IRC log for #asterisk on 20120802

03:12.56*** join/#asterisk infobot (~infobot@rikers.org)
03:12.56*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.7.0 (2012/07/30), 1.8.15.0 (2012/07/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
03:13.05*** join/#asterisk ritztech (446c874c@gateway/web/freenode/ip.68.108.135.76)
03:13.13ChrisInSydneylooks like someone made it work with a SPA3102 on the FXO port
03:13.30ChrisInSydneyThinkGNU-
03:13.58ritztechanyone know why when in asterisk cli  that commands  dahdi is not appearing (like dahdi show channels) and such
03:15.28kaldemarritztech: chan_dahdi.so is not loaded.
03:15.41ritztechi loaded it like 4 times
03:15.54ritztechreloaded processes a couple
03:16.46ritztechSOB ive been trying to restart modules and asterisk processes for the past hour and a final reboot fixed it  ..... fyi i was doing a service stop dahdi service stop asterisk and /etc/init.d/asterisk stop / start etc
03:16.55ritztechbut a good ol rebooty tooty worked
03:18.01ThinkGNU-I just saw your message ChrisInSydney
03:18.22ChrisInSydneycool
03:19.18ThinkGNU-thanks :-)
03:21.08ChrisInSydneymight help
03:21.26ChrisInSydneyI'd better get back to what I am suposed to be doing
03:22.09ChrisInSydneymanaged to not pay attention and brought across a live system config onto a dev machine. Forgot to disable the SIP registrations
03:22.11ChrisInSydneyoops
03:22.19ChrisInSydneyonly 1 call in
03:22.26ChrisInSydneythen  I notied
03:23.10ChrisInSydneys/notied/noticed/
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03:56.30*** join/#asterisk AliciaK (~AliciaK@142.167.221.85)
03:56.49ChrisInSydneyhi all, was I haloucinatiing, but I thought I read somewhere about choosing a codec for outbound calls in the dial plan ?
03:57.10ChrisInSydneyis that possible in 1.8.1x.x ?
04:04.11*** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg)
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04:35.28*** join/#asterisk lorsungcu_ (~anonymous@65.103.31.37)
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04:55.05ThinkGNU-I GOT IT!!!!!
04:55.11ChrisInSydneywoohoo
04:55.13ChrisInSydneyhow
04:55.14ChrisInSydney??
04:55.19ThinkGNU-damn thing!
04:55.48ThinkGNU-I had to tell the ata that the LineInUseVoltageThreshold was 20 instead of 25
04:56.07ThinkGNU-I'm going to ramp it up again until it doesn't work, just to see where the breaking point is
04:56.51*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
04:57.37ChrisInSydneyso I was sort of useful with the voltage thingy
04:57.41ChrisInSydney:)
04:57.42ThinkGNU-YES!
04:57.57ThinkGNU-It took me a while to really pin it down in my configs
04:58.08ChrisInSydneyanalog is a bitch
04:58.15*** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net)
04:58.22ChrisInSydneyI'm slack and use SIP endpoints
04:58.27ThinkGNU-It sure can be
04:58.31ThinkGNU-it's good to know how to use though
04:58.34ChrisInSydneymakes for less head aches
04:58.44ChrisInSydneyhelpful if you can justify the effort
04:58.50ThinkGNU-you never know how much antiquated stuff you might end up working with at one point or another
04:59.08_Corey_ThinkGNU-: Congrats...  :)  Analog stuff is definitely harder than SIP
04:59.20ChrisInSydneyI swing between both cheap but time consuming and hand over the cash
04:59.31ChrisInSydneyit depends on how many I install
05:00.06ChrisInSydneyI hardly ever do pagers so I just use SIP, whereas I have hacked away at DD-WRT to do many thiings you twould think it couldnt
05:00.07ThinkGNU-Well, I've spent a fair amount of time rewiring our buildings for this system to go in this weekend and I wasn't about to let this old pager set me off schedule
05:00.19ChrisInSydneyhe he
05:00.21ThinkGNU-dd-wrt is pretty awesome
05:00.50ChrisInSydneyI got caught today. A 30 pair tie between two buildings on opposite sides of the street.
05:01.02ThinkGNU-nice
05:01.06ThinkGNU-gotta love those
05:01.24*** join/#asterisk mintos (mvaliyav@nat/redhat/x-tsidwhakqsclwlrv)
05:01.24ChrisInSydneywe had a pair of Cisco 878s running a 2Mb link over a pair. I upgraded the link to 2 pairs = 4.6Mbps
05:01.42ChrisInSydneybut the lines were a little dodgy and the quick disconnect shorted
05:01.48*** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net)
05:02.01ChrisInSydney= no phones for around half an hour in the sales dept
05:02.08ChrisInSydneynot happy jan
05:02.11ChrisInSydney:-(
05:02.50ThinkGNU-We have some CAT 3 going bewteen two buildings that wouldn't make an ethernet link with anything but this old 3com switch. I tried everything but couldn't get it to link with any other hardware. just too long of an appearent loop distance
05:02.54ChrisInSydneyIt took me 26 minutes to realise what happened, find a single reliable pair and reprogram back to 2.3Mbps single pair
05:03.10ThinkGNU-I just did a wireless point to point link for mine
05:03.11ChrisInSydneyI had to cross a 3 lane dual carrage way 4 timies to fix
05:03.14ChrisInSydneyduring peak hour
05:03.32ThinkGNU-the fun things you run accross huh?
05:03.38_Corey_so, ChrisInSydney is in Sydney I presume...  ThinkGNU- where in the world are you?
05:03.49ChrisInSydneyif you want to run VoIP and you have existing runs of Cat3
05:03.58ChrisInSydneyyou can back to back 2 SHDSL modems
05:04.23ChrisInSydney_Corey_: Not only certifiied in Asterisk, but observant as well ;-)
05:04.32ChrisInSydneydamn i key
05:04.32ThinkGNU-_Corey_ I'm in Wyoming, USA
05:04.38_Corey_ahh
05:04.38ChrisInSydneycool
05:04.46ChrisInSydney_Corey_ yuourself ?
05:04.53_Corey_I'm in the Philadelphia area
05:04.57ThinkGNU-I was going to do two modems but the wireless is actually point-to-multipoint
05:05.09ThinkGNU-so I can hit a third building that's never had service
05:05.14_Corey_just up late watching today's TiVO haul of the olympics ;)
05:05.18lorsungcu_gnu, what mfg for your radios?
05:05.31ThinkGNU-Ubiquiti
05:05.40lorsungcu_good stuff
05:05.53lorsungcu_i use mikrotik routers on all my deployments
05:06.00ThinkGNU-Those are pretty good too
05:06.09lorsungcu_wonderful things
05:06.12ThinkGNU-I was impressed to find out that the old canopy engineers are working on the Ubiquiti stuff
05:06.23lorsungcu_yes that air fiber is impressive
05:06.32lorsungcu_i need 6k and an excuse to buy some..
05:06.45_Corey_lorsungcu_: "The Dude" eh?
05:06.57lorsungcu_yes :)
05:07.00_Corey_lol
05:07.06_Corey_damn nice gui
05:07.06ThinkGNU-the dude? What am I missing here?
05:07.13lorsungcu_really?
05:07.18_Corey_it's been a few years since we've used it
05:07.35lorsungcu_i haven't found anything quite like it
05:07.57_Corey_yeah, they haven't done a really good job promoting it
05:08.12lorsungcu_maybe its just in the US, but yeah
05:08.13_Corey_i found it about 5 years ago in "linux journal"
05:08.30lorsungcu_found it after i'd already bought a rb 450 heh
05:08.39lorsungcu_pretty sweet bonus
05:08.57lorsungcu_$100 router does gigabit, mols, ppp
05:09.06lorsungcu_+awesome management?
05:09.10lorsungcu_over many devices?
05:09.16lorsungcu_can't beat it.
05:09.40_Corey_we found it was pretty abusive from a bandwidth perspective
05:09.56lorsungcu_you can tone it down.  its most nice for upgrades
05:10.25lorsungcu_don't do any wireless, but I can imagine it'd be good for checking on links, etc.
05:10.36_Corey_we had some tunnel issues when it was running, we should revisit though...  it's been a while
05:11.14lorsungcu_what sort of tunnels?
05:11.23ThinkGNU-Well, it's late here and I have to be up in about six hours to continue on tomorrow morning. Thanks again guys!
05:11.58*** join/#asterisk Sean-Der (~sean@cpe-68-175-54-64.nyc.res.rr.com)
05:12.05_Corey_ThinkGNU-: good luck
05:12.21Sean-DerI have tried Set(${LANGUAGE}=te) Set(${LANGUAGE()}=te) and a couple others :/
05:12.35lorsungcu_wat
05:12.43Sean-DerI can't seem to get at the language global variable!
05:12.54ThinkGNU-Thanks _Corey_
05:12.57*** part/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210)
05:13.29kaldemarSean-Der: Set(CHANNEL(language)=te)
05:13.48ChrisInSydney<Sean-Der> SET(LANGUAGE=te)
05:14.29ChrisInSydneyuse ${VARIABLE} when you want to look at the contents such as NoOp(${LANGUAGE})
05:14.31Sean-Derkaldemar: Thank you! That worked perfectly
05:14.47lorsungcu_+1 kaldemar
05:14.57ChrisInSydney-1 me :-/
05:15.04Sean-DerChrisInSydney: +2
05:15.11ChrisInSydneythanks
05:15.12_Corey_lol
05:15.13Sean-DerThank you for trying to help!
05:15.35ChrisInSydneyits 3:15pm here I have no real excuse
05:15.41Sean-DerIf only Asterisk had PyDoc or something, I find myself wandering so much
05:15.46ChrisInSydneyexcept that I am trying to work
05:16.00kaldemarthe LANGUAGE variable was deprecated in 1.4 already.
05:16.08_Corey_ChrisInSydney: Do you do telecom stuff where you work?
05:16.13ChrisInSydneyfor myself
05:16.24_Corey_ah
05:16.26Sean-DerChrisInSydney: Tell your boss that if you don't contrib Asterisk will fail... you need to help out on IRC
05:16.28ChrisInSydneySmall - mid size
05:16.42ChrisInSydneyI remind myself every day
05:17.19_Corey_Sean-Der: well, we wouldn't want to portray Asterisk as some fragile thing not "business grade" would we?
05:17.26ChrisInSydneyI used to post on the forums, but I havent done so for a couple of years now
05:18.16_Corey_what forums?  you mean the mailing lists?
05:18.18Sean-Der_Corey_: I have given up trying to convince people to use Free software at this point. People either get it or they don't it seems
05:18.37ChrisInSydneyhttp://forums.asterisk.org/
05:18.44_Corey_ah
05:19.13ChrisInSydneythat way, if its in the forums, it sticks around, and I can always go back and find what I forgot about
05:19.28ChrisInSydneyreminid myself of how clever i "Used" to be
05:19.35ChrisInSydney:/
05:19.39Sean-DerI have one boss that takes my advice and it works, the other DEMANDS that some overstarched soul-less company is behind it.
05:19.41_Corey_Sean-Der: well, open source has consistently represented 18% of the north american telephony market for the past 3+ years
05:19.58_Corey_Sean-Der: Asterisk represents 80+% of that
05:20.02ChrisInSydneySean-Der: I had that same argument with my recent partners
05:20.11Sean-DerChrisInSydney: I cry everytime I see my old forum posts... they are so bad
05:20.15_Corey_that's more than Cisco, NEC, Mitel, etc
05:20.32ChrisInSydneywouldnt listen. Ended up sneaking some FOSS on a few jobs. No issues what soever
05:20.41_Corey_if people are dismissing open-source or Asterisk, they just don't get it
05:21.18_Corey_I have Fortune-500 and ivy-league University customers running Asterisk
05:21.24ChrisInSydney_Corey_: And the Open Source part has been critical in augmenting some of the features for commercial deployments
05:21.44_Corey_scratch the F on FOSS
05:21.59_Corey_open-source is many things...  free is not one
05:22.05_Corey_FLEXIBLE is the keyword
05:22.08ChrisInSydney_Corey_ True, its never ever "free"
05:22.21ChrisInSydneyit always comes at a cost
05:22.24Sean-Der_Corey_: Asterisk has gotten a lot more power behind the brand, but people who have been out of the industry for a little while still only accept Cisco, IBM, Mircosoft. At home I am an adamant Free Software guy ... at work I lax
05:22.31ChrisInSydneymine si tmie with my family
05:22.47Sean-DerMake sure not to release under BSD or they will just steal it all :p
05:22.52ChrisInSydneyso put Ciisco handsets on tehir desks and switches in their racks
05:23.14_Corey_Sean-Der: IBM gave a keynote 2 years ago at Astricon about their choice of Asterisk on their SMB appliances
05:23.24ChrisInSydneys/ so put Ciisco handsets on tehir desks and switches in their racks/ so put Cisco handsets on their desks and switches in their racks/
05:23.46ChrisInSydneys/so put Ciisco handsets on tehir desks and switches in their racks/so put Cisco handsets on their desks and switches in their racks/
05:23.50_Corey_Cisco switches are fine... stay away from their handsets
05:24.01lorsungcu_yeah...
05:24.03ChrisInSydneydepends on the model
05:24.09_Corey_they haven't supported SIP well since... um, like 2007
05:24.12lorsungcu_had a customer that wanted ~400 handsets moved to SIP
05:24.17Sean-Der_Corey_: And * is only gonna grow from there.
05:24.18lorsungcu_this was last year
05:24.31ChrisInSydneyDone a heap of SPA525G2s and SPA504s for a client. Thats what they wanted
05:24.33_Corey_unless you're talking about the SPA stuff, which has suffered since the Cisco acquisition
05:24.41lorsungcu_these were all 7940
05:25.02_Corey_well, if they have the investment in the 79xx stuff it is what it is
05:25.08lorsungcu_its crap, either way
05:25.17_Corey_I have a customer that has a few hundred endpoints on it
05:25.17ChrisInSydneyGot a client next fortnight wit around 30 7971G-GEs and another 40 or so 7940s
05:25.25Sean-DerLuckily I don't have to do any hardware side stuff :) I enjoy being a software developer.. I really should checkout the Asterisk repo, I doubt I could do much though :(
05:25.30ChrisInSydneyfound a box of 7912s as well they gave me
05:25.39lorsungcu_7912 is a bitch
05:25.44_Corey_7912s don't really support SIP well
05:25.45ChrisInSydneyreally ?
05:25.47lorsungcu_and 7905
05:25.48Sean-DerThe biggest C project I worked on was Fish-shell
05:26.29_Corey_yeah, the 7940s and 7960s are the only Cisco "enterprise grade" handsets that support SIP
05:26.38lorsungcu_support is used loosely.
05:26.45_Corey_7941s and 7961s are VERY sketchy
05:26.48lorsungcu_they _do_ make and receive calls.
05:26.49ChrisInSydneySean-Der: its not too hard to tweak things, such as back ports and upgrading patches to newer versions
05:27.14_Corey_lorsungcu_: No, they're not bad...  comparatively speaking
05:27.21ChrisInSydney_Corey_ lorsungcu_ The 7971s are working now on SIP
05:27.40lorsungcu_yes the 79x1 works on sip for a while
05:27.42ChrisInSydneyI did play with chan_sccp-b a while ago
05:27.50Sean-DerExcuse my ignorance, but what is Cisco pushing with their handsets? I have only ever encountered IAX*, RTP & SIP
05:27.55_Corey_lorsungcu_: The features are limites on the 79x1
05:27.57lorsungcu_but users expecting cisco proprietary functionality on cisco sip firmware are always disappointed.
05:28.02ChrisInSydneythey worked better on those
05:28.14ChrisInSydneySean-Der: skinny
05:28.25_Corey_Sean-Der: well, the world is moving to SIP and Cisco gets that
05:28.29ChrisInSydney= not SIP = cisco proprietory
05:28.42ChrisInSydneybut they are moving to "Cisco SIP"
05:28.44_Corey_they're still not 100% there with their Call Manager deployments
05:28.58ChrisInSydney= wont work properly with Asterisk SIP handsets
05:29.17_Corey_but the SPA series (formerly Sipura) stuff is all designed for SMB and is SIP
05:29.21ChrisInSydneytrue
05:29.22lorsungcu_and pretty decent
05:29.34_Corey_eh
05:29.41_Corey_I've found them to be buggy
05:29.48ChrisInSydneyreaonably. Mind you we just ripped out 60 faulty 525G2s which randomly lock and reboot
05:29.50lorsungcu_yes, but for $80, you can get a good handset.
05:30.06Sean-DerChrisInSydney: Do they have a pretty open bug tracker and the such? I should check out the code and close a few bugs to get involved. Even if its stupid stuff (docs)
05:30.08ChrisInSydney525G2s are a little more than $80
05:30.08_Corey_lorsungcu_: Yeah, you can get Digium phones
05:30.12lorsungcu_bleh
05:30.16lorsungcu_not a fan
05:30.30*** join/#asterisk SQS (~SQS@94.20.72.138)
05:30.44ChrisInSydney_Corey_ lorsungcu_: +1 Snom
05:30.52lorsungcu_yes
05:30.59lorsungcu_370 is my main bitch
05:31.03_Corey_The Digium phones stack up better than Polycom or Snom and cost 20-30% less
05:31.16_Corey_plus, they offer TONS of extra Asterisk features
05:31.35ChrisInSydneySean-Der: Not sure, I just play with the released stuff, haven't published my "mods"...yet
05:31.45ChrisInSydneynot too sure how to present the .patch file
05:31.55lorsungcu__Corey_: someday ill find time to learn how they work
05:32.08lorsungcu_until then, i'll keep using the tons of other phones that _all work the same_
05:32.23_Corey_lorsungcu_: They have an XML spec that's pretty sweet, and then there's the DPMA...
05:32.36lorsungcu_yeah, i've got a few here
05:32.52lorsungcu_had a short look at them, just don't have time to get into it
05:32.57lorsungcu_until i do, i like my snoms :)
05:33.02_Corey_Don't forget...  80% of the Asterisk development is done by Digum on its own $...  let's try to support them when we can
05:33.10lorsungcu_i will
05:33.28_Corey_especially when they actually make a better product than their competitors!
05:33.38lorsungcu_but the way they handled the phones bugs me a bit.  pm me if you want to hear it, not going to get into it out here..
05:33.42_Corey_that's how we keep Asterisk free at the end of the day
05:34.31coppiceDigium is left to do a lot of the development because they actively drove other developers away
05:36.01Sean-Dercoppice: What did they do?
05:36.38coppicevarious things. leaving contributions to rot has been one of their more popular moves
05:37.32Sean-DerThats sad..
05:39.51ChrisInSydneyThere have been some nice patches such as the codec negotiation patch from the Sippy people that would have been nice
05:40.32ChrisInSydneyAs I always say, if you dont like how an ope source project is run, you can always go and "fork yourself"
05:40.35ChrisInSydney:D
05:41.14coppicefreeswitch is largely developed by people who wrote a lot of asterisk, but finally got pissed off and moved on
05:43.17Sean-Dercoppice: Is freeswitch a fork or a rewrite?
05:43.54Sean-DerThey use git... that might be enough reason for anyone to favor it haha
05:43.56coppicethere is no common code in freeswitch, other than things both projects absorbed from elsewhere
05:44.10_Corey_Freeswitch is another product altogether
05:44.17ChrisInSydneyThe anglican church was created by Henry VIII because he finally got pissed off with the catholic church and moved on
05:44.50coppicehe forked. he didn't start from scratch
05:44.55_Corey_well, Catholicism still has more souls
05:44.57_Corey_;)
05:44.59ChrisInSydneythats right
05:45.14ChrisInSydneyI thiink it was his forking around that pissed off the pope
05:45.20Sean-DerAnd in the end they all killed each other over a mystical man in the sky.....
05:45.36ChrisInSydneyat least that is what catholic historians would like us to believe
05:45.58Sean-Derhaha
05:46.02ChrisInSydneySean-Der. No they killed eachother over money, land weath and power
05:46.08coppicemost of europe was pissed off with the pope at that time
05:46.12ChrisInSydneythey just used God as an excuse
05:46.44_Corey_Well, anyhow, Asterisk (and Digium)  has a community, a support infrastructure, as well as great annual conference...
05:46.45Sean-Dercoppice: Sorry for bothering you so much, but how different are the paradigms of VoIP in Freeswitch compared to Asterisk?
05:47.24Sean-DerIt is pretty easy to jump back and forth, or did they attack problems differently?
05:47.36coppiceoooh, you're just trying to start some rants :-)
05:47.47ChrisInSydney_Corey_: True. You heading south to Astricon this year ?
05:48.07ChrisInSydney+1 coppice
05:48.29_Corey_ChrisInSydney: yeah, I'm doing a session
05:48.45ChrisInSydneySean-Der: More so than OpenSIPS v Kamailio
05:48.58ChrisInSydney_Corey_ cool, what on >
05:48.59ChrisInSydney?
05:49.25ChrisInSydneySean-Der: Just threw that one in for you ;-)
05:49.49Sean-Dercoppice: hahha maybe I am... No just curious I have fooled around with stuff like (Bluebox?), but Asterisk is the one I always come back to
05:49.57Sean-DerChrisInSydney: Thank you :p
05:49.57lorsungcu_Sean
05:49.58lorsungcu_http://freeswitch.org/node/117
05:50.31_Corey_ChrisInSydney: It's a talk about a case study of a large call center running asterisk... definitely worth seeing if you'll be there :)
05:50.52ChrisInSydneybit of a haul, but I'll see what the $$$ are like
05:51.24ChrisInSydneynot too good right now, and I have a few family things to do in Oct / Nov
05:51.46ChrisInSydneyholiday budget and my training budget is already spent
05:52.03Sean-DerI wish I could go to AstriCon! Maybe next year
05:52.03ChrisInSydneywould like to go though, catch up with some of the VUC folks too
05:52.29ChrisInSydneySean-Der: http://vuc.me
05:52.50*** join/#asterisk salz212 (~chatzilla@203.215.176.22)
05:53.05ChrisInSydneyoh yeah, and a house to repair
05:53.27ChrisInSydneythats the discressionary spend gone
05:53.49Sean-DerWell its 2 AM here, so I have to get up for work in 5 hours sighhh.. Thanks for the link ChrisInSydney I am looking it right now
05:54.30ChrisInSydneydownload some of the podcasts, there is some good stuff there, Asterisk, Freeswitch, YATE, SER/Kamailio, SIPS
05:54.55ChrisInSydneyplus heaps of good VoIP / SIP / online communication stuff
05:55.05Sean-DerCool! Thanks
05:55.06ChrisInSydney(shameless plug)
05:55.44ChrisInSydneyI must also get back to it.
05:56.01ChrisInSydneycheers all
05:56.12lorsungcu_later
05:56.29ChrisInSydneyone thing before I go
05:56.36ChannelZBABABOOEY
05:57.09ChrisInSydneya post from before:
05:57.13ChrisInSydney<ChrisInSydney> hi all, was I haloucinatiing, but I thought I read somewhere about choosing a codec for outbound calls in the dial plan ?
05:57.13ChrisInSydney<ChrisInSydney> is that possible in 1.8.1x.x ?
05:57.40ChrisInSydneysave the translation when calling out on g711 but internally on 722
05:57.49ChrisInSydneyanyone ?
05:58.04ChrisInSydneyHi ChannelZ
06:00.04kaldemarChrisInSydney: there are variables for it in chan_sip.
06:00.33ChrisInSydneykaldemar: So set the variable, then dial ?
06:00.48kaldemarChrisInSydney: SIP_CODEC and SIP_CODEC_OUTBOUND
06:00.51ChrisInSydneyhttps://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
06:00.55ChrisInSydney:D
06:01.03ChrisInSydneyill play and report back
06:01.06ChrisInSydneythanks
06:01.54ChrisInSydneyI dont care what they say about you on #freeswitch. Youre alright :D
06:02.18kaldemarsay what?
06:02.40ChrisInSydneyhe he he he
06:03.23ChrisInSydney;-)
06:09.35*** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net)
06:19.02ChrisInSydneyhi, a quick "best practices" question
06:19.32lorsungcu_go on
06:20.06ChrisInSydneyusing the '-' character is discouraged in peer names, what about context names. ?
06:20.15ChrisInSydneyI am tidying up legacy code
06:20.19_Corey_tmm
06:20.21_Corey_umm rather
06:20.27_Corey_it's pretty common in context names
06:20.42_Corey_FreePBX uses it extensively
06:21.13ChrisInSydneythey do too, and we all know what they say about kaldemar ;-)
06:21.30ChannelZI use them
06:22.03ChannelZ[handle-incoming]  and  [fax-in] etc
06:22.08_Corey_Yeah, I'm not sure what you mean but don't let the FreePBX thing throw you then... they're fine
06:22.31ChrisInSydneyI was using them in peer names, which made for a lots of changes to code to cope with them being used asa seperator character
06:23.03ChrisInSydney_Corey_ A previous post when kaldemar helped me
06:23.15_Corey_ah
06:23.26ChannelZhmm I use them in my peer names too
06:24.01ChrisInSydneyZ: II find it buggers up some reports and packages that manage channels and CDRs
06:24.15ChrisInSydneys/II/I/
06:24.55ChrisInSydneys/II/I Must get my keyboard fixed as well/
06:25.20ChrisInSydneyall cool
06:25.22ChrisInSydneycheers
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06:31.41kaldemarChrisInSydney: make up your mind already. freeswitch of freepbx? :)
06:31.55ChannelZFREEBIRD!
06:32.09ChrisInSydneyANGRYBIRD
06:33.10ChrisInSydney+1 <kaldemar>
06:33.34ChrisInSydneyFREEBEER (my favourite)
06:33.43kaldemarfreebeard!
06:36.05ChrisInSydneyits the beard without the mo' that has always creeped me out ever since I was a kid
06:37.24ChrisInSydneyfound something with dialplan logic and pattern matching...
06:38.24ChrisInSydney... if you have a pattern in a context and an included context with an exact match, the pattern in the original context is used first. It ignores the exact match in the included context
06:38.42ChrisInSydneyeg [context1]
06:38.50ChrisInSydneyinclude => context2
06:39.18ChrisInSydneyexten => _123X,1,Playback(tt-monkeys)
06:39.24ChrisInSydney[context2]
06:39.43ChrisInSydneyexten =>1234,1,Playback(tt-weasels)
06:39.49ChrisInSydneyit plays monkeys
06:39.58ChrisInSydneyast 1.8.13.something
06:41.42ChrisInSydneyit doesnt matter if the include is before or after the pattern
06:42.17lorsungcu_hmm
06:42.35lorsungcu_i think that is on purpose
06:43.41kaldemarChrisInSydney: that's how the order works. when in a context, extensions are matched first, then includes and switches.
06:43.59ChrisInSydneyits a bit of a pain. i am trying to separate out different tenant's inbound numbers, but I want a catch all if I miss one
06:44.14ChrisInSydneyeach tenant's inbound is in its own file
06:44.23kaldemarChrisInSydney: and includes in the order they appear in the context, they are not handled as a single entity.
06:44.34ChrisInSydneykaldemar: thanks
06:45.07ChrisInSydneya coffee and a short rethink is in order
06:45.20ChrisInSydneyback to it
06:47.10*** join/#asterisk Maliuta (~nobusines@59.167.214.92)
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06:53.09*** join/#asterisk ahmadhasssani (~ahmad.2.h@203.130.22.202)
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06:53.34schmidtsgood morning
06:53.41lorsungcu_hi
06:54.07ahmadhasssanigudmorning
06:54.26ahmadhasssanii am preparing asterisk server 1.4
06:54.45ahmadhasssanican somone guide me
06:54.52ahmadhasssanicoz i am tottaly new
06:54.55ChrisInSydney~book
06:54.56infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
06:56.56ahmadhasssanihello
06:57.05ahmadhasssanican someone plz guide me
06:57.08ChrisInSydneyahmadhasssani: There are a heap of hello world examples to get Asterisk going. Check out the googles. The book is an excellent place to start. There are also some Pakt publications wich walk you through step by step
06:59.15ahmadhasssanii am setting up a server
06:59.24ahmadhasssani<PROTECTED>
06:59.37ahmadhasssanii have downloaded all pakes
06:59.46ahmadhasssani*pakages
07:00.05ahmadhasssaniusing centod 5.5
07:01.05lorsungcu_why 1.4?
07:02.05kaldemarahmadhasssani: you might want to reconsider the version: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
07:02.17ahmadhasssaniany one there
07:02.18ahmadhasssani?
07:02.48ahmadhasssanii am using 1.4.13
07:02.58ahmadhasssaniasterik 1.4.13
07:03.11ahmadhasssanineed help in insttalation
07:03.35li3pwhat's problems?
07:03.59ahmadhasssaniproblem is that when i insttal libpre pakage
07:04.11ahmadhasssaniit says permission denied
07:05.03ahmadhasssanilibpri-1.4.11.5
07:05.50ChrisInSydneykaldemar: I put the catchall into an included context. If you cant beat them, driink them under the table after
07:06.18li3pyou install the program as root?
07:06.29rolandow[Aug  2 09:05:44] WARNING[6237]: app_dial.c:2444 dial_exec_full: Invalid timeout specified: '-1'. Setting timeout to infinite
07:06.34rolandowhow can i set this properly?
07:06.38rolandowwill 0 be infinite as well?
07:07.20ChrisInSydneyouch !!! I just core dumped on a dialplan reload
07:07.38ChrisInSydneyAsterisk 1.8.13.0
07:07.48lorsungcu_Chris, i think it just drank you under the table.
07:08.01ChrisInSydneylast drinks
07:08.24ChrisInSydneyactually I had a few at the pub the other night
07:08.37ChrisInSydneygot home, tapped the wife on the shoulder.
07:08.57ChrisInSydneyShe said "You only want to have sex with me when you are drunk"
07:08.59ChrisInSydneyI replied....
07:09.11ChrisInSydney"no, sometimes when I am drunk, I want a kebab
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07:09.19gg608fhaha
07:09.39lorsungcu_love kebab
07:09.43lorsungcu_sober too..
07:09.43ChrisInSydneythats a true joke
07:09.53ChrisInSydneyive finished with it
07:10.08lorsungcu_i have not found a similar thing in america
07:10.08ChrisInSydney+1 lorsungcu_
07:10.20kaldemarrolandow: or just don't set it.
07:10.24lorsungcu_only when i go to europe.  then i get fat til i return
07:10.25ChrisInSydneyyeros = greek kebab
07:10.46ChrisInSydneyno shwarmas ?
07:10.49ChrisInSydneyheaps here
07:10.53lorsungcu_nada
07:10.56rolandowkaldemar: yes, but i put it in a variable which depends on an if statement
07:11.04lorsungcu_i swear if i made it, i would be a millionaire.
07:11.07gg608fmexican tacos al pastor is similar to kebab
07:11.10lorsungcu_drive through kebab
07:11.22rolandowkaldemar: something like: if you can find second-line SIP, timeout 30 sec, otherwise, don't timeout.
07:11.23ChrisInSydneyCleavelan street has so many turksish, lebanese, etc places
07:11.25kaldemarrolandow: 0 and empty translate to 136 years or something.
07:11.33gg608fgood kebab replacement in america
07:11.44rolandowkaldemar: ok so 0 would do
07:12.05lorsungcu_never had it, gg608f
07:12.10ChrisInSydneymaybe thats what I need to be doing, launch my kebab chain at AstriCon
07:12.15lorsungcu_do it
07:12.18lorsungcu_i will cook
07:12.23lorsungcu_we will bank
07:12.34rolandow[Aug  2 09:12:16] WARNING[7067]: app_dial.c:2444 dial_exec_full: Invalid timeout specified: '0'. Setting timeout to infinite
07:12.43lorsungcu_roland
07:12.45rolandowhm... what is the NULL variable
07:12.46lorsungcu_make it empty
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07:13.08ChrisInSydneyback to work for me
07:13.13rolandow<PROTECTED>
07:13.24rolandowhow do i make that empty?? should i say "":30 at the en?
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07:13.28ChrisInSydneynow I feel hungry
07:13.29rolandowor NULL:30 ?
07:13.41Roelt?:30 might work
07:14.08lorsungcu_or an empty variable, if that doesnt?
07:14.26rolandowok "" works
07:14.36lorsungcu_hm
07:14.36rolandowRoelt: :30 wouldn't set it if the if is true
07:14.43gg608fyeah tacos al pastor is an evolution of kebab, made by lebanese in mexico, they add onion, cilantro and spicy sauce
07:14.45gg608fsometimes pineapple
07:14.48rolandowRoelt: and it's in a loop, so it could be set previously :)
07:15.09lorsungcu_where do i get this mexikebab, gg
07:17.04gg608fany mexican place
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07:17.16gg608fi live in california, so bunch of mexican places
07:17.16ChrisInSydneycilantro = coriander ?
07:17.21lorsungcu_im in MN
07:17.27lorsungcu_there are only some.
07:17.42lorsungcu_but i will be on the lookout
07:17.54gg608fyep, coriander
07:18.00gg608fi may try kebab :)
07:18.31ChrisInSydneywe have a mex place in staggering distance. He goes back to Mexico to record with a bunch of his mates each year.
07:18.42ChrisInSydneyI have a few of his CDs
07:19.00ChrisInSydneyhe is a good guitarist, not a bad voice
07:19.12lorsungcu_i used to have a similar setup
07:19.15lorsungcu_but it was taco bell...
07:19.20ChrisInSydney:-?
07:19.32lorsungcu_did many drunk drive through-walks
07:19.36lorsungcu_:/
07:19.38ChrisInSydney:D
07:19.49gg608fnice
07:19.51ChrisInSydneydone that, in a shopping trolley
07:19.55lorsungcu_man
07:20.15lorsungcu_its maybe the funniest thing, ever
07:20.16gg608f, i heard couple of mexican gutarists in coachella "rodrigo y gabriela" amazing
07:20.19lorsungcu_esp if you make car sounds..
07:20.20gg608fhahhaa
07:20.38*** part/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net)
07:20.46*** join/#asterisk lorsungcu_ (~anonymous@209-173-236-18.usfamily.net)
07:21.43coppicegg608f: the heavy metal nylon guitarists :-)
07:22.40gg608f:)
07:22.58lorsungcu_here is something important
07:23.00lorsungcu_http://en.wikipedia.org/wiki/List_of_sandwiches
07:27.38gg608fman…irc is making me fat
07:28.26gg608fdagwood....
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07:48.09ChrisInSydneyexten => _04XXXXXXXX,n,Set(CHANNEL(SIP_CODEC_OUTBOUND)=alaw)
07:48.09ChrisInSydney<PROTECTED>
07:48.27ChrisInSydneyis that correct, or am I missing something ?
07:49.08kaldemarCHANNEL is a function that has certain specified fields. it is not used to set any channel variable.
07:49.26kaldemarSIP_CODEC_OUTBOUND is a variable, not a field of the CHANNEL function.
07:49.27ChrisInSydneywhat about these: https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
07:49.32ChrisInSydneyahh
07:49.41kaldemarso Set(SIP_CODEC_OUTBOUND=alaw)
07:50.08ChrisInSydneycool
07:50.12ChrisInSydneybrb
07:52.15ChrisInSydneyback
07:52.16ChrisInSydneynup
07:52.43ChrisInSydneytried _SIP_CODEC_OUTBOUND and got [Aug  2 19:05:17] NOTICE[17760]: chan_sip.c:6455 try_suggested_sip_codec: Changing codec to 'alaw' for this call because of ${SIP_CODEC} variable
07:52.43edgarsyello
07:52.54ChrisInSydneybut
07:53.05ChrisInSydney192.168.231.201  HTI_extn202      3ec4263c2814-qr  0x1000 (g722)    No       Rx: ACK                    HTI_extn20
07:53.09ChrisInSydneystill transcoding
07:53.19ChrisInSydneyedgars: Hey
07:53.44edgarsyo ChrisInSydney
07:54.15ChrisInSydneyso 722, then alaw on the handset and alaw only on the SIP trunk
07:56.36ChrisInSydneyexten => _04XXXXXXXX,n,Set(_SIP_CODEC_OUTBOUND=alaw)
07:56.53ChrisInSydneyisnt good enough
07:56.59ChrisInSydneyexten => _04XXXXXXXX,n,Set(_SIP_CODEC=alaw)
07:57.00ChrisInSydneyexten => _04XXXXXXXX,n,Set(_SIP_CODEC_OUTBOUND=alaw)
07:57.03ChrisInSydneyworks :-)
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08:10.16ScapalCisco SPA500DS : extension pannel with LCD labels. I've bee waiting for those. I could only find traces of those in the latest Release Note of the phones firmware (20120724).
08:10.35Scapalhttp://www.static-cisco.com/assets/prod/pct/spa500ds.jpg
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08:11.37ScapalNot yet released apparently
08:12.00schmidtswow looks nice scapal
08:12.22ScapalThere is no full product description yet but you can get some by comparing it with the 500S: http://www.cisco.com/en/US/products/ps10499/prod_models_comparison.html
08:13.25schmidtsi guess it will just be the same like the 500S but with display ;)
08:13.44schmidtshmm maybe it will also work to display a callerid on a ringing blf key ;)
08:13.54schmidtsthen this stuff will really rocks
08:20.21Roeltshow waiting calls in a queue, number of voicemails
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08:46.25aberriosI see the MYSQL application has a connection timeout argument but there doesn't seem to be a way to timeout a query that takes too long to respond. Would I have to sort this out at the MySQL end?
08:49.46dexterukI have a stupid problem with AGI using a php script
08:50.13*** join/#asterisk dexteruk (~dexteruk@77.70.92.171)
08:51.02dexterukhi im trying to write a simple AGI script using PHP
08:51.14dexteruki cannot get it to STEAM FILE
08:51.29dexterukit runs the command, but i hear no sound
08:52.05dexterukif i PLAYBACK before i execute the AGI the audio file plays
08:52.29dexteruki am running php -q
08:52.45dexterukto stop any HTML codes being returned to asterisk
08:53.22kaldemardexteruk: perhaps the AGI stream command does not answer the line like Playback does and you need to use the answer command before it.
08:53.23dexterukDoes anyone have any experiance with AGI scripts?
08:54.09dexterukthere is an Answer at the begining before it calls the AGI script
08:54.30dexterukor you mean in the AGI script itself
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08:55.56kaldemardexteruk: enable agi debug with "agi set debug on" and pastebin the CLI output of a call.
09:05.22dexterukhttp://pastebin.com/KbxB5g3Y
09:05.26dexterukhere is the output
09:06.55dexterukhere is the AGI script itself http://pastebin.com/hAbUSkkY
09:07.48dexterukwhen it says the number it works fine
09:07.54dexterukjust when it comes to streaming the file
09:08.02dexterukit fails
09:08.18dexteruki have tried with full path to the file too
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09:55.17sbw_DvDdexteruk: you could try the phpagi class http://phpagi.sourceforge.net/
09:55.59sbw_DvDthey have some better output handling which avoids the broken pipe errors
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10:07.13awkhmm, anyone using this amr codec from sourceforge... I'm not sure about the 1.8.*_crypto_offer patch, it remove SHA1_80 encryption offer used by SRTP (Polycom, Aastra, Snom, etc)
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10:27.47ChanSpyI have an issue with Chanspy / Extenspy
10:28.44ChanSpyIt works if it's started before any spied channel
10:29.17ChanSpyif i start it after, it connects to the channel, but no sound is coming
10:30.04ChanSpywhisper / quiet mode doesn't change anything to this issue
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10:32.46jurij1hello i'm using elastix. is there a way i can have fax clients see only their received/sent faxes?
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10:43.41pbxMandoes anybody know whether is possible to change the installation directory for all asterisk folders without using links?
10:44.40pbxManis it right to change the default.h file?
10:45.55ChrisInSydneyII think you can do that during make ? :-/
10:46.07bulkorokcheck ./confgure --help
10:46.53bulkorokthere you'll find Installation directories: --prefix=PREFIX --exec-prefix=EPREFIX
10:50.08pbxManOK cheers mate
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10:58.08dax_roc<PROTECTED>
11:15.54*** join/#asterisk danfromuk (~IceChat77@2.27.1.139)
11:16.24danfromukHi, how can I convert origtime of a voicemail message to standard time format?
11:16.35*** join/#asterisk Sidrov (~sid@85.186.159.201)
11:16.38danfromukwhat format is origtime in?
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11:19.29Sidrovhello all
11:23.29Sidrovis there any active user ?
11:25.48leifmadseneh?
11:31.04dax_rocCan you call a skype user via sip like channel originate SIP/skype/username extension ### ?
11:31.27leifmadsenperhaps if you have Skype For SIP
11:31.55dax_rocI get a 484 , "address incomplete"
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12:17.36ChrisInSydneyhey, can someone help me with call park on 1.8.x
12:17.37ChrisInSydney?
12:19.26ChrisInSydneyporting from my legacy 1.4 systems, I have parking defined between 70000-79920 (long story)
12:19.59ChrisInSydneyusually I would do a Set(PARKINGEXTEN="70202") followed by a Park()
12:20.02ChrisInSydneybut it doesnt work
12:20.05ChrisInSydney....
12:20.19ChrisInSydneybecuase I have "" in the Set :-/
12:20.24ChrisInSydneyjust noticed that
12:20.36ChrisInSydneyis embarrassed
12:20.56ChrisInSydneyis crawling back into his hole
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12:28.03pouyouhi
12:28.08ChrisInSydneyhey
12:28.36pouyoui have a quick question someone might be able to answer
12:29.08ChrisInSydneyshoot
12:29.30ChrisInSydneybut not like in a batman movie
12:29.36pouyoulet's say i have a sip call between user A and B. Is it possible at some point via the dialplan to play a certain sound or video file to one of them while they are on the call
12:29.52dax_rocAnyone here worked with a matrix comsec GS12?
12:29.52pouyoubang !
12:29.56ChrisInSydneylook in features
12:30.27dax_roc*GS12 GE12
12:30.53ChrisInSydneypouyou: Just had this page open myself: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/AdditionalConfig_id256654.html
12:32.00ChrisInSydneythere is some DTMF stuff there for playiing recordings or running AGI
12:32.15ChrisInSydneyI have played with it and it worked on my old 1.4
12:32.32ChrisInSydneyworks with 1.8 as well. Just a bit better I guess
12:32.58leifmadsenpouyou: try looking at my presentation from AstriCon last year
12:33.23pouyousure, do you have a link ?
12:34.07leifmadsenhttp://www.astricon.net/videos/Cooking-with-Asterisk.html
12:34.25pouyouChrisInSydney, do not really see something I could use in the additionalConfig so far
12:34.27leifmadsenit's the 5th recipe I show
12:35.31ChrisInSydneypouyou:  additionalConfig...smells like freePBX
12:35.55ChrisInSydney?
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12:36.35pouyoui was mentioning the link you gave me :)
12:36.37leifmadsenunfortunately the player kind of sucks and doesn't have a time stamp on it, so I can't really find and tell you where to start watching
12:36.52ChrisInSydneyahh
12:37.07ChrisInSydneyI was only reading the parking stuff
12:37.10ChrisInSydney:/
12:37.42leifmadsenpouyou: leifmadsen.com should also have my presentation from last year that matches the video link I gave you and might be easier to get to what you want
12:37.43pouyouyeah
12:38.03pouyoui kinda was trying to dl the video as the player sucked so much :P
12:38.14leifmadsenindeed
12:38.25WIMPynever managed to get at the videos.
12:38.33ChrisInSydney[applicationmap]
12:38.35leifmadsenreally? it's just a link....
12:38.41ChrisInSydneytestfeature => #9,peer,Playback,tt-monkeys
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12:38.49ChrisInSydney[myGroupName]
12:38.55ChrisInSydneytestfeature => #9
12:39.20ChrisInSydneythen put Set(DYNAMIC_FEATURES=myGroupName) in the dial plan
12:39.43ChrisInSydneyleifmadsen is a far better cook than me.
12:40.06rolandowleifmadsen: why not use vimeo or youtube? i can't even play the video
12:40.07ChrisInSydneyI can only cook toast
12:40.11WIMPyA link to a page wit a video none of my download helpers or video plugis detects.
12:40.27leifmadsenrolandow: I had zero say or interaction with creating the video other than to do the presentation
12:40.54rolandowtoo bad.. maybe you can rip it and upload it to youtube under a different nickname ;-)
12:41.01leifmadsenmaybe you can
12:41.34WIMPyMaybe not that easy if we can't even watch it.
12:41.44leifmadsenI was watching it just fine
12:41.53rolandowit doesn't play without stuttering
12:41.59rolandowor what is it called in english ..
12:42.00leifmadsenFedora 17 w/ Google Chrome
12:42.03leifmadsenno issues here
12:42.04kaldemarleifmadsen: did you do it with applicationmap or with originating a call to ChanSpy?
12:42.16leifmadsenkaldemar: not sure, you'll have to check my presentation
12:42.21leifmadsenit's been over a year since I looked at what I did
12:42.29kaldemarleifmadsen: i tried, but gave up. :P
12:42.33leifmadsenI probably used stuff and things
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12:46.06jkroonis it possible to disable's asterisk's internal log rotation?
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12:46.19rolandowhow can you be so sure if you don't remember
12:48.01rolandowmaybe you used some Techniques .. who knows?
12:51.40leifmadsenrolandow: I know because I wrote it down and tested it
12:51.46ChrisInSydneyhmmm. Just playing with park and such and even though the console says "Will timeout back to extension [t02-parkinglot] s, 1 in 120 seconds" it throws the call back to the extension who parked the call
12:51.50leifmadsennot remembering doesn't mean it didn't happen
12:52.14ChrisInSydneyleifmadsen: Ive had nights like that
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12:52.25leifmadsenChrisInSydney: :)
12:52.25rolandowof course, but how can you be sure you used stuff and things then ..
12:52.37leifmadsenbecause I use stuff and things in everything i do
12:52.50rolandowi see
12:52.52rolandowi rest my case then
12:53.29rolandowjust curious, where do you draw the line between stuff and things?
12:53.51leifmadsenrolandow: I tend to use more stuff than things, but it depends on the implementation
12:54.12ChrisInSydneyyou can always use a thin veneer of BS when you get stuck
12:54.37rolandowdo you use anythings as well?
12:54.46rolandowor more somethings
12:54.48leifmadsenrolandow: nah, totally overrated
12:55.31ChrisInSydneyI used to have a good supply of whatcha-ma-giggys
12:55.34rolandowtaking notes
12:57.05ChrisInSydneybut the supplier started supplying lots of you know and didnt have room for both
12:57.11rolandowi just used two apples
12:57.22rolandowtrying to get the hunger feeling go away
12:57.35rolandowJunami apples .. the Elstars aren't sold anymore .. i guess Elstar season is over.
13:00.06Roeltis currently jailbreaking an apple
13:00.37Roeltbut, had some grapes this morning as breakfast
13:00.51ChrisInSydneyis is going to grab an orange
13:01.26ChrisInSydney2 bags of Aussie Navels for $5. Cant complain about that
13:06.03Kattyhello my asterisk does not work at all how to fix??? answer plz.
13:06.17malcolmdKatty: did you try turning it off and on again?
13:07.03Kattywhat is turn off???? *hee*
13:08.11ChrisInSydneyKatty: Hold down the shift key
13:09.34ChrisInSydneyDo you think I should take my son to Astricon for his birthday. He is turning 3 that week
13:09.59Kattyno
13:10.00ChrisInSydneyhmmm, how many hours from Sydney to LA
13:10.07Kattyi think that's a terrible idea for a 3 year old
13:10.18Kattythey need things to entertain them, and naps
13:10.24ChrisInSydneythats what I thought too
13:10.26Kattyand if he starts bawling in the middle of a talk, no one wants to hear that.
13:10.42ChrisInSydneyThe reminder email turned up
13:10.52ChrisInSydneylooked at the date, already booked
13:11.08Kattytake him to someplace COOL
13:11.49ChrisInSydneygood idea. I'll take him for a ski this year
13:12.18ChrisInSydneyhes old enough and end of September / OCtober the weather is niceish
13:12.26Kattyare there lots of fun Hands On stuff at sking?
13:13.40ChrisInSydneymy dad took me skiing at 2.5
13:14.15ChrisInSydneyhe held on to me and I skied between his legs
13:14.28ChrisInSydneyget him started early
13:15.30ChrisInSydneybut not too long. It has to be fun
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13:59.54areayhi all.. when installing asterisk 1.8.15.0 i'm getting an error with libspeex1: libspeex1 conflicts with asterisk (<= 1:1.4.18.1~dfsg-1)
14:00.13areaydoes this mean i need to downgrade libspeex1?
14:00.59ChrisInSydneyKatty: Seriously, with a toss up between Astricon and hanging around with my son and a bunch of three year olds running around on a sugar high....Ill take the sugar high any day
14:01.41pabelangerareay: you need a newer version of libspeex
14:01.53pabelangerwhat version of Debian?
14:02.10areaypabelanger it's ubuntu 10.04
14:02.28pabelangerwhere are you installing asterisk from?
14:02.41areayi'm installing from source and using checkinstall
14:02.55areaycompiling rather. sorry it's early
14:09.47Kobazanyone have problems with polycom phones losing their mwi?  like there is 2 messages waiting, and after restarting asterisk then there's no message light
14:10.10areaypabelanger, what's the best way of upgrading libspeex?
14:14.49ChrisInSydneyKobaz: are you using Asterisk Realtime ?
14:15.20ChrisInSydneyalso, FYI, the MWI message is a SIP notify message from the Asterisk box to the handset to set the MWI
14:15.59Kobazyeah
14:16.12Kobazit should redo the NOTIFY's on startup
14:16.27Kobazi'm using static realtime voicemail
14:16.34ChrisInSydneyI think there was somethng I read where the handset would request the MWI, while others relied on a push
14:16.49Kobazwell there's voicemail subscribe
14:16.50ChrisInSydneytry without the realtime voicemail
14:17.00Kobazthat's not really an option
14:17.35Kobazthe phone will subscribe to the box which will then tell asterisk to send it mwi's
14:17.38ChrisInSydneythere is a line in the blues brothers movie I could use right now
14:18.07Kobazwhat i should do is some tests and do some packet dumps on restart and see if it's resending mwi
14:18.41ChrisInSydneyI ran into this with someone else, ast realtime VM and dodgy MWIs
14:18.49ChrisInSydneycant remember how it was resolved
14:18.55ChrisInSydneysorry
14:18.57ChrisInSydney:(
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14:21.10Kobazhttp://www.voip-info.org/wiki/view/Asterisk+Realtime+MWI+Hacks
14:22.12Kobazwhat i can do, is add a timer to chan_sip to send out periodic notifys if it doesn't do so already
14:22.47ChrisInSydneysounds right
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14:26.27bulkorokhi... if I delete the txt and WAV files in the "Old" vm-folder, are they gone, so that the box is at zero!?
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14:30.17ChrisInSydneybulkorok: you can trash the whole extension folder and it will be recreated when Asterisk starts
14:32.59bulkorokI caonnot restart asterisk :-(
14:33.45WIMPyhands bulkorok a kill -9
14:34.04ChrisInSydneyyou can, but you dont want to
14:34.08ChrisInSydneygo on
14:34.15bulkorok:)
14:34.36bulkorokI'll try with a new box :-D
14:34.39ChrisInSydneyrestart when convenient
14:34.58bulkorokbut good to know anyway ...
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14:35.31ChrisInSydneythe trouble is convenient for the system and convenient for me sometime are different
14:35.45WIMPyDon;t say you can;t if you can, but don't want to.
14:36.07ChrisInSydneyyes, you can always choose the bullet
14:36.51ChrisInSydneygoes away on a short brain break
14:39.52uskerinehi, i have installed asterisk package in my distro and i have noticed that some extensions and dialplan (demo-ael) are installed by default
14:40.16uskerineaprat from the files I have modified (sip.conf, extensions.conf, queues.conf, chan_dahdi.conf, etc.)
14:40.22uskerinewhat else should be kept in /etc/asterisk?
14:40.53mirela666uskerine: probably extensions.ael file
14:41.51uskerinei also see pbx_lua  and pbx_config while (re)loading dialplan
14:41.58uskerineis that also added stuff?
14:43.19uskerineRegistered extension context 'app_queue_gosub_virtual_context'; registrar: app_queue
14:45.14mirela666uskerine: all the config files in /etc/asterisk are mostly needed to run normaly, the number of them depends on modules and apps you chose on 'installation' ./menuselect
14:46.24[TK]D-FenderWhich he didn't do
14:46.32mirela666:)
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14:48.05uskerineok thks
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14:50.57bulkorokinfo: I moved the wav and txt files out of the Old folder and the voice in of voicemialmain told me there are no messages... problem solved without restart :-)
14:51.16ChrisInSydneycool
14:51.51bulkorokstill the problem why the option to delete messages was not announced :(
14:52.08bulkorokthere were 123 messages...
14:52.13ChrisInSydney?
14:52.15ChrisInSydney:/
14:53.26bulkorokah ok... I don't investigate this...: version 1.4.23.2
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15:07.35straterraI know its not an asterisk question, but you guys are knowledgeable about phone stuff in general..
15:08.27straterraI need a device that will take a 3.5mm audio input and let me stream it over a network to a decoder with 3.5mm. The two sites are connected by fiber (no copper) and need to have shared paging..any recommendations from experience?
15:08.54uskerinerecorded audio with Mixmonitor sounds strange (like metalical sound)
15:09.18uskerineso human voice sounds a bit like a robot
15:09.22uskerineany idea on what could it be?
15:10.24straterraPoint to multipoint would be awesome too
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15:22.39mcolombohi all
15:23.04Kattyhihi
15:25.14slav3_kittenyo Katty *waves*
15:25.22Kattyohai
15:25.24Kattyhow'rechu dear?
15:27.17mcolomboi've a problem with hangupcause. In the SIP CANCEL Request the field "cause" is not present!
15:27.38mcolomboI've already add in my dialplan this : NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})})
15:28.38mcolomboand add use_q850_reason=yes in sip.conf
15:29.00mcolombobut does not work, anyone can help me?
15:29.01mcolomboThanks
15:29.49VultureZAnyone utilizing OpenSIPS?
15:29.58VultureZfor LB
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15:34.10dax_rocCan I do a pass through on a pri interface with asterisk?
15:35.13dax_roc* I want to filter only some calls and reinject them to an exsisting pabx via a man in the middle setup. Which would put us in a better place to migrate group by group ( if that makes sense )
15:36.07dax_rocPri -> Asterisk(pass - through) -> Legacy (PABX)
15:38.20[TK]D-FenderCall it ... acll out.
15:38.22[TK]D-Fendercall*
15:38.25[TK]D-Fenderal there is to it
15:38.56carrarYou can do anything with asterisk!
15:44.44straterraCan I call an extension and have my coffee maker make me a cup?
15:44.51Qwellstraterra: Yes, yes you can.
15:44.58QwellIt has been done (more than once).
15:47.23*** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com)
15:51.28[TK]D-FenderYes,, that is what my * does for me already
15:51.32[TK]D-Fenderand is my jukebox...
15:52.47[TK]D-Fenderhttp://purl.rikers.org/%23asterisk/20100423.html.gz
15:53.30[TK]D-Fenderhttp://ibot.rikers.org/%23asterisk/20081111.html.gz
15:53.34[TK]D-FenderOooohh, 2008..
15:53.41[TK]D-Fenderwonder just how long ago I did that...
16:02.58ChrisInSydneystrange thing. CONNECTEDLINE stopped working on my dev system, but it was working :-?
16:03.25*** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35)
16:04.44Kattyi have ham and potatoes in my fridge.
16:04.47Kattywhat should i make for dinner
16:05.40chuckfthat sounds like a good dinner to me, maybe make them hash browns
16:07.37dax_roceggs ham and spam ...
16:09.43newtonrjust microwaving a sweet potato is cool sometimes
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16:16.52vastinanewtonr: how modest
16:17.10newtonrAND THEN ADD BACON
16:17.23vastina\o/! bacon makes everything better
16:36.19ChrisInSydneyWierd. I have two accounts on the same box with the same phone
16:37.10ChrisInSydney"Set(CONNECTEDLINE(name,i)=Something" works on one reg but not the other
16:37.12ChrisInSydney:-/
16:38.05ChrisInSydneysip is the same except for alaw v g722 being first in the list
16:38.14ChrisInSydneyclues anyone ?
16:38.29ChrisInSydneysame profile on the handset
16:38.34ChrisInSydneyhandset is a Snom 970
16:38.36ChrisInSydney870
16:38.46ChrisInSydneys/970/870/
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16:56.01kukuI can't get cdr adaptive odbc to come up as a cdr module.. any ideas ? I'm using 1.8 - and after confnigure cdr adbaptive odbc has XXX - and its written that its in the core.
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17:04.49numenoryhaving bri issues on a b200p, supplier says configs are correct asterisk 1.8.15, libpri-1.4.12, openvox-dahdi-linux-complete-2.6.1+2.6.1, call connects briefly is crackly, then cuts off, console outputs.... SIP/33-00000001 answered DAHDI/i2/07952015843-1
17:04.51numenory[2012-08-02 10:59:12] ERROR[7682]: chan_dahdi.c:14135 dahdi_pri_error: PRI Span: 1 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established)
17:04.52numenory== Primary D-Channel on span 1 down
17:04.54numenory[2012-08-02 10:59:12] WARNING[7682]: sig_pri.c:1126 pri_find_dchan: Span 1: D-channel is down!
17:04.55numenoryPRI Span: 2 TEI=0 MDL-ERROR (C): UA (F=1) in state 7(Multi-frame established)
17:04.57numenory== Primary D-Channel on span 1 up
17:04.58numenory-- Span 1: Channel 0/1 restarted
17:04.59numenory[2012-08-02 10:59:12] ERROR[7683]: chan_dahdi.c:14135 dahdi_pri_error: PRI Span: 2 TEI=0 MDL-ERROR (J): N(R) error in state 7(Multi-frame established)
17:05.01numenory-- Span 2: Channel 0/1 restarted
17:05.03numenory== Primary D-Channel on span 2 down
17:05.04numenory[2012-08-02 10:59:12] WARNING[7683]: sig_pri.c:1126 pri_find_dchan: Span 2: D-channel is down!
17:05.06numenoryafter that you cannot connect anymore until you restart dahdi
17:05.08numenorydebian 6, box
17:05.12*** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
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17:14.21Kobazaaaaah
17:14.38Kobazso asterisk does not resend sip notify after restart
17:14.47Kobazusing realtime voicemail
17:14.59Kobazin 1.8.12 anyway
17:15.49KobazChrisInSydney: should be easy to fix
17:16.16ChrisInSydneyyou would hope so
17:16.39ChrisInSydneyyou can use the AMI to send SIP NOTIFY messages, but your content has to be in reverse
17:16.42ChrisInSydneylast line first
17:17.39Kobazno i mean like
17:17.48Kobazjust make app_voicemail send NOTIFYs on load
17:19.14Kobazthat's the easy way
17:22.16ChrisInSydneycool
17:22.32ChrisInSydneymy Park BLFs dont work
17:23.01ChrisInSydneyregular BLFs do work
17:23.12Kobazooooh, more interesting tidbits
17:23.21Kobazit actually does send a notify, but it's inaccurate
17:23.33KobazMessages-Waiting: no
17:23.34Kobaz<PROTECTED>
17:25.24ChrisInSydneythese phones are on drugs. Now Park 1 works, park2 nahda
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18:42.43xSmurfhey all, I'm trying to get asterisk 1.6.2.9 and meetme working in a domU, but all the instructions I find seem outdated, zaptel is gone and the dahdi source do not include USE_RTC anywhere....
18:43.10xSmurfeverything else I find is "google, the answer is there".. super helpful considering all the answers on google do not work
18:43.33sruffellxSmurf: if you can load / compile DAHDI you're good.  No more need for dahdi_dummy, etc...
18:43.41xSmurfI see
18:44.26xSmurfI already have the dahdi pkgs but I can't seem to modproble dahdi
18:44.32xSmurf*modprobe
18:44.40xSmurf(Debian squeeze)
18:45.17sruffellwhat does it say when you try to modprobe?
18:46.17xSmurfmodule not found
18:47.11xSmurfstarts a 'find / -name "dahdi.so"'
18:47.18xSmurfyeah, non existant
18:47.22sruffelluse .ko not .so
18:47.41xSmurfthx, I tried so cause "module load dahdi" returned "Error loading module 'dahdi': /usr/lib/asterisk/modules/dahdi.so: cannot open shared object file: No such file or directory"
18:47.44xSmurfno .ko either :/
18:48.16xSmurfmaybe I need to install the non-free pkg?!
18:48.22sruffellI would just build from source then.
18:48.36sruffellI'm not that familiar with the debian packages.
18:49.16xSmurfbuilding from source is not the greatest for production usage
18:49.39xSmurfI can be harsed to go and manually check every other day for new security fixes and recompile everytime
18:49.45VultureZAnyone utilizing OpenSIPS as a load balancer for multiple * deployments?
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18:53.52xSmurfsruffell: nevertheless, thanks for the help, it was enlightening
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19:16.25sruffellxSmurf: yeah…I hear you there.
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19:52.04UnixDevhi everyone, I'm using realtime peers but for some reason mohsuggest does not seem to be pulled from the database… is there something I don't know about this?
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20:01.13navaismoHi if anyone is interested in a call monitor program like YAACID here is another one,  for linux and windows http://dl.dropbox.com/u/1277237/DM_CallMonitor.tar.gz
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20:05.59sebastianpersicI didn't create the asteriskpbx user before install... what should I do
20:06.24[TK]D-Fendermake it after if you care
20:06.34[TK]D-Fenderit's all int he book
20:07.21sebastianpersicok.. because I saw some text during boot up my system that it didn't knew user asterisk and group asterisk
20:17.01[TK]D-Fendersebastianpersic, and we don't know what your base OS is, or how you installed *, and what any of it is expecting
20:25.22[TK]D-Fendercheckout time, later all
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20:43.27WIMPynumenory: Looks like you've got a bad connection. Either in hardware or in software.
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20:49.26Mango45In what variable is the audio IP for an in-progress call?
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20:55.59kukusomeone please help me with getting the adaptive odbc showing up as a registered backend. pretty please.
20:58.14doolittleworkwhat do you mean kuku?
20:58.17kukuI've spend 3 hours on trying to get it to work. I have odbc.ini settings, res_odbc.conf, cdr_adaptive_odbc.conf, cdr.conf, odbcinst.ini configured.... Running 1.8, so adaptive is part of "core" and doesn't ahve to be compiled.
20:58.48kuku"cdr show status" shows cdr-custom and csv  only
20:58.48mjordankuku: uhm, no, it still has to be compiled
20:59.19kukumjordan: which option in menuselect?
20:59.44xSmurfso "|" was changed to ",", is the warning just a deprecation warning or is | actually broken?!
20:59.45kukucdr_Adaptive_odbc has XXX  where other options have [*]
21:00.12xSmurfsruffell: on top of that dahdi includes binary blobs which is really unacceptable in my situation
21:00.23mjordankuku: then that would mean it can't be compiled
21:00.42mjordankuku: highlighting it should tell you what the dependencies are
21:00.43xSmurffor the |... I should specify from 1.4 > 1.6
21:00.57kukumjordan: res_odbc
21:01.11mjordankuku: okay, is res_odbc enabled?
21:01.27kukuaaaaaaaaaaaaaaaa :)
21:01.57kukuits not enabled, because it needs generic_odbc, and ltdl
21:03.00sruffellxSmurf: cool…yeah, I guess just need to get the debian packages installed
21:03.13kukulet me run ./configure again...
21:04.14xSmurfsruffell: I mean, the dahdi-non-free pkg didn't install the appropriate kernel module... building from source downloads a whole bunch of binary blobs
21:04.17kukumjordan: :) can I kiss you ? I've had a very bad day
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21:04.24kukuand now it works!!
21:04.39mjordangood :-)  Glad to hear it
21:04.40navaismo0_o
21:04.50mjordan(but I'm happily married, thanks)
21:04.58kukunow i can go and do the other 200 things I have to do
21:05.26sruffellxSmurf: there is only one binary blob that is code..the rest is firmware for the various cards. You can edit Kbuild to only build dahdi.ko if you just want software mixing.
21:05.39kukumjordan: much thank you. two days wasted.
21:05.47xSmurfsruffell: thanks, that's good to know
21:05.48sruffell(and that binary blob is to load firmware too….no way around that unfortunately)
21:06.29*** part/#asterisk Mango45 (~Mango45@d209-89-215-139.abhsia.telus.net)
21:08.23sruffellxSmurf: if you're only just using software mixing too…and you're ok with building…change DAHDI_CHUNKSIZE in include/dahdi/kernel.h from 8 to 40 in order to mix audio in 5ms increments instead of 1.  Much more efficient if you don't need to talk to hardware.
21:08.38sruffellwhen all the audio is coming in at typically 20ms intervals anyway.
21:09.00xSmurfwow man, I'm gonna have a lot of documenting to do in our wiki ;)
21:09.13sruffell:) …or not.
21:09.31xSmurfwell I a) need to remember all this b) should make this available to all more easily
21:09.48xSmurfwe usually have pretty hefty docs on things we use and a lot of people seem to use it
21:10.19sruffellhttp://article.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/51777
21:10.45xSmurfDanke!
21:10.57sruffellxSmurf: why do you even need DAHDI in a virtual machine?  You can't use app_confbridge? You need page app? or some other reason?
21:11.25xSmurfhmmm maybe, we were using app_meetme for conferencing
21:11.39xSmurfI'm new here, and was charged with migrating the voice boxen
21:11.51xSmurf(I think they wanted to laugh at me banging my head on the desk :P)
21:12.16sruffellok…afaik with a newish kernel (that supports timerfd) and asterisk-10….app_page is the only functionality that can't be done without dahdi currently.
21:12.41xSmurfwhen I tried app_meetme bitched at the lack of pseudo device :/
21:12.47xSmurf*when I tried without dahdi
21:12.52mjordansruffell: and in Asterisk 11, that won't be true either!
21:13.18sruffellyeah…app_meetme does all the mixing in the kernel.  It was easier that way in the beginning (I'm assuming, since I wasn't there) since the drivers already implemented mixing in order to bridge channels between cards.
21:13.47sruffellmjordan: woot!
21:14.41xSmurfsweet
21:15.45Qwellsruffell: SLA :(
21:15.53Qwellbut we don't talk about that
21:16.23sruffell?  I'm not following. Shared Line Appearence?
21:16.38sruffellxSmurf: a little Zaptel / DAHDI history http://thread.gmane.org/gmane.linux.kernel/40267
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21:16.56sruffellregarding why mixing in the kernel.
21:18.13Qwellsruffell: yeah, app_meetme has some weird SLA stuff still.  confbridge will never do that.
21:18.23Qwell(because it's a terrible idea)
21:18.36sruffellahhhhhh....
21:18.46xSmurfsruffell: sorry for the dumb question... what do you mean by "page app"
21:19.11sruffellif you wanted to broadcast a page to a bunch of endpoints…
21:19.23xSmurfah I see
21:19.31sruffellbasically it will connect to a bunch of endpoints, and drop them in a DAHDI conference to do the broadcast.
21:19.54xSmurfgood question..  we might use it for our nagios integration for oncall admins
21:20.03sruffellwait for Asterisk-11
21:20.05sruffell:)
21:20.05xSmurf(I just can't wait to get my name on that list ;p /s )
21:21.33Qwellwhat list?
21:22.19_Corey_there's a mailing list ;)
21:22.51Qwell_Corey_: I haven't heard you shouting from the rooftops about how awesome DPMA 1.2 is.  I'm sad. :(
21:22.58_Corey_haha
21:23.05_Corey_I loaded it the other day
21:23.20Qwellwell, I'm not hearing you shout at me either, so I guess it went okay.
21:23.48_Corey_no problems to speak of...  haven't had time to shake out the new features yet
21:23.56Qwellclose enough
21:24.16_Corey_it's a little unclear how the Queue app is supposed to work, so I was planning to poke at it a bit later this evening
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21:24.47Qwellit's a little funky, just know that you need a separate type=queue for each phone *and* for each Asterisk queue.
21:24.54Qwellso 3 users with 3 queues = 9 entries
21:25.05Qwell(assuming all 3 users are in all 3 queues)
21:25.14xSmurfQwell: the on call admins ;)
21:25.23QwellxSmurf: oh.  yeah, good luck with that.
21:25.27xSmurfso I gotta make sure that asterisk box is rock solid ;)
21:25.50_Corey_xSmurf: You don't want to be on that list...  hire an answering service
21:25.56xSmurf:p
21:26.17xSmurfwhat do you think asterisk is for :P
21:26.53_Corey_Qwell: makes sense... my mistake was trying to reverse engineer the FreePBX stuff you guys added for that
21:27.03numenorywimpy: still there?  tried another isdn2 box on the wall same result, both boxes function ok with the existing pbx (not asterisk)
21:27.17_Corey_if it works, I'm sure I'll figure it out...
21:27.24Qwell_Corey_: You probably don't need the login/logout_exten
21:27.59QwellFreePBX uses them, because it has a bunch of magic it needs to do.
21:28.45numenorywimpy: not sure if i need the molex power plugged in, in te mode, and whether having it plugged in could cause the issue
21:31.32_Corey_Qwell: hmm, I have type=application,application=queue in the queue entries in res_digium_phone_applications.conf
21:32.00Qwellfyi, nothing actually reads that file - in FreePBX, it's included
21:32.04Qwell#included*
21:32.32Qwellie; just put it in res_digium_phone.conf, unless you've done #include magic
21:32.44_Corey_yeah, res_digium_phone.conf #includes it
21:32.46Qwellif that's not the problem, I'd be happy to look over your config if you pastebin parts of it
21:33.02Qwellreally, just the queue contexts
21:33.17_Corey_thanks, I'll poke around a bit with it later when I have more time...  I may bother you about it tomorrow :)
21:33.43Qwellalso don't forget your application=contextname in type=phone contexts
21:35.33_Corey_I'll let you know what happens.  I'm having load balance problems at the moment...
21:43.39WIMPynumenory: Don't know that card, but it's unlikely you need extra power. Receiving garbage is often caused by IRQ issues, however.
21:43.56WIMPyFaulty cabling would be another possibility.
21:44.05WIMPy(i.e. wrong termination)
21:45.39numenorywimpy: sitting on irq 21 on its own
21:46.23numenorywimpy: was going to try a different slot
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21:53.29numenorywinpy: do get this in dmesg - IRQ 21/b4xxp: IRQF_DISABLED is not guaranteed on shared IRQs
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22:19.26rocksfrowHi
22:19.41rocksfrowI was hoping somebody could enlighten me
22:20.00navaismo~ask
22:20.00infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
22:20.16rocksfrowi'm typing! heh
22:21.04rocksfrowI have a callmonitor script setup that's watching queue_log, and I didn't realize it but a few months ago the structure of queue_log changed... on CONNECT events the agent used to show in the format including the extension, for ex: Local/1234@from-queue
22:21.28rocksfrowNow I notice this has changed a few months ago and instead the fullname of the extension is being output there instead of the device/extension
22:21.40rocksfrowIs this configuration that was added, or changed default behavior?
22:22.54rocksfrowso anyways, the script was relying on the extension being there (via regex), but now instead of 'Local/5200@from-queue' being the value, it's changed to 'John Doe'
22:23.36rocksfrowRight now i'm looking at modifying the script and doing a lookup based on name instead which isn't a huge deal, but obviously my preferred solution would be a configuration change to cause * to log the extension # instead of name in queue_log
22:23.43rocksfrowSuggestions please?
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22:33.39rocksfrowAnyboddy? :-/
22:37.21rocksfrow_afkplz any tips would be nice, bbs.
22:43.21MiccDoes anyone still have the aastra problem that the phone never stops ringing sometimes?
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22:51.09newtonrrocksfrow_afk: you hit the non-busy time, you might try posting on the asterisk-users list
22:53.16vastinahey phreaks... how can I get a serial number from a digium card from the CLI?
22:53.30vastinatried dahdi_scan() but couldn't find serial
22:53.51WIMPyshutdown -h now; use screwdriver
22:54.11vastinaWIMPy: you assume so damned much when I speak... remote site...
22:54.15vastina:)
22:54.21citywokrocksfrow_afk: adapt your script to work either way :p -- lol
22:54.24WIMPyok
22:54.34WIMPyshutdown -h now; drive there;use screwdriver
22:54.42citywokvastina: shutdown now -h; call onsite person; have htem use screwdriver
22:54.43vastinafacepalms
22:54.58WIMPyMaybe better drive there before shutting down.
22:55.11citywokvastina: look at the invoice from when you bought it?  the SN is probably on there.
22:55.24WIMPyAgain, call before doing the shutdown.
22:55.26vastinaok so there's no way from software
22:55.28vastinathanks.
22:56.04citywokvastina: probably not, that would require embedding it in each chip which seems like more work.
22:56.05WIMPyI don't think they have a serial number other than in ink on paper.
22:56.40citywokthe more softwarey devices have it, but that's because they have memory. (sip/pri gateway)
22:58.02doolittleworklol srewdriver
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