)
16:31.39eject_ckI need to set delay 10 seconds for example
16:31.43eject_ckis it possible ?
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16:34.30slimapaulc, pabelanger: I think it's good: GotoIf($["${REGEX("10[0-9]" ${CALLERID(num)})}" != "1"]?4)
16:35.54paulcslima: There you go! :-)
16:37.06paulcYou could use CUT to examine just the first 2 digits if it's 10x (where X is anything). I thought earlier it was only 101 through 104, in which case REGEX is perfect
16:37.49slimaanyway, thx for help
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16:38.55Knightfaleject_ck: If I understand you right you may want to set the b option in mixmonitor
16:39.41eject_ckKnightfal: not sure :(
16:40.03Knightfaleject_ck: Try it out :)
16:40.19eject_ckIn my scenario I'm calling to ext and it connects me with SPA-3102, then it calls to actual number
16:40.20eject_ckok
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16:47.07ManxPowerI have a new theory about FreePBX.  I believe that the ghost of Rube Goldberg possessed FreePBX developers!
16:48.08*** join/#asterisk TimeRider (~steve@109.224.131.68)
16:48.11jayteeor the guy who invented the Mousetrap game
16:49.26Corydon76-digjaytee: Um, same thing
16:51.10Corydon76-digOne of the main problems is that they sought to develop an interface around it without realizing that they could have influenced its development.  They therefore created all sorts of workarounds for bugs... and when those bugs were fixed, their workarounds broke
16:51.34Corydon76-digand then they complained vociferously that the bugfix broke their workaround
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16:56.05shaderdoes ConfBridge require a Zaptel timer like MeetMe does?
16:57.10Corydon76-digNo, it does not
16:57.26Corydon76-digbtw, they're DAHDI timers now
16:57.43shaderoh? voip-info needs some updating then
16:57.48Corydon76-digThe owner of the Zaptel trademark would appreciate you not calling it Zaptel anymore
16:57.57shaderok
16:58.04shadermakes sense
16:58.43shaderdo you use either MeetMe or ConfBridge?
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17:01.11Corydon76-digYou can use both, if you like
17:02.54shaderdoes ConfBridge have a means of finding the first available room? or does it work differently?
17:03.55Corydon76-digConfBridge does not have the set of features that MeetMe has.  You'll need to build that into the dialplan
17:04.05shaderok
17:04.13Corydon76-digI'd suggest using the groupcount feature
17:04.39shaderok. where can I find more documentation on it? I haven't seen anything about groupcount
17:04.56Corydon76-digThe GROUP() function is a core of groupcount
17:05.07*** join/#asterisk diegomad (~mad@190.146.200.120)
17:05.41leifmadsenwaves to Corydon76-dig
17:07.24shaderwaves at leifmadsen
17:07.32leifmadsenzup yo?
17:07.56shadermessin' with conference bridging
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17:26.47shaderhow long do you have to wait before playing a sound file, so that the beginning doesn't get cut off? or is that just a side effect of the softphone I'm using?
17:28.33paulcI'm playing with ConfBridge too.. vs MeetMe.. vs Konference
17:28.35paulcit's a fun day so far
17:28.52shaderpaulc: how's that going for you?
17:29.00paulcMeetMe does everything I want, I just wonder about 100s of listen only peeps with dahdi_dummi as a timing source
17:29.00shaderlike any of them?
17:29.18shaderhow do you set up dahdi_dummi?
17:29.29paulcI like how MeetMe gives me recording and DTMF to exit the conference, as well as "everyone waits till the big cheese arrives"
17:29.32shaderi.e. do you have any documentation on it I could see?
17:29.40shaderok
17:29.44paulcConfBridge gives me the "wait for the big cheese" but no DTMF
17:29.49shaderhmm
17:30.02shaderI bet you could add it
17:30.07paulcKonference gives me "exit when the big cheese quits" and DTMF in a slightly "different" manner - work-around-able
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17:30.17paulcso.. bit of a mish mash of options today :)
17:30.22shaderyeah
17:30.32shaderI'm still stuck on getting the timing device to work for MeetMe
17:30.45shaderit seems that I've been following outdated wiki instructions
17:30.53shaderthat nobody's bothered to update
17:30.55shader:(
17:31.14[TK]D-Fender"instructions"... cute..
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17:31.27[TK]D-FenderBasic DADHI isntall works just fine.
17:31.31paulcYeah - documentation is often the last bit to get consideration..
17:31.51[TK]D-FenderAnd the WIKI is ancient shit you should reference only as a last resort
17:31.57shaderok then
17:32.04shaderanything more recent I could look at?
17:32.09paulc[TK]D-Fender: Should I see dahdi_dummy in lsmod if it's loaded? I see a bunch of stuff but no specific "dummy" reference
17:32.37*** part/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua)
17:32.41[TK]D-Fenderpaulc: modprobe is run dahdi_cfg -vvv, then test
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17:37.49Knightfal~sip
17:37.50infobotsip is, like, Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP!
17:38.04Knightfal~providers
17:38.05infobotproviders is probably http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43
17:38.54KnightfalHrmm whats the command for trusted providers
17:39.08Qwellnone are trusted
17:39.10Qwellbut
17:39.12Qwell~itsplist-us
17:39.13infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
17:39.13Knightfallol
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17:42.19Ad-Hochi ppl
17:44.14SaiSomahi guys, i am trying to find an asterisk cmd that will count the number of digits in a dialed number.  google and other searches are turning up null.  I'm obviously not using the proper terms.  Any pointers?
17:44.52WIMPySaiSoma: len
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17:45.12SaiSomaWIMPy, rgr.  *sigh* so simple:).  thank you!
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17:46.29paulc[TK]D-Fender: I see "No hardware timing source found in /proc/dahdi, loading dahdi_dummy" but no dahdi_dummy specifically when I lsmod | grep dahdi - so.. I'm good? or it's not actually loaded?
17:47.35leifmadsenSayNumber(${LEN(${myVariable})})
17:47.36[TK]D-Fenderpaulc: USE IT
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17:54.29ecraneanyone know why asterisk sip invite has a line in it with 'a=silenceSupp:off - - - -'? What's with the ' - - - -'?
17:54.50KNERDAfter all this time I am just starting to use the sound files. Are all different langage file titles suppose to in English?
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17:55.07moldyhi
17:55.14[TK]D-FenderKNERD: the STOCK ones, yes
17:55.25KNERD[TK]D-Fender: okay..thanks for that info.
17:55.35[TK]D-FenderKNERD: KNERD Because the languague is used as a prefeix to finding them
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17:55.49leifmadsenall the sound files names are in english yes
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17:56.05KNERD[TK]D-Fender: yes it woul dmake it easier to  to stay consistent
17:56.09leifmadsendifferent languages are in /var/lib/asterisk/sounds/XX/ where XX is the language code
17:56.17leifmadsenen, es, and fr (english, spanish, french)
17:59.18shaderpaulc: did you get it to work?
18:01.02paulcshader: gimme a sec, got people at my desk
18:03.35Slugs_at or under?
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18:06.08moldywhen i want to move talks between different phones, do i need support on the phones for that?
18:10.57paulcLOL if only it was under ;)
18:11.06paulcmoldy: Do you mean transferring calls?
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18:12.42sprite--Are AMI connections more reliable in 1.6 vs 1.4? I am building an Adhearsion app and it is losing AMI connection. Jason Goecke advised me that it is an Asterisk issue not an Adhearsion issue. That Asterisk sometimes drops the socket.
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18:14.46moldypaulc: yes, sorry
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18:16.46*** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein)
18:17.32KNERDleifmadsen: leifmadsen: so when calling the different language (other than English) I write it as "/es/tt-allbusy" for example?
18:17.52QwellKNERD: no
18:17.52leifmadsenKNERD: no, you set the LANGUAGE() in the dialplan
18:18.29KNERDSo i have to make more than one dialplan
18:18.35leifmadsenno
18:18.56leifmadsenyou just have to determine which language you want to use based on either input or whatever other factors you need
18:19.25KNERDThat seems easy enough...thanks
18:19.29leifmadsenthe filenames are the same, so if you specify the language to be used, it will automatically use the language files for the language specified
18:19.47*** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br)
18:19.56paulcmoldy: Hmm.. Transfer is usually a phone function, because it deals with renegotiating the connection between two end points. You can do a transfer within Asterisk (using the * key typically, after specifying a T or t option in your Dial() command) but it's generally better to use the native transfer function of the phone (off a dedicated button etc)
18:20.49anonymouz666putnopvut: ping
18:20.50moldypaulc: hm, ok, thanks
18:21.08*** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d)
18:21.11putnopvutanonymouz666: pong
18:21.21moldypaulc: i am currently thinking (again) about which hardware to buy. not sure if the phones i am looking at right now support such functions.
18:22.04anonymouz666putnopvut: about that thread "utility of PLC" do you care to share how did you make the tests dropping approx 5% of the RTP streams?
18:22.56putnopvutanonymouz666: Sure, I patched main/rtp.c so that a random number is used to determine if we should send the packet or not. I can post the patch if you would like.
18:23.15paulcmoldy: What phones are you looking at? Practically all phones support transferring calls.
18:24.29moldypaulc: gigaset a580
18:24.48*** join/#asterisk dzup (dzup@unaffiliated/dzup)
18:26.42moldypaulc: my idea is: buy 6 gigaset a580 phones and 2 gigaset a580 hybrid base stations, connect the base stations to asterisk via voip.
18:27.28putnopvutoh, he quit...
18:27.42putnopvutIf anonymouz666 comes back, then http://pastebin.org/249285
18:28.21WIMPymoldy: I wouldn't recommend the gigaset. I think it's identical to the (DTAG branded) Sinus 501V.
18:28.28Qwellputnopvut: cheeky
18:28.33d_preston215Anyone used a redFONE before?
18:28.48leifmadsenputnopvut: I sent him a memo with MemoServ with your contents
18:28.53[TK]D-Fenderd_preston215: According to most of their users : avoid.
18:29.00leifmadsen<PROTECTED>
18:29.04leifmadsenfyi :)
18:29.09WIMPyThe later has some issues, like not ringing other phones any more if one of them has gone e.g. bue to empty batteries.
18:29.30moldyWIMPy: hm. is there anything you would recommend?
18:29.57d_preston215Yeah....Don't really have a choice right now to avoid it....
18:30.16WIMPymoldy: Maybe the Snom M3, otherwise I'd probably go for a ISDN DECT base.
18:30.21Qwellleifmadsen: oh burn
18:30.26leifmadsenoh burn indeed :)
18:30.44Qwell-MemoServ- Memo 2 - Sent by lmadsen, May 18 18:29:44 2010
18:30.44Qwell-MemoServ- your face is a nub
18:30.50d_preston215My boss swears by them....
18:30.50Qwellyou's trollin
18:31.14leifmadsenI don't disagree
18:32.26*** join/#asterisk digiv (~mlhess@141.214.234.28)
18:32.31moldyWIMPy: the snom m3 is above my budget. i will take a closer look at the isdn/dect models :)
18:34.44moldyWIMPy: my reason for going for voip bases was to avoid the need for multiple isdn interfaces on the asterisk box
18:35.55moldyit seems most cheap dect bases can only take 6 phones, but i would need at least 6 cordless ones plus 1 non-cordless one
18:36.02WIMPymoldy: I had the same idea, but at least I found out that you don't want the cheap ones.
18:36.04[TK]D-Fender[14:30]<d_preston215>My boss swears by them.... <- most swear AT them
18:36.15*** join/#asterisk atis_work (~atis_work@193.238.212.171)
18:36.30[TK]D-Fenderd_preston215: So if you don't have a choice... why aer you asking?
18:36.36leifmadsenI swear at my boss all the time
18:36.38moldyWIMPy: hehe :)
18:37.21WIMPymoldy: The Gigaset SIP also only takes six, ond only two simultaneous calls. (can be limited to one, yay)
18:38.17d_preston215lol.
18:38.41d_preston215I was wondering if anyone knew why this damn thing doesn't send any traffic.
18:38.54d_preston215I can use fonulator and run configs and stuff.
18:39.01d_preston215But no traffic.
18:39.09moldyWIMPy: yep, but at least i could connect those to the asterisk box using ethernet
18:39.14WIMPyI see the A580 IP also has POTS. So it's not identical to the 501V, that's IP only. But I'd fear the software is (at least) similar.
18:40.13[TK]D-Fender[14:36]<leifmadsen>I swear at my boss all the time <- enjoying being self-employed? ;)
18:40.38leifmadsen[TK]D-Fender: yep :)
18:40.57sprite--Are AMI connections more reliable in 1.6 vs 1.4? I am building an Adhearsion app and it is losing AMI connection. Jason Goecke advised me that it is an Asterisk issue not an Adhearsion issue. That Asterisk sometimes drops the socket.
18:41.40russellbdo you get errors at the asterisk console when it happens?
18:41.41*** part/#asterisk waKKu (~blah@unaffiliated/wakku)
18:42.00*** join/#asterisk datacompboy (~opera@l49-3-84.cn.ru)
18:42.12WIMPysprite--: The only thing I can tell you is that I didn't have any trouble so far. Neither on 1.4 nor 1.6, but I used it less on 1.4.
18:42.15sprite--I haven't seen any. It seems to happen overnight. So not sure how long the app actually runs before losing a connection.
18:42.29datacompboyHi! I have peer with qualify=yes. Is there any function for GotoIf[], that i can use to jump if it UNREACHABLE ?
18:42.50*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net)
18:42.50[TK]D-Fenderdatacompboy: "core show function SIP_PEER"
18:43.03[TK]D-Fenderdatacompboy: Might not have the underscore.. I forget
18:43.09sprite--I guess best way to find out will be to upgrade to 1.6 and see if I still have the issue.
18:43.40leifmadsen1.6 is ambiguous
18:43.48leifmadsenyou mean, "upgrade to 1.6.2"
18:43.55datacompboy[TK]D-Fender: yea! it without underscode, but that i needed. Lot of thanks!
18:45.24d_preston215dahdi_tool gives me nothing but RED alarms for my spans.
18:46.27datacompboyAny way to test what function returns without change of dialplan and dial?
18:48.21[TK]D-Fenderdatacompboy: AFAIK... no
18:48.31[TK]D-Fenderdatacompboy: make a little dialplan and originate it...
18:49.11*** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d)
18:49.39datacompboy[TK]D-Fender: yes... already doing so... thanks for help. now, looks, i can go to sleep... auto switching routes working.
18:50.50*** part/#asterisk datacompboy (~opera@l49-3-84.cn.ru)
18:56.55*** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com)
18:57.41idespinneranyone know offhand how hunting for a DAHDI group is done? e.g. Dial(DAHDI/g1/number) ?
18:57.59dohdyou mean in what order a free line is looked for?
18:58.04idespinneryes
18:58.15*** join/#asterisk TimeRider (steve@5ac7b347.bb.sky.com)
18:58.27idespinneror where in the code to look
18:58.47dohdhttp://www.mail-archive.com/asterisk-users@lists.digium.com/msg226642.html
18:59.19dohdI've read it yesterday at voip-info.org, but this page has the same info
18:59.31dohdor at least it answers your question :-)
18:59.35idespinnerthanks! thats exactly what i needed
19:06.41WIMPyIt's also at the top of extensions.conf by default.
19:08.17idespinnerthx WIMPy i knew i had seen it somewhere but couldnt remember
19:09.32*** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com)
19:10.12SaiSomaDo * 1.6.x support native call parking on Polycom 331/560? (using feature, not softkeys)?
19:12.04*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)
19:12.51*** join/#asterisk retentiveboy (~pdugas@69.169.199.82)
19:14.42*** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk)
19:20.48*** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk)
19:22.50TSM2i noticed that howler now has a g722 codec for asterisk but its marked as only for 1.4.26.2, any ideas where i can get later versions?
19:23.24ManxPowerask howler?
19:23.44TSM2well i was going to do that just checking if anyone here knew more about it
19:24.03ManxPowerSaiSoma, I strongly doubt it.
19:24.36ManxPowerTSM, people that need G722 use 1.6.x.x
19:24.48SaiSomaManxPower, *nod*  actually, it does:), just not one button parking.  i just figured that much out, thanks though.  trying to do a softkey now for one button parking
19:25.25ManxPowerSaiSoma, As I understand it the "server based" stuff in the polycom phones uses an XML protocol that Asterisk does not support.
19:25.55SaiSomaManxPower, rgr.  thanks
19:26.55TSM2ManxPower: its built in on 1.6 then, ile have to see if my disto can migrate to 1.6 easly without breaking FPBX and the other bits
19:27.16*** join/#asterisk retentiveboy (~pdugas@69.169.199.82)
19:31.48*** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net)
19:34.42shaderhey guys, I'm trying to create a dialplan where if an internal user dials any extension not in the dialplan, they get sent to an IVR. Unfortunately, when I try useing _X., it matches even for valid extensions
19:35.07*** part/#asterisk Trixboxer (~praju@datacenter3.supportdepartment.net)
19:35.30shaderany ideas on how to do that, that doesn't block extensions like 911?
19:36.42bmoraca_workasterisk should match based on the most exact pattern available, so "exten=>911" should be matched even when "exten=>_X." exists.  additionally, it should also match based on the order they are written in extensions.conf
19:37.14bmoraca_workif it doesn't, i will of course be flogged summarily
19:39.35shaderany idea why it wouldn't?
19:39.46*** join/#asterisk fifer (~fifer@67.208.108.228)
19:39.46*** join/#asterisk FinboySlick (~shark@74.117.40.10)
19:39.49shaderbecause that's definitely not happening
19:40.01bmoraca_workwithout logs and configs, i could only imagine.
19:40.09shaderhmm
19:40.36shaderis there a pastebin where you can upload files?
19:40.50*** join/#asterisk kotp (~vgoff@96.2.187.66)
19:40.50bmoraca_work~pb
19:40.51infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
19:41.14bmoraca_workjust upload the relevant portions: the context in extensions.conf and the console log the failed call
19:41.21shaderok
19:47.01*** join/#asterisk retentiveboy (~pdugas@69.169.199.82)
19:48.06*** join/#asterisk kerx (~kerx@72-57-179-244.pools.spcsdns.net)
19:48.35shaderhere it is: http://pastebin.com/BYb0Wkhb
19:49.21shaderthis is just a rought draft/demo dialplan
19:49.25shader*rough
19:49.33shaderbut it should still be able to dial 911
19:51.55WIMPyWhat context is the phone you used in?
19:52.05WIMPyphones?
19:52.08shaderyes
19:52.36WIMPyThen exchange the two includes.
19:53.27shaderwhy should that help?
19:54.46WIMPyAFAIK order matters on includes, which would explain your observations.
19:55.04shaderso, if I exchange the two includes, I can reach 911, but not the extensions in internal
19:55.08*** part/#asterisk FinboySlick (~shark@74.117.40.10)
19:55.08shaderhmm
19:55.11bmoraca_workyou include "internal" in your context before "outgoing" which means it processes _X. before it even sees 911
19:55.25shaderok
19:55.32bmoraca_workwithin a context, it will look for the most exact match, but you have multiple contexts here
19:55.38*** join/#asterisk jmacz (~jmacz@200.85.225.62)
19:55.39shaderah
19:55.45shaderso they aren't really "included"?
19:56.00WIMPyno
19:56.08shaderhmm
19:56.09shadernow what
19:56.15shaderany better ideas?
19:56.29kerxhi all, i wanted to confirm my bandwidth and pps calculations are done correctly for g729 audio codec
19:56.34*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net)
19:56.39WIMPyYou could create a new context for the catchall and include it last.
19:56.39kerxi have 30 calls running w/ g729 audio codec.
19:56.45leifmadseninclude order does matter
19:57.08leifmadsenso make sure the more significant matches are listed prior to the pattern matches
19:57.13shaderWIMPy: good idea
19:57.46*** join/#asterisk albasheers (~basheer@188.116.235.226)
19:57.55kerxwould this come out to 3,000 packets per second on 30 g729 calls in asterisk?
19:58.01*** part/#asterisk albasheers (~basheer@188.116.235.226)
20:02.12*** join/#asterisk albasheers (~basheer@188.116.235.226)
20:02.55*** part/#asterisk albasheers (~basheer@188.116.235.226)
20:04.44shaderwhat are the units of TIMEOUT(digit) and TIMEOUT(response)?
20:04.49leifmadsenseconds
20:04.53shaderodd
20:04.59leifmadsenyou can use something like 3.4 though
20:05.06leifmadsenwhy odd?
20:05.12shaderdoes Set(TIMEOUT(response)=10) not do what I think it does?
20:05.19shaderbecause it times out in 3 seconds
20:05.20leifmadsenI don't know what you think it does
20:05.27[TK]D-Fenderkerx: No.
20:05.30shadertime out in ten seconds if you don't do anything
20:05.46leifmadsendo anything where?
20:05.53shaderi.e. push a button on your phone
20:05.58leifmadsenwhen?
20:06.03[TK]D-FenderSOON!
20:06.04leifmadsenin an auto-attendant?
20:06.04*** join/#asterisk albasheers (~basheer@188.116.235.226)
20:06.11shaderan IVR, yes
20:06.15[TK]D-Fendershader: SHOW US
20:06.17leifmadsenI think you mean auto-attendant
20:06.22shaderthough I might be doing it wrong
20:06.29[TK]D-Fendershader: Likely
20:06.31shader[TK]D-Fender: of course
20:06.31leifmadsenbut I'd like to see the dialplan and the console output
20:06.35leifmadsenIVR != auto-attendant
20:06.35*** part/#asterisk albasheers (~basheer@188.116.235.226)
20:06.44shaderok, which is which?
20:07.24leifmadsenhttp://en.wikipedia.org/wiki/Interactive_voice_response
20:08.08shaderok
20:08.34shaderso IVRs involve reacting to the user's voice?
20:09.08[TK]D-Fendershader: PASTEBIN
20:09.15shaderI'm getting to it
20:09.19leifmadsenthat's not what it says
20:09.37leifmadsenIVR is something that interacts with an external resource
20:10.11leifmadsenaccepts data from a user and returns data back to the user. A good example of this is something like when you call a pizza place and it says, "If you would like the exact same order as last time, press 1"
20:10.26leifmadsenyou press 1, then it says, "your order will arrive in 40 minutes, thanks for calling Pizza Pizza!"
20:10.43bmoraca_workcopyright infringement!
20:10.52bmoraca_work"Pizza Pizza" is trademarked by Little Ceasers!
20:11.13leifmadsenPizza Pizza is the name of a pizza company in at least the Toronto area :)
20:11.22bmoraca_workinteresting
20:11.22*** join/#asterisk neurosys (~neurosys@69.199.183.150)
20:11.36leifmadsenhttp://pizzapizza.ca
20:11.50bmoraca_workyou crazy canadians and your lack-of-rule-following ways
20:13.01leifmadsenwe spit in the face of US patent and copyright laws!
20:13.08leifmadsenhides from the Feds
20:13.15bmoraca_workgood, someone should.
20:17.35Qwellleifmadsen: Little Caesars was dumb.  Their trademark is on "Pizza! Pizza!"
20:17.44leifmadsenhaha nice
20:18.04shaderlol
20:18.12shaderso Pizza Pizza! is ok?
20:18.18[TK]D-FenderQwell: The catch-phrase so nice they have to say it.... AGAIN....
20:18.25Qwellleifmadsen: http://en.wikipedia.org/wiki/Little_Caesars#Trademark_in_Canada  :)
20:19.37leifmadsensomeone slipped up :)
20:19.39shader[TK]D-Fender: http://pastebin.com/bqteF6QZ
20:20.00Corydon76-digleifmadsen: How long has the restaurant been around?
20:20.06leifmadsenyears
20:20.17bmoraca_worklol
20:20.19leifmadsenit's not a restaurant though -- it's a pizza delivery service
20:20.25Qwellleifmadsen: actually, looks like Pizza Pizza came before Little Caesars starting using the catchphrase.
20:20.28[TK]D-Fendershader: exten => s,n,WaitExten(5) <-takes precedence
20:20.37Corydon76-digI'd say Little Caesars has lost the trademark in canada, then
20:20.38leifmadsenweak sauce
20:20.41leifmadsenaye
20:20.42[TK]D-Fendershader: Here you explicitly overrode the channel timeouts
20:20.43Qwellso if they were to expand into the US, they probably *can* use it
20:21.07shaderah
20:21.19shaderso it shortened to 5 from ten?
20:21.31shaderbtw, does it automatically wait for the end of the audio file anyway?
20:21.54[TK]D-Fendershader: 5s from end of audio
20:22.12shaderso the timeout starts counting after the audio finishes
20:22.16[TK]D-Fendershader: Should
20:22.20shadergood
20:22.58shaderI tried fixing the syntax error mentioned in the log, but I couldn't
20:23.36shaderit seemed like it was jumping into the invalid extension macro, even though no invalid extensions were entered
20:23.49shaderam I supposed to terminate macros in a special fashion?
20:24.00shaderor is something simpler going on?
20:24.10[TK]D-Fendershader: exten => s,n,GoToIf($[ISNULL(${NUMTIMEOUTS})]?set:main) <- wrong braces to reference a FUNCTION
20:24.28leifmadsenyep, missing ${   }
20:24.44*** join/#asterisk cusco (~trilili@213.63.137.210)
20:24.45cuscohello
20:25.02leifmadsenGotoIf($[${ISNULL(${NUMTIMEOUTS})}]?set:main)
20:25.04cuscocan I have two different asterisks with the SAME queues?
20:25.07*** join/#asterisk gospch (~gospch@unaffiliated/gospch)
20:25.16cuscowe have two oficces, both with PRI
20:25.17shaderso it's supposed to be ${ISNULL(${NUMTIMEOUTS})}?
20:25.24[TK]D-Fendershader:   -- Timeout on SIP/201-0000006d, going to 't' <-- its went to "t" just like it should
20:25.39[TK]D-Fendershader: As one way, yes
20:25.44cuscoso if WAN goes down, office1 cnnot come and queue in asterisk in office2
20:25.44leifmadsencusco: if you mean have two different boxes but expect the queues on those boxes to communicate the delivery order, then no, it will not work like that.
20:26.20shader[TK]D-Fender: there's another way?
20:26.28kerx[TK]D-Fender, what should 30 simultaneous calls in queue w/ g729 codec generate for packets per sec?
20:26.29cuscoleifmadsen: so how is it done?
20:26.36leifmadsencusco: how is what done?
20:26.43kerxI thought it was ~1,500 pps based on my calculation with http://www.bandcalc.com/
20:26.45[TK]D-Fendershader: Yes.  Go read up on Asterisk Expressions.
20:26.54kerxI select the top radio box that says Payload is
20:26.56cuscoI mean.. BIG LARGE call centers have loads of asterisk boxes
20:27.06cuscodo they use the same asterisk for queuing?
20:27.12leifmadsencusco: very carefully -- you have different queues on each asterisk box
20:27.14kerxcusco, what do you consider BIG LARGE? how many agents on the floor?
20:27.29cuscoright now we expect about 300 operators..
20:27.31leifmadsencusco: you cannot share caller position across multiple asterisk boxes
20:27.37cusco100 in office1 and 200 in office2
20:27.41leifmadsencusco: they would be separate queues
20:27.42kerxok that's definitely big large :)
20:27.51cusconor queue weight
20:27.58kerxanything greater than 50 is considered big large in my opinion
20:28.11kerx[TK]D-Fender, you there? Still trying to figure out packets per sec :)
20:28.14leifmadsencusco: ok, lets make this even more general:  you cannot share queue information across physical boxes
20:28.27leifmadsencusco: they would be independent queues with their own non-shared information
20:28.34shader[TK]D-Fender: given that the wiki is a horrible reference, any other recommended place to look it up?
20:28.46leifmadsenshader: http://www.asteriskbook.org
20:28.48leifmadsenerrr...
20:28.52leifmadsenshader: http://www.asteriskdocs.org
20:28.52cuscoleifmadsen: ok ...
20:29.00cuscothanks for clearig my doubts
20:29.57[TK]D-Fendercheckout time, BBIAB
20:30.36kerxanyone can help me figure this packet per second question
20:30.41kerxi'd really apprecite it
20:31.46*** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58)
20:33.45Qwellkerframil: what is the question?
20:33.47Qwellkerx:
20:34.24kerxQwell, I've got 30 simultaneous calls in queue w/ g729 codec
20:34.36kerxI want to know how many packets per second those 30 calls together are generating
20:34.47Qwell20ms per packet
20:34.49kerxI'm using iptraf and I'm seeing something around 1,900 packets per sec
20:35.08Qwell(1000 * 30) / 20
20:35.12WIMPyThat depends on how much data you put into one packet. Codec doesn't matter.
20:35.45kerxhow do i figure that out ?
20:35.51kerxin my sip.conf i dont have that specification
20:35.53Qwell<Qwell> 20ms per packet
20:36.15WIMPy(Unless you can't put as much data into a packet as you want due to network restrictions)
20:36.35kerxwell i'm having major packet loss right now
20:36.45kerxbecause the carrier in my building i just moved to, doesnt have a good router
20:36.47Qwellchanging the size of the packets isn't going to help that..
20:36.49*** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net)
20:37.09Qwellin fact, increasing the packetization will only make the problem more audible
20:37.18kerxfor informational purposes, how do i change packet size in asterisk configuration?
20:37.51WIMPySamller packets -> mor stress for the router, bigger packets -> more impact on the audio.
20:37.52lanningWIMPy: RTP packets are small, you don't stuff as much as you can. it's about keeping a steady flow.
20:38.21bmoraca_worki believe that asterisk uses the linux network stack, which means that your MTU is set there, and not within asterisk
20:38.52Qwellbmoraca_work: your MTU isn't going to be < 50 bytes...
20:38.55WIMPyLike lanning said, you will under no normal circumstances hit the MTU.
20:39.15bmoraca_workQwell: i was just going to mention that it likely doesn't matter because your packets are much smaller
20:44.06devmodIs there any way to dynamically add/remove sip peers ?
20:44.53pabelangerdevmod: Asterisk realtime
20:46.41*** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br)
20:46.50devmodIs this still true "There is no support for NAT keep-alives"
20:50.35pabelangerdevmod: you can decrease your SIP registrations to 30 seconds
20:54.40*** join/#asterisk retentiveboy (~pdugas@69.169.199.82)
20:55.12devmodpabelanger, I guess I could do that, the problem is that i cannot always control the endpoints
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21:01.08*** part/#asterisk jplank (~GBove@208-104-67-26.dyn.fttp.comporium.net)
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21:13.52devmodis there any way to filter the events I receive through AMI? like receive only a specific type of event, etc?
21:16.37*** join/#asterisk settntrenz (~joe@137.216.121.70.cfl.res.rr.com)
21:17.35settntrenzWith the Cisco 7940 IP phone, missed calls from external #'s are correctly displayed as 10 digit (north american) numbers. The only problem with that is a "9" is required to dial that number back so trying to place the call from the missed call menu doesn't work. Any ideas for a workaround. I don't think adding a 9 in front of the CLID is the best way to solve.
21:18.20p3nguinFix your stupid dialplan so you don't have to dial a 9 before the real number.
21:18.30p3nguinIt's dumb and useless.
21:18.31WIMPyWhat else would you do, if not make the 9 unneccessary?
21:19.35settntrenzp3nguin: wish I could. Personally I agree, unfortunately the people who sign the checks don't.
21:20.32p3nguinIf some people desire to dial a 9, leave that part of the dialplan to keep them happy, but fix it so it isn't required anymore.
21:21.27p3nguinUnless you have internal extension numbers of 7, 10, or 11 digits, it doesn't make any sense to require a special code to distinguish between internal and external phone numbers.
21:22.48settntrenzp3nguin: thanks for the suggestion.
21:23.10[TK]D-Fendersettntrenz: change the callerid before calling your phones to add the 9 in front
21:29.17shaderdoes anyone have a list of standard dialing patterns to cover international calls?
21:29.22*** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net)
21:29.37Qwell.
21:29.39Qwellshader: ^^
21:30.13shaderyes, Qwell?
21:30.23Qwell. will cover international
21:30.28shaderlol
21:30.42jdoep3nguin: never got why people did that.
21:31.03Qwellshader: Qwellzakistan uses 1 digit phone numbers.
21:31.15shaderawesome
21:31.24shaderI bet you have less than 10 people living there
21:31.29shaderdo you have any spare numbers?
21:31.30QwellThere is no international regulatory body.
21:31.32shaderhow much do they cost?
21:31.35Qwellyes, 9 of them.
21:31.39*** join/#asterisk zatriz (~asdf@static-98-117-149-122.sttlwa.fios.verizon.net)
21:31.43Qwell$1,000,000/min
21:31.51shaderhmm
21:32.02shaderI think I'll buy my DIDs from a less exotic location
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21:32.17shaderthanks for the offer though
21:32.21QwellBut you could have a prime DID like...7.
21:32.28shadervery prime
21:32.39shaderbut with a price to match
21:33.37shaderbill gates himself could only afford #7 for about a month
21:33.42zatrizWondering if anyone has had any issues with a Polycom ip6000 connected to asterisk 1.4 not able to dial to pstn.  Its able to recieve pstn calls and also able to call local peers
21:34.02*** join/#asterisk retentiveboy (~pdugas@69.169.199.82)
21:34.31shaderzatriz: but the other peers can dial pstn?
21:34.47zatrizyes all the other peers are able to dial pstn no problem
21:35.00shaderare they in the same dialing context?
21:35.13zatrizbut the other peers are also cisco 7961 and yes all in the same context
21:36.27shaderdo you have a log of a call attempt?
21:36.36[TK]D-Fenderzatriz: IP 6000 has nothing to do with PSTN <-
21:36.38devmodis there any way to filter the events I receive through AMI? like receive only a specific type of event, etc?
21:36.43zatrizyeah let me find some place to upload to
21:36.44[TK]D-Fenderzatriz: it is a SIP phone.
21:36.58[TK]D-Fender~pb
21:36.59infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
21:37.05[TK]D-Fenderzatriz: ^^^^^^^^
21:37.47shadertoo bad lisppaste doesn't support arbitrary irc channels
21:37.55shaderit could have been so much more useful
21:40.11zatrizshader:http://pastebin.com/3AAU58ws
21:40.20zatrizits got pri debuging enabled as well
21:40.59[TK]D-Fender<PROTECTED>
21:41.04[TK]D-Fenderzatriz: Calling an invalid #
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21:41.33zatrizI have a capture of a cisco calling the exact same number and its going through
21:41.41WIMPy.. or incomplete
21:41.45[TK]D-Fenderzatriz: Where?
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21:43.51[TK]D-Fender<                  Ext: 1  Cause: Unallocated (unassigned) number (1), class = Normal Event
21:43.54[TK]D-FenderOccurs twice
21:43.58zatrizhttp://pastebin.com/nFZBPRi7 with cisco calling out
21:44.31[TK]D-FenderCisco = -- Executing [5642383@stations:1] Dial("SIP/desk-5878-00001117", "Zap/G1/5642383") in new stack
21:44.40[TK]D-FenderPolycom =     -- Executing [8774936@stations:1] Dial("SIP/5839-0000111b", "Zap/G1/8774936") in new stack
21:44.45[TK]D-FenderNOT the same number
21:45.21zatrizwrong paste but with the polycom i can dial any number and it will do the same thing 911 even
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21:45.43[TK]D-Fenderzatriz: So far I see one number fail and a story that doesn't match
21:46.22*** join/#asterisk timmyd (~timmyd@pool-173-79-13-149.washdc.fios.verizon.net)
21:46.30zatrizTK: What doesn't match
21:46.50[TK]D-Fenderzatriz: the PHONE NUMBER
21:47.50timmydare there any major security issues for asterisk 1.4.21.2? i'm getting scanned (e.g. [May 14 07:35:52] NOTICE[28107]: chan_sip.c:15236 handle_request_register: Registration from '"999"<sip:999@173.79.13.149>' failed for '74.115.162.15' - No matching peer found), then a little later [May 14 07:36:35] NOTICE[28107]: chan_sip.c:14035 handle_request_invite: Call from '103' to extension '0020121002828' rejected because extension not
21:48.04Qwelltimmyd: asterisk.org/security
21:48.13zatrizwould it make it any easy if i uploaded more logs that should it failing with other numbers because its failing
21:48.29[TK]D-FenderTimeRider: Do something stupid like use a purely numeric password or a dictionary word?
21:48.32*** join/#asterisk retentiveboy (~pdugas@69.169.199.82)
21:48.47timmydno, using generated random passwords
21:49.07shadertimmyd: you might want to restrict the ports on your firewall to only accept connections from your service provider, unless you're accepting connections from random other asterisk computers over the internet
21:49.27[TK]D-Fenderzatriz: You tell me the PHONE is at fault and that the same number works on your Cisco.  And you fail to show me that case
21:50.03timmydshader: i thought contactpermit=10.120.0.0/255.255.0.0 would have fixed this issue but the line must be ignored?
21:50.06[TK]D-Fendertimmyd: Mixed alpha-numeric?
21:50.24timmyd[TK]D-Fender: yes
21:50.34[TK]D-Fendertimmyd: that should be "permit", not "copntactpermit"
21:51.13[TK]D-Fendertimmyd: And yes, there are plenty of security issues.  Why do you think we're at 1.4.31?
21:51.38timmydwell ubuntu hasn't backported it so i guess i'll need to find an updated repostiory
21:51.54p3nguinAnyone know if the Cisco 7912G with SIP can do distinctive ring?
21:53.03*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
21:53.19p3nguinIt doesn't respond to Alert-Info in a SIP header like the 7960/7940 does.
21:53.21shaderhow does the directory command make use of its context? where does it get the numbers it uses to call the mailbox owners?
21:53.35zatrizTK: i never said its the phone, but it could be the phone, or some asterisk configuration
21:53.43[TK]D-Fendertimmyd: Funny... Mine says 1.6.2.5
21:54.09[TK]D-Fenderzatriz: You said "same number works from Cisco".  Are you telling me different now?
21:54.26timmyd[TK]D-Fender: you're probably not on jaunty?
21:54.30[TK]D-Fenderzatriz: Because right now all we have is your telco telling you "that number isn't valid"
21:54.51zatrizTK: http://pastebin.com/q2Lnc3KK
21:54.56[TK]D-Fendertimmyd: Welcome to Antiquity... population : YOU
21:55.23zatrizAnother number that fails from the polycom but successfully completes from the cisco
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21:55.56zatrizI've have not been able to make even one successful call out to the pstn from the polycom from the 30 cisco phones i've had 1000's
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21:56.52[TK]D-Fenderzatriz: #   -- Executing [8774936@stations:2] Dial("SIP/5839-0000111b", "Zap/G1/8018774936") in new stack
21:57.14[TK]D-Fender> Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '7178774936' ]
21:57.21[TK]D-Fender<PROTECTED>
21:57.26[TK]D-FenderBull
21:57.30[TK]D-Fender^^^^
21:58.21zatrizbecause i didn't do a search replace all on the area code ?
21:58.50[TK]D-Fenderzatriz: Don't waste our time fucking with the evidence
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22:00.17timmyd[TK]D-Fender: oh well, time to upgrade to 9.10. thanks all
22:00.21*** part/#asterisk timmyd (~timmyd@pool-173-79-13-149.washdc.fios.verizon.net)
22:00.23hardwireat this point I'm wondering if verizon made up their own phrases for all of their rate sheets.
22:00.28zatrizTK: if you can't help me then dont just because you dont understand how this could be happening. I've install enough asterisk systems to understand that this is whats happening
22:01.17[TK]D-Fenderzatriz: You tell us "this is the situation".  Then you preceed to NOT show us that this is the case.  Then you forge a new BS story.  How the hell is anyone supposed to figure out whats going on?
22:01.41[TK]D-Fenderzatriz: So far you are altering the numbers, and your telco says "that number.... we don't like it"'
22:01.46[TK]D-Fenderzatriz: There is nothing here we can trust
22:01.51p3nguinAnd tamper with the logs and expect you to solve the problem.
22:01.54*** part/#asterisk settntrenz (~joe@137.216.121.70.cfl.res.rr.com)
22:01.58[TK]D-Fenderzatriz: And you've provided nothing else to go on
22:02.36[TK]D-Fender"Hi, here's a picture of me in a clown suit....... NOW TELL ME WHAT'S WRONG WITH MY DAMN CAR!"
22:02.47[TK]D-FenderAnd in other related news...
22:02.55p3nguinafter he photoshopped his face on the picture.
22:03.08[TK]D-Fenderp3nguin: No... Someone ELSES's face :)
22:03.22*** join/#asterisk shader (~40846872@gateway/web/freenode/x-qkpoecyvbyycadbu)
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22:03.47zatrizDamn i didn't think you could be that thick
22:04.14TimeRiderfender been drinking too much peosi again?
22:04.38p3nguinI could use anice cold Pepsi right now.
22:05.26[TK]D-Fenderzatriz: What have you given us?  nothing consitent, no configs, no versions, we don't know even what telc you're using.  The fact you seem to pass of 7 AND 10 digit numbers to them in the first place.
22:05.35[TK]D-Fenderzatriz: What have YOU done to help us help you?
22:07.09zatrizAsterisk 1.4.27.1 Libpri 1.4.10.2 Zaptel 1.4.12.1
22:07.22zatrizwhich configs do you want ?
22:07.37zatrizsip.conf extension.conf ?
22:07.54[TK]D-Fenderzatriz: zatriz sip shouldn't matter.
22:08.01[TK]D-Fenderzatriz: zatriz ZAPTEL <-----------
22:08.07[TK]D-Fenderzatriz: Your PRI is rejecting you.
22:08.15zatrizYes its not running DADHI
22:08.24[TK]D-Fenderzatriz: Now * is placing that call.. and the signalling is there, and it is the other side telling you to GTFO.
22:08.50[TK]D-Fenderzatriz: zapata.conf please.
22:10.14zatrizhttp://pastebin.com/hRGgNMzK
22:10.53[TK]D-Fenderzatriz: That is not what I asked for.
22:11.35zatrizsorry hang on
22:12.40zatrizhttp://pastebin.com/PUnT1iRE
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22:12.47*** mode/#asterisk [+o Cresl1n] by ChanServ
22:13.26Qwellzatriz: use a different site...  surely you see the giant warning at the top of the page?
22:13.47[TK]D-Fenderzatriz: pridialplan=national <- change to "unknown"
22:14.58ManxPower[TK]D-Fender, odd how following the documentation (that says you almost never need to set pridialplan) is totally ignored by so many people.
22:15.16ManxPowerlike zatriz, for example.
22:17.03devmodany recommendations on a reliable asterisk manager proxy ?
22:18.06Qwellasterisk
22:18.11Qwellno need for a proxy
22:19.09devmodQwell, even if i will have multiple clients connecting to it?
22:19.16Qwelldevmod: sure
22:19.49p3nguinHahaha... like Asterisk is only capable of a single user agent connecting to it at a time, or something.
22:21.22*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
22:21.31[TK]D-Fenderp3nguin: My sweater is multi-threaded, but only fits one user!  HALP!!!!!!!!!!!!
22:21.43p3nguinlol
22:21.55Qwell[TK]D-Fender: little known fact - sweaters are single-threaded.  it's just a really long thread.
22:22.37[TK]D-FenderQwell: LIES
22:22.50[TK]D-FenderQwell: I *know* the arms were separate!
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22:34.23ManxPowerIs devmod still asking silly questions?
22:34.25*** part/#asterisk albasheers (~basheer@188.116.235.226)
22:34.43devmodManxPower, yup
22:39.37hardwirewhat would you guess "switchless 1+ service" would mean?
22:41.52hardwiredoes it simply mean interstate trunk since you don't need to originate on the PSTN?
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22:45.30drmessanoDid someone say something about arms?
22:45.51drmessanoBecause, you know, you can't hug with nuclear arms.
22:46.24[TK]D-Fenderdrmessano: You can... but it tends to make the other parties nervous :)
22:46.33drmessanolol
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22:47.51drmessanoI need a pet shock collar that works with X10..
22:48.04drmessanoNeed to be able to shock my cat via Asterisk
22:48.06ManxPowerI need one that work ON XO
22:48.10p3nguinClick a key on the computer, kill a dog.
22:48.24[TK]D-FenderDON'T TASE ME BRO!
22:48.28drmessanoHAHAH
22:48.40ManxPowerI'd love to ZZZZTTT! sales reps when they report a problem with a call but don't include any other information
22:48.49drmessanoI just spit my drink.. brb
22:49.31drmessanoWell timed "Don't tase me bro"..lol
22:50.42ManxPowerOr verizon!  Zap!  What do you Zap! mean the customer Zap! was not there?  Zap!  They were Zap! there all day!  Zap!
22:51.51[TK]D-FenderManxPower: REMINDS ME OF THE OPENING SCENE TO gHOSTBUSTERS
22:51.55[TK]D-Fenderdarn caps
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23:46.22riddleboxhello, if I have a sip endpoints as something like Tom, Brian,etc... then in my extensions.conf I set global variables as 110 => SIP/Tom, is there an easy way to say if someone dials a _1XX dial the global variable? or am I going about it wrong?
23:49.47ChannelZhttp://dlisted.com/node/37318
23:49.53WIMPyriddlebox: Interesting Idea. It's possible, yes.
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23:52.54riddleboxWIMPy: I am sure its possible but now my dialplan is exten => _5XX,1,Dial(SIP/${EXTEN},20)
23:53.17riddleboxoops thats supposed to be a 1XX
23:53.54bmoraca_workyou need dynamic variable names
23:54.41WIMPy${${EXTEN}}
23:54.45bmoraca_workyep
23:55.09[TK]D-FenderLast I heard variables had to start with a letter
23:55.15bmoraca_workit does
23:55.28bmoraca_work${PEER_${EXTEN}}

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