IRC log for #asterisk on 20080318

00:00.46SteveTotaroi pay a $.01/min except instate is $.05/min but that is easy to work around
00:00.59SteveTotarosix second rounding
00:02.04SteveTotaro~tdm
00:02.04jbotit has been said that tdm is Time Division Multiplexing. It is a scheme in which numerous signals are combined for transmission on a single communications line or channel. Each signal is broken up into many segments, each having very short duration.
00:02.18mitchelocSteveTotaro: mog wants to know how many minutes you push
00:02.42SteveTotaronot enough to seek out a better deal yet
00:02.46*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
00:03.07SteveTotaroplus i love to have a real t1
00:03.31mitchelocjust got his first pri :)
00:03.32SteveTotaronot some "virtual T1", i guess i am old school like that
00:03.40JTvirtual pris suck
00:03.56JTi just got a pri in the datacentre brought up on the weekend
00:04.22SteveTotaroonce managed a t3 a year ago
00:04.54JTcabling was finally done last week
00:04.57SteveTotarobrought into a adtran 2800 m13 and broken into 28 t1 PRIs
00:05.07JTgot the telco to run 2 * 16 pair screened cables to my rack for free
00:05.18JThmm
00:05.32SteveTotaroi got the telco to run two coax past the demarc
00:05.45SteveTotarofor free
00:06.14JTin my case they had to run the cables vertically up 2 floors, then horizontally 30metres
00:06.30JTand the cables need to be tagged every 3 metres under datacentre rules
00:06.38SteveTotarowe already had fiber at the demarc
00:06.54SteveTotaroand a great big mux
00:06.59mitchelocJT: is there a fee to run the cables between floors?
00:07.08adeelin the beginning of all my calls (zap -> sip, sip -> sip) there's a 2-3 second audio delay....is there anyway to get around that? or reduce it at the least?
00:07.13JTmitcheloc: usually whatever a cabling contractor charges
00:07.22JTmitcheloc: i got the telco to pay for the contractor though
00:07.27SteveTotarobetween floors is the easy part
00:07.39SteveTotaroyou just feed it down the conduit
00:07.49JTthere is no conduit here
00:07.54mitchelocyes, but some buildings charge monthly for that kind of a cross connect
00:08.00SteveTotarocable contractors are for sissies
00:08.05JTjust wiring cupboards and firestopper materials
00:08.20JTSteveTotaro: the datacentre won't let unlicensed people cable
00:08.27JTmitcheloc: those buildings suck ;)
00:08.39*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
00:08.53SteveTotaronot sure where you get a license for low voltage cabling around here.....
00:09.10SteveTotarosometimes you need a permit but that's about it
00:09.19JTyou need a licence to run telecommunications cables here
00:09.32*** join/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net)
00:09.32*** mode/#asterisk [+o mog] by ChanServ
00:09.42SteveTotarowhat if you are just crossconnecting?
00:09.44dansazcan hard phones interface with asterisk through an ethernet switch? or do you need a channel bank?
00:10.02SteveTotaroif it is past the demarc then you shouldn't need a license, it's CPE
00:10.04JTyou mean plugging in an RJ45 or punching down?
00:10.09JTi'm not in america.
00:10.16SteveTotaroi understand that
00:10.21mitchelocdansaz: a channel bank or ata
00:10.24SteveTotarojust makes no sense
00:10.38JTif it's not involving plugging in premade stuff, you need a licence
00:10.49SteveTotarowow, that's tough
00:10.53JTwell they're worried about people screwing up telecommunications infrastructure, i guess
00:11.06SteveTotaroi am old school with my cabling skills too, has to perfect
00:11.11JTi wish i got the licence before they made it harder to get
00:11.45SteveTotaroi have nice long 200 pair in my trunk ;)
00:13.08JTi do my own telecomms, data and electrical cabling at home :)
00:14.03SteveTotarowell i have seen alot of telco guys including verizon yank someone else's existing pair with dialtone and cross connect it with new circuits just to close a ticket (historic Washington DC)
00:14.29SteveTotarothere just isn't enough copper
00:14.36riddleboxSteveTotaro, my experience has shown the old school guys were way sloppier with cabling than guys now
00:15.08mipstergotta love those wax covered strings
00:15.11SteveTotaroi would disagree with that but it is just from my personal experiences
00:15.21*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
00:15.30fujinIs there anyway (a channel variable) to get the name of the agent who answered a queue call for MONITOR_FILENAME?
00:15.52fujinI'm tryign to build a filename which looks like date_agent-${AGENTNUMBER}_caller-${CALLERID(num)}
00:15.56riddleboxSteveTotaro, I will take some pics on job sites of things guys did before us young guys came in
00:15.59fujinworking out how to get agentnumber is the prob
00:16.28SteveTotarowell i have seen some rats nests too but usually in 100 year old buildings
00:17.17xacatecashow do I rewrite a number of the form 0ZX. to be of the form 27ZX. ?
00:17.22SteveTotarofujin:  you could look it up in your cdr or queue_log and rename it
00:17.25*** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25)
00:17.28xacatecasbasically replace the leading zero with 27 ?
00:17.39riddleboxin st. louis I guess it was just accepted to throw cabling everywhere, and the phone rooms whoa they are bad sometimes, we take before and after pics for customers
00:18.02*** part/#asterisk dansaz (n=dan@c-68-58-81-102.hsd1.in.comcast.net)
00:18.10*** join/#asterisk hijacked (n=argh@cerebus.clandestineresearch.com)
00:18.48mipsterexten => 0ZX.,n,Dial(Chan/Prov/27${EXTEN:1})
00:18.49fujinSteveTotaro: ok, how about a real suggestion?
00:18.50mipsterI think
00:18.53fujina channel variable is what I'm after
00:19.01SteveTotarothat was a real suggestion
00:19.39fujinthat was a fail suggestion
00:19.50SteveTotaroyou could load your queue_log into mysql query it and rename the file
00:19.57SteveTotaroas a scheduled cron
00:20.07*** join/#asterisk really_phukt (n=chatzill@209.216.64.44)
00:20.55mipsterxacatecas, obviously you'd need to replace the n with a 1 if it's the first step in the dialplan for that extension
00:20.59jblackfujin: You can set variables in sip.conf
00:21.09fujinjblack: uh?
00:21.20fujinI want a variable to contain what agent the call was delivered to, for queueing
00:21.22jblackso, perhaps you could set FULLNAME for your authentication for your phones, and use that variable.
00:21.34fujinI'm doing hotdesking, Phones are irrelevant really
00:21.40jblackoh
00:21.51xacatecasexten => 0ZX.,n,Dial(Chan/Prov/27${EXTEN:1}) <-- ... i've got extensions of the form 27ZX., but a client may well dial 0ZX. instead, so I want to handle both.
00:21.57*** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk)
00:22.17xacatecasthe way I handled that until now was to have two exten lines, one for the 27 case and one for the 0 case ... i was hoping for a more elegant solution.
00:22.28fujinhm, setinterfacevar might do it.
00:22.59SteveTotaromy suggesting is totally workable unless you are in some kind of sweatshop outbound call center
00:23.30SteveTotarohangups every couple of seconds
00:23.58xacatecasok, i'm betting some sleep is going to get more done for me than trying to work more.
00:24.01xacatecasadios
00:24.15fujinhum
00:24.17mipsterxacatecas, ok, hmm.  in that case how about a exten=> 0ZX.,n,Goto(context-for-extenstions, 27${EXTEN:1}, 1)
00:24.31SteveTotarohe's gone in the wind
00:25.09jblackI would have told him I think he's missing a leading _
00:25.15mipsteroops
00:25.16SteveTotarogoto is the best thing evar invented in programming
00:25.20mipstergood catch
00:25.29mipsterI always forget the danged _
00:25.34fujinheh. Stupid MONITOR_FILENAME is evaluated before a call is delivered.
00:25.40fujinwhat a failburger
00:27.00SteveTotaroobviously if you are using monitor in asterisk you cannot be doing more that ~70 simultaneous calls (without ramdisk or some hack) right?
00:28.33SteveTotaroyou could do a little AMI magic to catch the event
00:28.50*** join/#asterisk atis_home (n=chatzill@193.238.213.215)
00:29.25fujinmm
00:29.37fujinIt'd be better to have monitor_format in the queues.conf, really
00:29.44fujinand it's evaluated @ mixdown time
00:29.47fujinnot the case
00:29.51fujinwill just have to ignore the agent id
00:30.30*** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net)
00:31.45SteveTotarosean has a way of making a room quiet
00:35.27*** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.135.38)
00:38.11*** join/#asterisk lpmusic (n=dballeng@reddy.d-11.denetron.net)
00:38.48Mavvie.... silly users are reporting crossed calls again (where the RTP stream gets delivered to the wrong endpoint).
00:38.52*** join/#asterisk draygon (n=lokbo@76.185.106.151)
00:39.01MavvieAnybody here knowledgable with the way to troubleshoot this?
00:39.11BobLutzraises eyebrow
00:40.07CCFL_Man2wctdm can be used with what hardware?
00:40.38*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
00:41.06BobLutzCCFL_Man2, TDM
00:42.44*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
00:43.32SteveTotaroMavvie:  this could very well be happening
00:44.04CCFL_Man2BobLutz: does that include ant T1 cards?
00:44.48SteveTotaroi had the same complaint and thought "silly user" but then when all calls were recorded, it was proven that calls were being crossed
00:44.53BobLutzCCFL_Man2, Never touched a t1, If it has TDM in the model name, I would think it could work
00:45.08MavvieSteveTotaro: I know it happens, that's for sure. But I don't know how to troubleshoot it.
00:45.16MavvieOr even where to start with gathering information.
00:45.29SteveTotaroi never figured it out either Mavvie ;)
00:45.37Mavviewrong answer :-P
00:45.42CCFL_Man2BobLutz: ahh, i want to create a T1 from the asterisk box
00:45.48SteveTotaroit was so rare and everything looked fine in the logs
00:46.02SteveTotaroi could not reproduce it
00:46.36SteveTotarowctdm works for pots lines
00:47.22JTyeah not TDM lines
00:47.27JTasterisk naming is weird like that
00:47.38BobLutzfeels liek :-[
00:47.43CCFL_Man2ahh shit
00:48.10robmac67SteveTotaro & Mavvie: I believe that this was identified as a bug and fixed in the latest 1.4.19 Release Candidate
00:48.23CCFL_Man2i can't use a T1 card with solaris
00:48.24Mavvierobeph: oh!
00:48.28Mavvierobmac67: oh!
00:48.45SteveTotarobut there are 29 new bugs introduced
00:49.12SteveTotaroi don't think i could run an RC in production
00:49.38SteveTotaroi would live with the random crossed call
00:49.42*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net)
00:49.50*** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net)
00:49.55robmac67From what I recall it was introduced in 1.4.18 but you would have to check that out - I don't know what .17 was like
00:50.05SteveTotaroCCFL wctdm is really just for the pots cards
00:50.15MavvieSteveTotaro: RCs are just for that.
00:50.19lpmusicwith either 1.4 or 1.6 can you change the announcement to the agent answering a queue to make it play the announcement after you pickup but before you hit pound to accept the call?
00:50.27SteveTotaroother cards use wct4xxp and the like
00:50.53SteveTotaromaybe a release candidate is that to you but not to me
00:50.53CCFL_Man2SteveTotaro: sucks, wonder if i can build those drivers for solaris
00:51.32SteveTotaroCCFL i bet google can tell you, i know you can run asterisk on solaris
00:51.51SteveTotaroi had a netra 100 running asterisk
00:52.12*** join/#asterisk adeeln (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net)
00:52.17JTSteveTotaro: zaptel is the issue
00:52.27SteveTotaroyes obviously
00:52.40SteveTotarothat is why i told him to google
00:52.51Mavvierobmac67: just gone through the changes, but it's as clear as mud which one it could be.
00:52.59CCFL_Man2SteveTotaro: i'm trying to run it on a netra t1 200, solarisvoip.com has the stuff, but only tdm driver is tdm over ethernet
00:53.31JTCCFL_Man2: let me know how it goes :)
00:53.40CCFL_Man2JT: lol!
00:53.41JTi have a pile of sun netra T1s lying about
00:54.06CCFL_Man2sun claims they are "carrier grade"
00:54.33JTreal carriers don't terminate to standard pci cards
00:54.48SteveTotarohttp://www.voip-info.org/wiki/index.php?page=Asterisk+Solaris+Support
00:55.28CCFL_Man2JT: i want to generate a T1 on the netra, not terminate
00:55.33CCFL_Man2SteveTotaro: yeah
00:55.50SteveTotaroyou are on the cutting edge my friend
00:56.04JTsure, but it is totally different aspects that get them the carrier grade certification :)
00:56.19CCFL_Man2SteveTotaro: lol
00:56.31CCFL_Man2JT: i know, wtf do telco's use them for?
00:56.36SteveTotaroany reason why you don't do it with other hardware?
00:56.50CCFL_Man2i don't care for linux
00:56.54JTCCFL_Man2: running software on
00:56.59SteveTotaroouch
00:57.15JTCCFL_Man2: they have DC models and all that
00:57.22SteveTotaroi am waiting for a VxWorks port myself, probably will be waiting a long time
00:57.51CCFL_Man2JT: yeah, but what software?
00:58.00CCFL_Man2SteveTotaro: ouch
00:58.20JTCCFL_Man2: expensive proprietary carrier software, that runs on solaris
00:58.23JTdatabases
00:58.28JToperations and maintenance
00:58.36SteveTotarothat is the OS running satellites and junk in space
00:58.51rkeeneWhat's up with DUNDI.com redirecting to Digium ?
00:59.06SteveTotaroDNS FUBAR
00:59.14rkeeneOh
00:59.16SteveTotarobut they own dundi
00:59.25SteveTotaro(tm)
00:59.27CCFL_Man2JT: in the CO?
00:59.28BobLutzWhy?
00:59.29rkeeneI was expecting to find useful information there :-P
00:59.59SteveTotaroprobably find more useful info on someone's blog
01:00.23SteveTotaroor howto, i know there is a large group out there for dundi
01:00.30SteveTotaroforget the name though
01:00.37riddleboxrkeene, just wait till [TK]D-Fender comes around
01:01.01JTCCFL_Man2: sure
01:01.01CCFL_Man2SteveTotaro: carrier access uses it on their adit 600 tdm card, motorola uses it on their satellite receivers too, both do run rock solid stable
01:01.23outtolunc'dundi, so easy a caveman could do it' <G>
01:01.23JTriddlebox: you sure that would help? :P
01:01.31SteveTotaro3com uses it on the NBX and V3000 PBXs
01:01.57SteveTotaroi think fender is part of that large group i was speaking of
01:02.43CCFL_Man2JT: think they use any telecom software on it?
01:03.22JTCCFL_Man2: yes heaps of telecom software runs on solaris
01:03.32JThowever not most of the stuff in the media path
01:03.41JTthat's usually embedded with custom firmware
01:03.55CCFL_Man2ahh
01:04.00SteveTotaro(busybox)
01:04.16JTlol
01:04.27SteveTotaroactually probably VxWorks
01:04.36JTor their own stuff
01:04.52mipsterAnyone set up a Portech 370 GSM-SIP Gateway?
01:05.05mipsterSeems it's delivering CIDNAME and CIDNUMBER backwards
01:05.25mipsterIOW, Name is coming across as number and vice versa
01:06.19SteveTotaromipster:  that would be so low on my priority list as long as all the info is getting there
01:06.24SteveTotaro:)
01:06.45mipsterYeah, its only a minor hassle.  Just wondering if it's an AT&TW thing or a Portech thing
01:06.54JTprobably a setting or bug on the portech?
01:07.20SteveTotaroyou could easily work around it if all you need to support is portech
01:07.27mipsterI don't see a setting.  I'm guessing a bug.  Just wondered if anyone had used one with another carrier and/or SIM card
01:07.35mipsterand if the NAM and NUM are reveresed
01:07.58SteveTotaroit is a feature
01:08.02SteveTotaro;)
01:08.03mipsterlol
01:09.13draygondoes anyone need a server in Dallas, TX? I have way too much space that im not using
01:09.38SteveTotarohow much for how much
01:09.40SteveTotaro?
01:10.02draygoni pmed you
01:12.53rkeeneI have my Polycom phones with the minibrowser all setup now... what are some popular/useful things to do with that ?
01:13.43SteveTotarodisplay the company logo
01:14.03rkeeneUseful ? :-P
01:14.15JTdraygon: just space, or a server too?
01:14.29SteveTotaroto the ceo, heck yeah, they eat that stuff up
01:14.40rkeeneWe don't have a CEO
01:14.45SteveTotaroyou could display weather info
01:15.04SteveTotaroi mean what is really useful in a minbrowser?
01:15.05draygonJT, colo or dedicated whichever you prefer.
01:15.13JTdraygon: ah cool, how much?
01:15.22SteveTotarowhen you have a pc right next to the phone?
01:15.28rkeeneI dunno, I was thinking like an interface to change preferences or something
01:15.38draygoncheck pm
01:16.11*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
01:16.12SteveTotaroget FOP working on it with touchscreen
01:16.33rkeeneI already set it up so they can change their SIP password through a special number :-P
01:16.48CCFL_Man2can the wcte11xp driver use the new te205p?
01:16.55SteveTotarocool, i don't give them options like that
01:17.09rkeeneI want to be out of the business of dealing with phones
01:17.22rkeeneSo I want to give them as much control as possible :-P
01:17.27SteveTotarothen you will be more in the business when they blow it
01:17.44SteveTotarowhat's my SIP password???
01:17.46rkeeneI already pull down all the user information from the LDAP server, hee hee
01:18.07SteveTotaroah, you are ahead of the curve
01:18.09rkeeneThey can change it from someone else's phone, using their VM password
01:18.28rkeeneAnd the documentation for this will be available... maybe even through the mini-browser ! :-P
01:18.58*** part/#asterisk Trevor_b (n=tbenson@69.12.220.201)
01:19.05fujinIs there anyway to 'automon' mixmonitor?
01:19.13SteveTotaropoint them to sharepoint?
01:19.24tainted_can someone help me with DTMF?
01:19.35SteveTotarorelaxdtmf=yes
01:19.45SteveTotarodtmf=rfc2833
01:19.58rkeeneSharepoint ?
01:20.22tainted_SteveTotaro relaxdtmf is for tdm stuff
01:20.36SteveTotarothought you were an M$ shop, sharepoint is like a intranet
01:20.43rkeene(All our documentation is generated using LyX, and available in HTML, PDF, or dead-tree format)
01:21.02rkeeneNo... We use Slackware for almost everything.
01:21.11SteveTotarowell you were not very specific with your question tainted
01:21.12rkeene(Well, and Cisco IOS and Foundry OS)
01:21.25rkeene(We're the network operations group)
01:21.33tainted_SteveTotaro one user is complaining that dtmf doesn't work
01:21.41tainted_i am using g729 / rfc2833
01:21.49tainted_and it works for everyone else
01:22.12riddleboxtainted_, I always have to use inband
01:22.25tainted_but 729 doesn't support inband :(
01:22.30fujinuh, so, I found this bug http://bugs.digium.com/view.php?id=10185 re: automixmonitor
01:22.33riddleboxahh
01:22.35fujinbut I can't see that in my svn checkout
01:22.37fujinhas it gone into 1.6 isntead?
01:23.11tainted_riddlebox do you ever find out what causes rfc2833 to work for some but not for others?
01:23.22*** part/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net)
01:23.39riddleboxtainted_, nope, I just know with my phones inband always works
01:23.56tainted_riddlebox hard or soft phones?
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01:24.04riddleboxhard
01:25.32SteveTotarohey riddlebox, ever get a chance to dig deeper into that project, just curious
01:26.06mipsterwhat's the other dtmf option?  notify?
01:26.26riddleboxSteveTotaro, nahh I have been real busy and havent had a chance to do much
01:26.38mipsterinfo
01:27.15mipstertainted, any chance that users set has dtmfmode set to something wonky?
01:27.26SteveTotarogotcha, well keep me informed, i plan on making it live to the public this month
01:27.52riddleboxcool
01:27.56riddleboxcongrats
01:28.57fujinAnyone know if it's possible to use ODBC storage to write to one database, but read off another one, for the same thing?
01:29.11fujine.g.; for voicemail, I want it to write to an upstream db which is replicated locally, but read off the local one
01:29.36CCFL_Man2http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=290213819512 <--my new asterisk phone
01:29.56mipsterdoes it support G.729a?
01:30.12mipsterdtmfmode=rotary
01:30.14CCFL_Man2no, just pots
01:31.01SteveTotarothey need a cover pump for their above ground
01:31.06outtolunci'd like to see you *clip* that to your ear <G>
01:31.26CCFL_Man2SteveTotaro: they do
01:31.53outtoluncthat pool cover looks about ready to cave in also <G>
01:31.56mipsterSo will an FXS car recognize pulse dialing?
01:32.08mipsters/car/card/
01:32.09SteveTotarothe silliest thing is the "air pillow" in the middle
01:32.20SteveTotaropulse is supported
01:32.39SteveTotarothey sell those airpillow like it keeps water off the cover or something
01:33.13SteveTotaroat best it just pools the water more to the sides
01:33.23SteveTotaroi prefer looploc
01:36.18outtoluncponders offloading the extra 40 pounds worth of stuff in my backpack so i do not have to hump it around at von <G>
01:38.41mipstergood evenin' folks...
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01:38.52*** part/#asterisk mipster (n=mipster@75.131.201.166)
01:53.39CCFL_Man2i can't find the subset i want
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02:09.30really_phukthallo?
02:10.45really_phuktlmadsen: hey man, I was just reading section of your book about templates for conf files
02:10.58really_phuktdoes this apply to 1.2 version too?
02:11.46lmadsenreally_phukt: maybe... I haven't used 1.2 in almost 2 years
02:11.54lmadsenI was an early adopter of 1.4, around 1.4.0
02:12.07lmadsenyou could try it and find out....
02:12.21*** mode/#asterisk [-o file] by lmadsen
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02:12.39outtoluncaaaaattack
02:12.46lmadsenindeed :)
02:13.05really_phuktlmadsen: I could try it... mess it up and get fired... at least I will know ;)
02:13.23lmadsenreally_phukt: so you have absolutely no test servers then... you should certainly be fired then
02:13.48lmadsenhow appropriate of a name to be doing testing on your production servers
02:13.58really_phuktLOL
02:14.13lmadsenI don't remember adding a smiley face to the end of my sentences
02:14.27jblackNo. The company he works for would be really_phukt.
02:14.40lmadsenah I see
02:14.58*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
02:15.05draygonjblack, arent you also on DALnet?
02:15.13drmessanoMust be a dot-com
02:15.17jblackNo, I am not on dalnet
02:15.20drmessanoLike overnightoothbrushes.com
02:15.23drmessanoor
02:15.36drmessanoelectrickazoofantasticandthemonkey.com
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02:16.45jblackI think dundi could use an accounting system like gnunet has. That could render dundi useful
02:18.28JTdundi seems like a lot of hype
02:19.56jblackwell, it's kinda unusable in any sort of public manner.
02:20.27jblackBut I think a trust accounting concept (which is poorly named), could make it rather usable
02:20.45JTenum seems to be where things are going
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02:21.40jblackYah. dundi hasn't been much of a competitor for enum (which has its own set of flaws)
02:22.10jblackanyone curious to hear more about  trust accounting?
02:22.52*** mode/#asterisk [-o lmadsen] by ChanServ
02:27.19drmessanoWell
02:27.26drmessanoOne person deopped and two more quit
02:27.31drmessanoI'll take that as a "no"
02:28.20jblackYeah. That's why I kept my mouth shut. :)
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03:07.15Iamnachotest
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03:10.31ciphercasthey guys
03:10.49ciphercastanyone set up a copy of 1.6b5?
03:11.23ciphercasti'm transitioning from 1.4, and i am trying to diagnose a problem
03:12.02ciphercasti want to make sure my problem is *not* a bug
03:12.28ciphercastfor the Realtime configuration, is the Goto syntax still the same?
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03:20.38*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)
03:20.52ShadowHntrgot a question re: zaptel/digium card support
03:21.08ShadowHntris the kernel code for x86 only, or would the module work (when compiled properly) on other architectures
03:22.39ShadowHntron linux
03:22.48ShadowHntri just acquired some old PPC hardware and thought about asterisk :)
03:23.44ciphercastI'm sure you can compile the driver for a different arch
03:24.18ShadowHntrcool cause last time i looked (i probably misread it) is that the support was for linux x86 only
03:24.27ShadowHntri'd like to try it on linuxppc
03:24.35ciphercastdigium uses the zaptel driver
03:25.07ciphercastyeah...hmm (digging thru ML's)
03:25.43shido6good luck with that
03:26.12ShadowHntrthx
03:26.13ciphercastwhat I would try doing is compiling zaptel on the ppc unit
03:26.26ShadowHntryeah i'm gonna probably try out yellow dog and compile a new kernel
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03:26.34ShadowHntri just acquired a free Power Mac G3  300MHz unit
03:27.20ciphercasti got a g4 lying around with a busted psu
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03:28.07ciphercastits pretty quiet in here today
03:29.57ShadowHntrciphercast: can i have it? ;)
03:29.58ShadowHntrj/k
03:30.28ciphercastlocation? :)
03:31.20ciphercastShadowHntr: unless you're in the nyc metro area, its not worth you're trip ;-)
03:31.26*** join/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu)
03:31.47ShadowHntrciphercast: TN/Nashville here
03:31.52ShadowHntrthough i do have a friend in DC... :()
03:31.53ShadowHntrer
03:31.53ShadowHntr:)
03:31.58husimonHey does anyone know how to modify the voicemail so that if someone checks the voicemail but hangs up before they finish listening to it that the wmi won't go off?
03:32.09husimonI have a group of people that share a line and they need to see if a message is for them or not
03:32.30husimonthe old system I had let you call in and listen to the message and if it wasn't yours hang up and it will count it as a new message
03:32.57ciphercasthusimon: first off, good question.  I'm not entirely sure, but can you give them diff vm boxes?
03:33.06[TK]D-FenderHasn't ceased to amuse me how so many masochists drag out decrepit non-compatible gear and keep beating away at it in some vain attempt to redeem crap and save a buck.
03:34.14husimon[TK]D-Fender: ya I'm tossing all the old sccp phones in a few weeks
03:34.24ciphercastwell, asterisk has been running fine @ home on my '97 era 600mhz ibm celeron
03:34.30[TK]D-Fenderhusimon, those HAVE a prayer.... you might not, but they do :)
03:34.43[TK]D-Fenderciphercast, See, its at least x86 ;)
03:34.44ciphercastlaughs in amazemet
03:34.51husimon[TK]D-Fender: well I have a few where all the softkeys don't work
03:35.25[TK]D-Fenderciphercast, And I was buying C466's at the office in '99.... I think your calendar needs some adjusting :)
03:36.09husimonwould be nice to be able to mark messages as new
03:36.48husimon[TK]D-Fender: any idea if that is possible for a voicemail message?  or where I should look to find out
03:37.32[TK]D-Fenderhusimon, Have you actually LOOKED at the files?
03:37.44husimonthe voicemail data files?
03:38.09ciphercasteh, this is true.  must be the guinness talking ;-), prolly round '01 then.  all i know is that box has been running asterisk since 1.0...purring in the basement
03:38.10ccvpwell this is weird, I just got a job offer to go from $20/hour to $35 hour doing something
03:38.15ccvpi dont even have experience doing
03:38.20husimonccvp can you learn it ?
03:38.27ccvpgoing from network engineering, to doing just backups in a unix environment
03:38.27husimoni think that's the real question :P
03:38.29ccvpat some huge defense company
03:38.34ccvpafter they do missile firing simulations
03:38.54husimondo you have unix experience?
03:38.56ccvpall this will be is backing up that data using some unix stuff, premade scripts, no knowledge necessary except being "an all around generalist"
03:38.56ccvpheh
03:39.17ccvpjust for recreation at home, 5 years
03:39.17ccvpim ccna, ccvp
03:39.18ciphercastthinks anything Un*x is never *kust* lol
03:39.25ciphercastugh, just
03:39.25husimoni don't see why you wouldn't take it then
03:39.34ccvpwell its something ive never done, its just weird
03:39.39ccvphow this friend of mien can get people in this easily
03:39.41outtoluncnotes The previous reload command is slow as hell, and i wish it would hurry up! <G>
03:39.50ChrisTSISAre there any known software/kernel version issues with the VPMADT032 that causes it to act more like echo suppression instead of cancellation?
03:40.22ccvpwould you ditch yoru fortay of network engineering to go into a new area of IT, that is practically double the pay
03:40.42ciphercastccvp: I would do it, as long as initally its not ttl
03:41.07ccvpi told him im not a engineer w/ EE degree or CS
03:41.24ccvpim just 3 years of MIS degree, with 1 year left, but aparently his boss is wanting people with business experience
03:41.56ccvphe was like so what, you'd be surprised how many we have here that are like that, because their all contract jobs
03:41.57husimonccvp i'd sure consider it unless you see yourself getting a big raise soon
03:42.04ccvpgovernment pays them massive amounts for the position
03:42.11ccvpthen they hire someone like me cheaper, while the pay is still high in my eyes
03:42.19ciphercastyes
03:42.49husimonis it contract or yearly or how long will they hire you
03:43.26ciphercastevery time they fire a missle...heh
03:43.42ccvpit's testing for for the GMD Program (Ground-Based Midcourse Defense), a DoD Missile Defense contract.
03:43.54ccvpthey'll get me a secret clearance too
03:43.59ccvpso thats massively good for my future
03:44.25ccvpthats a paste from his email
03:44.27husimonif you are going to do more defense work
03:44.50ccvpwell lets just say im sick of 20/hour
03:45.03ccvpfor 4 years of what i do, this job got me at a cheap rate years ago
03:45.07ccvpand cant give a big jump
03:45.42husimonccvp i'm sick of 27
03:45.44ccvpso you say why im confused, going from network engineering to wtf? a unix backup specialist?
03:45.51ccvps/say/see
03:45.59ccvpno clue what the title would be
03:46.25husimonccvp you neec ccie to make the big bux with cisco
03:46.30*** join/#asterisk _alex_df_ (n=Alex@dsl-200-67-125-45.prod-empresarial.com.mx)
03:46.36ccvpwell that is beyond my brain
03:46.40ccvpits to much thinking/studying
03:46.49ciphercastthat's a hell of a title
03:46.50husimonat least that's what I would think would be the big bux
03:46.59ciphercast*so* much work
03:47.03ccvpscrew it
03:47.09ccvpim already probably 3x underpaid what i do now
03:47.10husimoni'd say go for it
03:47.12ccvpthing is, i haven o degree
03:47.23ccvpfamlyfriend got my current job, thats why im crazy low hourly rate
03:47.24husimonyou can always use that degree again in a few years
03:47.32*** part/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net)
03:47.35husimonjust take a refresher recourse
03:47.41husimonrecourse = course
03:47.42ccvpim 20/hr, i should probably be at 45/hour
03:47.47ccvpdoing call manager daily
03:47.59ccvpim told at this big company when degree is complete
03:48.04ccvpthey give a substansial boost
03:48.07ccvplike 25-30%
03:48.20ciphercastand you have 1 yr left?
03:48.23ccvp1.5 about
03:48.33ccvpMIS, its scaled down comp sci basically
03:48.45ccvpwith 3 levels of stat classes, some accounting classes, databases, business
03:49.04ccvpmixed with programming here/there, and management theory classes
03:49.08ciphercastand the GMD program lasts how long?
03:49.11husimoni'd take the new job and complete the MIS degree
03:49.24husimonyeah that's what i was wondering, if it's only 6 mo
03:49.27husimonyuck
03:49.37ciphercastthen go back after the contract expires :)
03:49.47ccvpdoubt that'll happen, these are questions to ask
03:50.20ciphercastgetting that security clearance is completely worth it
03:50.40ccvphe says if i get a offer and i accept
03:50.40husimonya i'm pissed because all the jobs here that are worth a crap require you to already have  a security clearence
03:50.43ccvpthey get me that
03:50.46husimonso how the hell do you get one
03:50.54ccvpwell u apply for a job
03:50.55ciphercastits hard
03:50.56ccvpthat needs one
03:51.04ciphercast& | expensive
03:51.07ccvpand they agree to get you one
03:51.07husimonbasically they probably mostly hire inside people
03:51.09ccvpcosts about 10k
03:51.13ccvpfor the company
03:51.28*** join/#asterisk Mavvie (n=edwin@ppp121-44-54-169.lns10.syd7.internode.on.net)
03:51.30husimonwhich is honestly not that much
03:51.33husimoncompared to your salary
03:51.38ccvpwell my friend told me, if i ever leave this potential one
03:51.41ccvpafter i get the clearance
03:51.48ccvpthe secret clearance = gold in this town
03:51.53ciphercastyeah
03:52.19ciphercast...but they interview every inch of your life
03:52.21ccvpplus the graduate program for MIS at my school
03:52.24ccvpgets you an IA certificate
03:52.35ccvpand the training necessary for CISSP in place of a MIS masters
03:52.49*** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net)
03:52.59ccvpive already seen jobs on monster in town for a siprnet admin
03:53.02ccvpbut needs SCI+cissp
03:53.06ccvp135k+/yr
03:53.14husimondon't count your eggs heh
03:53.21ciphercastlol
03:53.22rkeeneis a SIPRNET admin :-P
03:53.29rkeene(It's not very interesting)
03:53.35JTjust a hint
03:53.36ccvprkeene, so you got sci?
03:53.54JTif you're really interested in jobs with any sort of classification clearance
03:54.00JTdon't talk about them on irc
03:54.01JT:)
03:54.18ccvpwell i see, note taken
03:54.26ciphercastas long as you're bouncing thru tor, you're fine :)
03:54.32rkeeneSIPRNET is Secret, not SCI.
03:55.06JTi know here, going for jobs with certain agencies, you are not even allowed to tell friends and family that you have *applied* for the job or it preclude you from being considered
03:55.22ccvpjt, why you trying to make me paranoid now
03:55.23ccvplol
03:55.28ccvplike i axed my shit
03:55.44ccvpi see what your saying, but that spooked me
03:55.55rkeenehas friends with many agencies and has known about it ahead of time for all of them... since I was notified as part of their background
03:56.04JTi'm not sure what sort of secrecy is involved with the job you're going for, but don't tell randoms what you're going to be doing :)
03:56.43ccvpwell its all done then
03:56.44ciphercastessentially, JT's right...
03:56.49ccvpI know
03:56.51rkeene(Including NSA and IC)
03:57.08ccvpi should of thought about that,but who tells you this
03:57.14ccvpas a start? i wouldnt have thought about that
03:58.18ciphercastanyone mess around with 1.6 yet?
03:58.20rkeeneAnyway, if you do get the job, don't put SCI on SIPR -- it's a spill and a PITA.
03:59.10ccvpso how far back to secrets/Ts's go
03:59.15ccvpin to your life? 5 years, 10,15?
03:59.25mostyrkeene
03:59.26rkeene7, and 15, IIRC
03:59.42rkeene(with periodic refiles)
03:59.45rkeenemosty
03:59.51ccvpi remember about 8 years ago, some military officer of some sort called me one day
03:59.55ccvpasking personal questions about my neighbor
04:00.07ccvpmom told me he does intelligence stuff at some fort back home
04:00.12ccvpso that was part of his BG check?
04:00.38rkeeneUsually the FBI does the background check
04:06.10*** part/#asterisk husimon (n=nhuisman@aeko.IfA.Hawaii.Edu)
04:20.07_alex_df_hello, 1.4 noob but been using asterisk in production since pre-1.0.  Today we put our first 1.4 server online.  This is a SIP to PRI system.  Past 110 or so active calls, core show channels was no longer able to show them all, and no summary at the end.  A few minutes later, * I began loosing my SIP registrations and core show channels would show only a couple and still no summary.  Any pointers where to start debugging this?
04:21.15shido6get openser and leave asterisk setup as a feature server
04:21.23shido6let openser handle your registrations
04:21.43mosty_alex_df_, probably a deadlock, i had similar issues with 1.4
04:22.48_alex_df_mosty, did you go back to 1.2?
04:23.28_alex_df_shido6, had same setup with less hardware working with 1.2, but yeah openser might be the way to go
04:24.07*** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id)
04:25.03rkeeneHas anyone tried YXA.
04:25.10rkeeneErr, has anyone tried YXA ?
04:25.42Kumbanghi guys, how can i get variable channel number , eg 23 from Zap/23-1
04:25.59mosty_alex_df_, yes
04:26.24_alex_df_mosty, was afraid you were going to say that :P
04:27.46Kumbangi want to get just channel number 23 not 23-1 from Zap/23-1 etc
04:27.58Kumbangnumber 1 from Zap/1-1
04:28.03Kumbanghow can i do this
04:28.41_alex_df_Kumbang, might be an easier way to do it, but it can be done running it through cut twice
04:30.45rkeeneecho Zap/23-1 | sed 's@^[^/]*/\([0-9]*\)-[0-9]*$@\1@'
04:30.53ccvpc:\documents&settings\kumbang\my documents\my-list-of-adult-sites-logins-pw's.txt
04:31.00ccvpwaves
04:36.28drmessanofail
04:37.29drmessanoC:\Documents and Settings\kumbang\Documents\my-list-of-adult-sites-logins-pw's.txt
04:38.01drmessanoThat's how you pwn someone
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04:40.12lmadsenfor anyone who reads this... the way to do it this like this:      Set(CHAN_NUMBER=${CUT(CUT(CHANNEL,/,2),-,1)})
04:40.23lmadsenI do that all the time
04:41.11lmadsentoo bad Kumbang didn't stay longer
04:41.42rkeeneOh, I didn't know you meant in Asterisk :-P
04:41.56drmessanoccvp ran him off
04:41.57lmadsenstaying a whole 14 minutes will get you no where
04:41.58ccvpi get the feeling "kumbang" is a native phrase or word in indonesian
04:42.04ccvpand not what us american perverts think it means
04:42.05lmadsenaye
04:42.05ccvphahaha
04:42.09lmadsenexactly :)
04:42.35drmessanokumbang is a providence of bangkok
04:44.21*** part/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net)
04:45.18rkeeneHas anyone tried YXA the Erlang-based SIP router with Asterisk ?
04:45.38JTno, but let me know if you do
04:45.40JTsounds interesting
04:45.59rkeeneI'm not sure I can justify it
04:47.11rkeeneI'll setup a new test bed once I roll the current one into production, and test it then
04:48.14lpmusicwith either 1.4 or 1.6 can you change the announcement to the agent answering a queue to make it play the announcement after you pickup but before you hit pound to accept the call?
04:55.15*** join/#asterisk adorah (n=Michael@87.69.130.248)
05:02.52BobLutzLOL kumbang
05:03.03BobLutzo shoot I just woke up
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05:09.16*** join/#asterisk the_5th_wheel (n=edd@41.223.244.9)
05:11.05the_5th_wheelhi. How can i forward all calls coming in on my zap interfaces with a certain context to a macro?
05:12.01jql[zap-channel] exten => s,1,Macro(foo)
05:12.09jqlor rather, use i
05:12.22jqlup to you
05:13.38the_5th_wheelin wich file would i do that?
05:14.25jqlyou modify extensions.conf to add the desired context, and perhaps you need to modify zaptel.conf to set the default context for incoming calls
05:14.48jqland by that, I mean zapata.conf
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05:21.15mvdkDoes anyone know about what they intend to do instead of applying the codec negotiation patch (bugs.digium.com/view.php?id=4825)?
05:23.52mvdkQwell: You closed issue 4825.  Could you please elaborate on what's going to be implemented instead?
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05:37.56mostymvdk, you can implement it in the dialplan manually in 1.4
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05:42.09CCFL_Man2who was it in here that collected vintage phones?
05:43.19jameswf-homehates phones
05:43.33CCFL_Man2jameswf-home: why?
05:44.18jameswf-homethey are awful little things that awful little people annoy you..
05:45.06jameswf-homeat home I dont answer the phone I let my wife or vm get it
05:45.25jameswf-homeIf you want to talk to me that bad you can email me
05:47.15TJNIImethinks jameswf-home has had a little too much "Irish spirit," if you know what I mean.
05:47.27jameswf-homeI wish :(
05:47.36jameswf-hometoo poor to drink
05:48.30TJNIIYou're not too poor to drink, you're just now willing to drink within your proce range.
05:48.48TJNIIs/now willing/not willing/
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05:53.43mvdkmosty:  Suppose I have two offices, with an asterisk server at each running IAX to each other, and a SIP provider that uses g.729.  Clients on the inside know how to use speex and how to use g.711.  How do I get (a) A client in one office to call one in the other office using speex, (b) A client in an office calling a PSTN line to use g.711 so that it may be translated by the server to g.729
05:56.00mostyi can't tell you how to make the client decide which codec to use, but in your asterisk dialplan you can examine the current codec for a call, then when you forward that call on you can use a sip or iax account that only supports the codec you want to use
05:56.27TJNIIUuh, isn't that the whole point of allow and disallow in both sip and iax.conf?
05:57.04mvdkTJNII: Not quite.  The whole point, TJNII, is to use *different* codecs depending on the destination
05:57.48mvdkIn other words, make the client use a different codec
05:58.00mvdkThe way I found to do it was to use SER
05:58.12mvdkAnd make the clients register with it
05:58.35mvdkAnd when they try to dial, pass to the asterisk server only those codecs that apply
05:59.07mvdkSo at call setup time, trim the offer list to just the ones that support the aim we're undertaking
06:00.16mvdkBut the dial plan doesn't allow you to do what I'm talking about, because by the time the call enters the dial plan, the codec for that leg of the call has already been negotiated
06:00.53mvdkSo the point of using SER in that context was to modify the INVITE request
06:01.17mvdkI consider this use of SER to be a hack at best, though
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06:02.44jameswf-homeconsiders a giant time conditioned dialplan that sends people to a random place that changes every minute.....
06:03.33mvdkconsiders such a dialplan to be a gigantic waste of everyone's time, unless sending one's enemies there....
06:03.47lsf_workIm having problems with quadBRI, I can get calls in, but can't call out. Im wondering if it has something to do with dialplan in zapata?
06:03.51mostyTJNII, no, allow/disallow doesn't allow you to minimise transcoding
06:04.00lsf_workerrr... zaptel.conf
06:04.19mvdklsf_work: How did you put your dialplan into zaptel.conf?
06:04.29mvdkThat would've taken some skill :)
06:06.02mvdklsf_work: You may wish to rephrase your question, it's somewhat hard to parse
06:06.06lsf_workmvdk: sorry, I meant zapata.conf hehe... in my extensions.conf I only have Dial(${TRUNK}/${EXTEN:2})... which gives Dial("Zap/73-1", "ZAP/g1/") in new stack. I have overlapdial=yes in zapata.conf but can't figure out whats going on.
06:06.12jameswf-homeI am going to be touching up and rehosting telemarketer torture.... that will be fun
06:06.23jameswf-home~rephrase
06:06.23jbotIf you feel the urge to repeat your question, you'll get a much better response if you try rephrasing it in different terms, or preferably by providing more information about the problem.
06:07.06lsf_workmvdk: asterisk doesnt wait for the outgoing dial (manual dial from fax). It just hangs up before the fax even starts dialing :(
06:07.30mvdkThat's what you expected to happen?
06:07.42mvdkAsterisk to wait for DTMF from a fax?
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06:08.36lsf_workmvdk: yes? overlapdial?
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06:10.03op3rhello
06:10.04lsf_workmvdk: I`ve tried changing immediate to no also, gives some more delay before it hangs up. But still the same problem. (this has been working on 1.2)
06:10.19op3rdoes anyone knows a good french voip provider that can handle calls made by a predictive dialer?
06:11.43mvdklsf_work: I just looked at http://www.voip-info.org/wiki/index.php?page=zaptelBRI
06:11.48jameswf-home~humor
06:11.48jbot[humor] Q: Why are the streets of Paris lined with trees? A: Because Germans like to march in the shade.
06:12.05mvdklsf_work: I suggest you look into WaitExten
06:12.17jameswf-home~ch6
06:12.18jbotRead about Advanced extensions DialPlans etc.. in Chapter 6 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/
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06:15.05lsf_workmvdk: thank you, strange that it worked on 1.2
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06:16.33tengulrewhere have asterisk gui (digium)?
06:17.22jameswf-hometengulre: could you repet that in the form of a question
06:21.04mvdkmosty: Anyway, a proper way of doing codec negotiation would delay deciding on a codec for both legs of the call until it knows what codecs are usable by both sides...
06:22.58mostymvdk, i only bother with sip client -> my * box -> upstream provider, i optimise on the second leg
06:23.23mostysince i have a list of codecs the upstream supports
06:23.38mvdkYeah, I do that too
06:23.59mvdkI mean, it's basically "always use g.729, as the upstream supports that"
06:24.28mvdkBut I saw no reason to just waste CPU on the server when there's so much of it sitting on the client...
06:25.03mostywhat i do is if the call comes on from the client using codec X, and the upstream supports X, i send it on using X
06:25.39mvdkI see, so you take the decision made a priori and run with it, right?
06:27.39mvdkBut you don't optimise the choice made by the client
06:28.10mostyi restrict what the client can use, then i optimise for minimization of transcoding
06:28.47mostyi have very few calls client->client so i don't bother optimising that
06:29.25mvdkIn other words, not a 2-office situation
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06:31.52mostyno. it sounds like you want to change the codec during a call, which i'm not sure how to do
06:32.11mvdkHardly "during", the second leg isn't set up yet!
06:35.03mvdkoej: What is the team doing about codec negotiation?
06:35.26oejmvdk: I am not up to date, too little I would say.
06:36.15mvdkOK, so basically, my quasi-solution of using SER to modify the INVITE request based on whether the destination is inter-office or external is the right approach?
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06:36.55mvdkOr rather, the only approach to allowing inter-office calls to be made by clients to be encoded in speex?
06:37.06mvdkBut the external provider calls to be made g.729?
06:39.00mvdkoej: Basically, the scenario is thus: Suppose I have two offices, with an asterisk server at each running IAX to each other, and a SIP provider that uses g.729.  Clients on the inside know how to use speex and how to use g.711.  How do I get (a) A client in one office to call one in the other office using speex, (b) A client in an office calling a PSTN line to use g.711 so that it may be translated by the server to g.729
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06:43.41oejmvdk: Well, you know you can negotiate multiple codecs in one call?
06:44.00mvdkReally?
06:44.13oejyes, it's quite common in Asterisk
06:44.50oejSo if your client sets up a call to Asterisk with speex AND g711 - check what codec will be used in the different call scenarious with ethereal. Asterisk may even change mid-call.
06:45.50mvdkSo the client INVITEs the server with both speex AND g711 in the codec list?
06:46.12mvdkAnd the server replies "Go ahead, send me both"?
06:47.32oejYes, that works
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06:47.45oejThen the client and server chooses what they believe is optimal
06:47.55oejYou can experiment with the order to get things right
06:48.20oejAnd when you call *to* the client, you can use the SIP_CODEC dialplan setting to control what you want.
06:48.43oejIt might not get exactly what you want, but it will help your situation
06:49.02oejAnd you might end up with the client sending speex and asterisk sending g.711 in the same call... :-)
06:49.50mvdkSo when the client initiates the call, the server *can* decide to have the client send it speex or g.711, depending on the destination?
06:50.22mvdkI ended up using SER to make the server comply, by trimming the list depending on the destination.  Very naughty, I know :)
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06:52.43mvdkAnyway, it seems to me that proper codec negotiation in asterisk would remove the need for these mind-bending tricks
06:52.56mvdkThey tend to be hard to explain...
06:53.16mvdkI mean, not to people here, but to the people I work for
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07:09.58mostyoej, do you have any further feedback on my manager event patch? 11959
07:10.20oejI haven't checked...
07:10.54mostyhttp://bugs.digium.com/view.php?id=11959
07:12.32oejmosty: You forgot changes to sip.conf... Tss, tss ;-)
07:13.16mostyoej, i'm not sure what you mean?
07:13.33oejIf you add a new config option, one has to update sip.conf.sample too
07:13.38oejI will fix
07:13.43mostyoh i see, sorry
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07:20.26adorahHi where I can find jitterbufer definitions?
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07:23.38oejmosty: Committed.
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07:37.22mostyoej: cool- thanks
07:40.28rkeeneGrr
07:40.35oejmosty: Thanks for reminding me
07:40.38rkeenekicks Erlang !@#$ing buggy software
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07:52.44Nuggetwhat drugs would I have to take that would make erlang make sense?
07:52.47NuggetI really want to know.
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07:57.24bjartischan_sip.c:1245 retrans_pkt: Hanging up call ZmNhOWI5MWI1ODg3ZmYyNjliMWEwOGFmNTNkOWMzYTc. - no reply to our critical packet
07:57.31bjartiswhat does that mean?
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09:04.14zepmantrahello there anyone using astribank,fxs+fxo modules? is it worth buying, can't seem to find reviews
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09:07.18cy3o3What's a good softphone for linux these days?
09:07.42tzafrirSIP? twinkle and ekiga, basically
09:08.26cy3o3Cool
09:08.34cy3o3Thanks dood
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09:14.00shitalhello all
09:15.17agxcy3o3, Zoiper
09:16.06agxi was testing chan_mobile but i get no audio on incoming calls and only 1 seconds of audio during outgoing calls; could it be the phone (Nokia 6210) the USB Dongle or some other stuff? "core set debug 255" didn't helped a lot...
09:17.38cy3o3agx: nice, IAX
09:17.40cy3o3thanks
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09:19.26shital<PROTECTED>
09:22.45shitalcan any body help, is thr any problem using this distro???
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09:29.18agxcy3o3, well with IAX you don't get the pre-ring stuffs too; i suggest to use it on phones only if you have NAT problems
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09:35.13tengulreanybody here come from CHINA?
09:36.02agxno, but i'm from Tibet
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10:02.04Diablushello all
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10:03.35Diablusi have a problem with my asterisk server. There is a 99 Sip-peers. Asterisk drop down every 10-15 minutes without any reason. This is a standard bug?
10:06.41Diabluscan anybody help me?
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10:56.52kombiI'm experimenting with audio codecs, how could I make asterisk play a high quality audio file for testing?
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11:04.13kombiif I crank up audio quality of i.e. speex, will asterisk's stdout still be 8k/8bit?
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11:10.23kombicorydon-76-dig: you probably know the above, just read about your patch for wideband speex..;)
11:12.30tzafrirkombi, g722?
11:12.58kombitzafrir: using speex at the moment
11:13.54tzafriranybody here cross-builds Asterisk on ARM and cares to answer the question in asterisk-embedded?
11:14.02kombitzafrir: I just wonder if, with any codec stdout of asterisk will be any different from 8k/8bit?
11:14.16tzafrirhttp://lists.digium.com/pipermail/asterisk-embedded/2008-March/thread.html
11:14.22kombitzafrir: that is way beyond me..;)
11:14.38wordzillais it possible to tell asterisk to auto-terminate a call if it exceeds, say, 20 mins?
11:14.58tzafrirkombi, what do you mean by "stdout of Asterisk"?
11:15.27tzafrirwordzilla, core show application dial
11:15.32kombitzafrir: like when you pipe it ot another audio device such as a stream client
11:15.46tzafrirwordzilla, I forgot the name of the option, but it is one of them
11:16.24wordzillayup looks like L
11:16.24wordzillathx tzafrir :)
11:18.05kombitzafrir: ..as you can do with app_ices. I wonder if that signal will vary depending on codecs
11:18.44jblackohh man. wondershaper broke.
11:19.55kombitzafrir: never mind, I'll just try it out after lunch..;) Would be great if it did..
11:20.07tzafrirkombi, app_ices... I think that the format_mp3 of Asterisk is for 8kHz only
11:20.58kombithat's right, I patched that to go to ezstream instead. Just wonder if standard output of asterisk ever goes beyond 8k sample rate
11:21.47kombianyway, off to lunch, cheers!
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11:48.24SteveTotarowhat if the fed cuts the rate by a whole point?
11:49.42cpmwhat if?
11:50.11SteveTotarodepression
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11:50.39cpmSteveTotaro, I'm sure, by the nature of the question, that you are pretty aware that this is already the case.
11:51.09cpmwe had a short period, where it was starting to make sense to finally start saving money again, that lasted for what? almost a year?
11:51.12tzafrirSteveTotaro, no need to get so depressed ;-)
11:51.35SteveTotarothe world is going to get really big again
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11:52.30SteveTotaroonly thing that has made sense this last year is ownership of gold offshore, bullion
11:52.49cpmnaw, gold in the mattress was fine too :)
11:52.58SteveTotarobullionvault is my favorite
11:53.14cpmjust can't bring himself to trust such outfits
11:53.28tzafrirSteveTotaro, but gold is much heavier than water. So you'd need a pretty big boat for that
11:53.30cpmpaper gold != gold
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11:53.50SteveTotarono gold in the US, history will tell you the US government seized all private gold and made it illegal to own gold
11:54.17cpmI was telling folks 3-4 years back, that gold would hit $1K per tz by 2010, guess I was overly conservative :)
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11:54.41SteveTotaroi was all about gold a year ago :)
11:54.42cpmSteveTotaro, they only seized gold from folks who were willing to give it up :)
11:55.04jblackpredicting the upward movement of commodities is a stupid pet trick at best.
11:55.13cpmjblack, noted
11:55.47cpmgold isn't exactly a commodity. It's whats known as real money. Folks like to diss that concept, but it has played out quite well thoughout history
11:55.55SteveTotarobut gold is and always has been the best form of money
11:56.54cpmif you were all about gold a year ago, you should be sitting good now. :)
11:56.56SteveTotarountil some alchemist comes up with a way to change atoms from one thing to gold it will continue to hold value over paper
11:57.11SteveTotaroi have a bit in Zurich
11:57.31SteveTotaroor so they say
11:57.36jblackI've never really subscribed to that rule of thought, because once you consider gold an exception, you have to consider other exceptions (diamonds, and particularly other precious metals for example)
11:57.41cpmwell, some folks say, and I think they are not far wrong, it's a tail wagging the dog thing. Gold remains consistant, other values slide against it. Not the other way around
11:57.53cpmdiamonds are a tightly controlled market.
11:58.26jblackheh. Gold isn't?
11:58.31cpmNope.
11:58.37SteveTotarothere is a much more bountiful supply of diamonds, they are just withheld
11:58.38cpmGold *is* the standard
11:58.58cpmit's the baseline currency
11:59.15jblackGold _was_ the standard, in an on-again-off-again basis. And it hasn't been the only standard.
11:59.30jblackEver hear of sterling? :)
11:59.43SteveTotaroany society that has printed money with no backing has failed
11:59.47cpmsure. it's of less value, hence more fungible.
12:00.26cpmThis is all pretty fun, and ultimately, it doesn't matter. It's now about what you can borrow, not about what you have. Has been that way for some time now.
12:00.50cpmbeen a long time since anyone plunked down a nice sack of metal and bought a car
12:01.16SteveTotarobartering is becoming a huge movement
12:01.19cpmbut having a sack of metal or two, is a nice hedge. that has worked out well. I don't think I'd buy any right now.
12:01.19jblackI don't agree with you, for reasons that are non-intuitive and non-obvious, but I can understand you.
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12:01.42jblackI suppose I was a little redundant there
12:01.44cpmjblack, that's cool, and your approach is probably quite likely more practical
12:02.02cpmoh yeah? how about 'probably quite likely' ?
12:02.11SteveTotarojust plot the values over the years
12:02.21jblackOh, definitely certain.
12:02.29cpmchuckles
12:02.33SteveTotaroand then look at the dollar and inflation
12:02.53SteveTotarolook at the dollar pre-gold standard and then now
12:03.19SteveTotaropre-no-goldstandard i should say
12:03.20cpmThe dollar is a fiat currency, like *all* other currencies. Floating away from gold was the best thing that ever happened for the worlds economic prospects.
12:03.37jblackI agree! I thought you thought the opposite?
12:03.40cpmfiat currencies work quite well, allows for open ended gains
12:03.59SteveTotarofiat currencies always fail, it is seen throughout history
12:04.11SteveTotaroi got it, let's print more paper
12:04.24jblackI definitely agree with you cpm. I'm particularly happy with the social effects.
12:04.37SteveTotarowill you be happy in a year?
12:04.52cpmthe downside is, when the press picks up on reporting about 300 folks defaulting on home loans in a market, where a few hundred thousand *didn't* default as some kind of abberation, and folks panic, , , then it starts to get sketchy
12:05.06SteveTotarowe have seen the bright spot in human history, lived it
12:05.53SteveTotarono it is the twilight, the sun is setting
12:06.03jblackcpm: Yeah, I mostly agree with you, yeah. I think there's enough blame to include over leveraging.
12:06.34SteveTotarothe fed is pushing for the amero like they euro
12:06.44jblackWhich I credit for making the runs that you're implying possible in the first place.
12:06.46SteveTotaroor should i say the illuminati
12:06.57cpmSteveTotaro, perhaps. As long as the fed keeps trying to prop up failed institutions, like it's doing right now, and messing about in the market, like they always do, it's going to cause real consequences. It's like using a CC to pay the minimum on another CC
12:07.00jblacklol
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12:07.54SteveTotaroin the end, one central, private bank for the world
12:08.34cpmSteveTotaro, yeah, it's called the WTO/World Bank, and Americans hold all the paper.
12:08.41SteveTotarothe fed can only cut interest rates and it is bottoming out
12:09.05jblackThe fed can, and has already done, much more than that, steve.
12:09.15SteveTotaroI argue China hold all the cards now
12:10.01jblackI'm gonna regret this, but based on one set of logic does "china hold all the cards"
12:10.07jblackon what set?
12:10.13SteveTotarowhat has the fed done?
12:10.38SteveTotarothe fed is group of privately held banks, nothing fed about them
12:10.41tomfmasonI am having trouble make installing zaptel. I have kernel-devel installed(2.6.18-53.1.14.el5) but it is complaining about 2.6.18-028stab053.4 . Any ideas?
12:11.07jblackThey've worked with the two target rates, they've changed the rules for the discount window, they've set up TEF and TSLF, and just this past monday, they guaranteed 30 bill of debt that JP Morgan 'bought' from stearns.
12:11.33jblackOh, and they've been printing a few dozen billion here, a couple hundred billion there.
12:11.41SteveTotaroJP Morgan was a traitor
12:12.03SteveTotaroyeah, that's what i am saying, fire up the presses
12:12.14SteveTotarothat will solve all of our problems, more paper
12:12.44jblackI must have misunderstood you when you said "the fed can only cut interest rates and is bottoming out"
12:13.04SteveTotaroyeah they can create inflation too
12:13.12SteveTotaroby printing paper
12:13.55jblackIt's pretty well agreed that there will eventually be heavy inflationary costs in the future.
12:13.59SteveTotaroall unconstitutional
12:13.59tzafrirtomfmason, what do you mean by "complaining"?
12:14.16tzafrirtomfmason, what is the output of 'uname -r'?
12:14.55SteveTotaroany private organization printing money is guilty of counterfeiting
12:15.07jblackRight now we're looking at an s shaped curve. Short inflation spike, followed by mild-to-severe recession for 1-4+ years (which is deflationary), followed by moderate to high inflation.
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12:15.28SteveTotarohow about straight depression
12:15.34SteveTotaropottersville
12:16.08jblackThat's my belief, but it's not the commonly stated one. Nobody wants to be the first person to start financial panics and bank runs.
12:16.37SteveTotarowhy would the government cut a check to almost everyone for at least $300 and mostly $600
12:16.38tomfmasontzafrir: it shows: You do not appear to have the sources for the 2.6.18-028stab053.4 kernel installed.  and uname -r is 2.6.18-028stab053.4
12:17.12jblackanyways, what the fed is trying to do right now is stave off deep deflation, because of the overleveraged hedge funds that are collapsing.
12:17.16tzafrirtomfmason, do you use distro kernel or your own built kernel?
12:17.49jblackIf they ignored that problem, we'd certainly be in a depression today. That's what they're staving off right now as they create money.
12:18.16tomfmasonI just installed it from yum last night- yum install kernel-devel
12:18.24*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
12:20.18SteveTotarothat is what they want you to believe just as the great depression was intentional
12:24.36*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
12:27.00*** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk)
12:27.20*** part/#asterisk gormux (n=julien@rei.genux.info)
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12:32.36tzafrirtomfmason, hmmm... that is for a newer kernel version than your running one
12:33.29tomfmasonhmm
12:33.32tzafrirtomfmason, do consider upgrading the kernel. keyword for your searches: vmsplice
12:34.09tomfmasonI really don't like idea of manually building a kernel on a remote box but it is not looking like I will have a choice
12:34.20tzafrirthat said, maybe using './install_prereq install' will install the older kernel-devel package for your running kernel
12:35.10tzafrirtomfmason, it's not a matter of building a custom one. It's installing a newer one for your distro
12:36.38*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
12:40.29*** join/#asterisk seanbright (i=seanbrig@65.207.74.18)
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12:50.19*** join/#asterisk warlock_mza (n=warlock@200-112-143-197.bbt.net.ar)
12:50.24warlock_mzahi there
12:51.02warlock_mzaI currently have a pbx that allows with one telefone line handling 4 incoming calls
12:51.15warlock_mzaif I want to put asterisk before this pbx
12:51.33warlock_mzaI mean line goes into asterisk and via an fxs -> old pbx
12:51.48warlock_mzacan I still redirect 4 calls with one fxs card ?
12:51.49*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl)
12:52.07warlock_mzaand let the old pbx handle the 4 of them
12:52.08warlock_mza?
12:52.16[TK]D-Fenderwarlock_mza: If you have 4 analog lines going into your old system, you'll need 8 ports in your * server
12:52.25[TK]D-Fenderwarlock_mza: 4 FXO, 4 FXS
12:52.40[TK]D-Fenderwarlock_mza: What do you want * to do for you?
12:53.56*** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com)
12:55.12*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
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13:05.07warlock_mzaok got it ...
13:05.16warlock_mzayou gave me the explanation I needed
13:10.37warlock_mza[TK]D-Fender, does that kind of config come on a single pci ?
13:10.48warlock_mzado you recomend something particularly ?
13:11.18*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:12.05[TK]D-Fenderwarlock_mza: Digium TMD800P or Sangoma A200d
13:13.29warlock_mzacool
13:13.54Kattyhewwoes.
13:15.19[TK]D-FenderKatty: Mew.
13:17.22*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
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13:33.17jblackI'm bored.
13:33.23jblackconsiders setting up a phone sex line for dogs
13:35.14MatBoywonders why he doesn't have his 3,3V cards PCI boards yet :P
13:35.36jblackwho is john galt?
13:35.50MatBoydunno, ask him
13:36.46jblackWho is Bart Simpson?
13:37.46jblackis so glad you didn't answer that one
13:38.14MatBoyhehe, Bart Simpson is Bart Simpson :P
13:38.25jblackdamn.
13:38.31jblackYou need to read more, and watch TV less.
13:39.10MatBoyjblack is jblack
13:39.14MatBoyis MatBoy
13:39.34*** join/#asterisk anonymouz666 (n=anonymou@201.19.227.137)
13:39.56jblackNot knowing who John Galt is, is like not knowing who Guy Montag is.
13:40.20MatBoynah, I'm busy with pricing... I don't want to think that much about other things :)
13:41.27*** join/#asterisk Newbie___ (n=Newbie__@213.8.50.60.kmr02-home.tm.net.my)
13:41.59Newbie___morning all, how do i do a dial command after hangup ?
13:44.24coppicewho is Guy Montag?
13:45.18*** join/#asterisk ccvp (n=ccvp@66.0.46.210)
13:46.29ccvp- hello fellow internet addicts - are we all looking forward to another long & glorious day of irc/internet addiction :)
13:49.29cpmcoppice, he's the hero in bradbury's fahrenheit 451
13:49.57coppicewell, its an awful long time since I read that :-\
13:50.09cpmheh
13:50.26cpmtook me a few moments
13:51.47*** join/#asterisk putnopvut (n=putnopvu@216.207.245.1)
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13:51.49coppiceI remember reading the bit about buildings being fire proof, and thinking they had the same idea bout the Crystal Palace
13:52.00jameswf-homewhats the term for bridging to sip callers then dropping out...
13:52.30ccvpis ekiga worthless?
13:52.35ccvpwhats a better client to get (free)
13:52.42jameswf-home~ekiga
13:52.43jbot[~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org
13:52.54jameswf-homewelll
13:53.05ccvpI know im using it, i did an ekiga to ekiga this morning on a 1GB lan
13:53.13ccvpboth with logitech quickcam pro9000 cams
13:53.18ccvpand the vidoe acceleration is garbage
13:53.22ccvpeven though the camera is high end
13:53.23*** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca)
13:53.36ccvpin windows, the video acceleration is near perfect in net meeting or skype
13:53.44ccvpwondering if ekiga uses crappy codecs by default for video
13:54.36*** join/#asterisk axisys (n=axisys@155.70.141.45)
13:54.41ccvpskype video conference w/ dual quickcam pro 9000's gave high quality , high def, near 30fps...ekiga gives randomized blocks, and squarey smears
13:57.12*** join/#asterisk quigon (n=matias@32.59.64.130)
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14:00.12jameswf-homelmao http://www.bash.org/?117002
14:01.02*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
14:01.02cpmchuckles
14:01.46dandreb
14:01.53dandre~pb
14:01.53jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste
14:02.42ccvpI dont think ive ever laughed
14:02.47ccvpat spam before, but just got this on efnet
14:02.48ccvplol
14:02.50dandreHello,
14:02.52ccvp<linuckzG> (SANDEEP PEHMDAYPEEZ) C1a_!5 TaB5 for $39.99.............www.pharma-ceuticals.com . . . . DONT THINK TWICE before YOU GO. (SANDEEP PEHMDAYPEEZ)
14:03.12ccvpwhat the hell is a sandeep
14:03.49KattyAnyone know how to make FOP show DID numbers?
14:04.02Kattythey don't really have a set zap line they come in on.
14:05.39*** join/#asterisk stoffell_h (n=stoffell@fw.catsanddogs.com)
14:05.46dandreI have put here part of log and dialplan: http://pastebin.ca/947402
14:05.46dandrewhen 4321 is dialed, I was expected to try extension 13, then on timeout extension 15 and then 16.
14:05.46dandrebut I have a hungup just after 13. Why?
14:09.07SteveTotaronot why local channel and i do not see a timeout
14:10.46SteveTotarotrixbox?
14:11.40*** join/#asterisk tcseke (n=chatzill@217.20.134.239)
14:13.05dandre<PROTECTED>
14:14.07dandreI use Local because I want to make my dial not dependant on technology at this point
14:15.06*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
14:15.15Kattyanyone know how to use DID numbers with FOP?
14:15.22*** join/#asterisk ciphercast (n=cipherca@pool-151-204-63-64.pskn.east.verizon.net)
14:15.38dandreit is not trixbox either
14:18.52*** join/#asterisk djs26 (n=djs@unaffiliated/djs26)
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14:24.29*** join/#asterisk BCS-Satori (n=somewher@75.148.21.113)
14:25.09UserReg_CLhi.... what port use xlite whe connect to asterisk ?
14:27.32BCS-SatoriHow do you pass the originating CID/CNAME from an external caller to the transfered caller?  When a receptionist answers the phone call they see the incoming callers cid/cname, but when the receptionist transfers the caller to extension 101 for example, the person on extension 101 has their callerid as the receptionist not the incoming caller for the rest of the call. This issue is on multiple asterisk's rev 1.4.11-1.4.18 w/ polycom and linksys.
14:27.35lirakisis away (leaving..."the internets" are safe ... for now)
14:27.59n0cturnQ: I'd like to hack around with Asterisk on a Virtual Machine. Would a pair of Zoom 5801's be appropriate for VoIP gateways, or is there a better product to use ? I would prefer lower cost hardware and I do not want to put in a PCI card.
14:28.03[TK]D-FenderBCS-Satori: Tell her to do BLIND transfers instead of ATTENDED
14:28.17[TK]D-Fendern0cturn: Zoom = bleh
14:29.28BCS-Satori[TK]D-Fender: that fine, but is there a way to send it with an attended ?
14:30.55n0cturnCan anyone suggest a low-cost 1-port FXS/1-port FXO VoIP gateway ?
14:31.47*** join/#asterisk axisys (n=axisys@155.70.141.45)
14:32.05[TK]D-Fendern0cturn: Linksys SPA-3102
14:33.11n0cturnThanks, D-Fender
14:33.41dandreI have put here part of log and dialplan: http://pastebin.ca/947402
14:33.41dandrewhen 4321 is dialed, I was expected to try extension 13, then on timeout extension 15 and then 16.
14:33.41dandrebut I have a hungup just after 13. Why?
14:40.45shasta[TK]D-Fender, by the way. does spa3102 handle sip register?
14:41.20shasta(i'd like asterisk to register on spa3102)
14:43.52[TK]D-Fendershasta: Yes.
14:43.58[TK]D-Fendershasta: IT registers
14:44.40agxwith asterisk 1.4 is possibile to send a call out via a specific channell of an MISDN port ? I've a GSM box connected via ISDN with 2 SIM on it
14:45.00shasta[TK]D-Fender, I know that _it_ registers, but I'd like Asterisk to register at spa :)
14:45.06*** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com)
14:45.29[TK]D-Fendershasta: don't think I saw an option for that... might be... go look.
14:46.10*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
14:46.10*** mode/#asterisk [+o russellb] by ChanServ
14:47.04BobLutzWhat type of files am I able to stream with app_read.c ?
14:47.18BCS-SatoriIs there any docs that explain how to block/route a call based on the caller id number
14:47.19BobLutzis looking at the API call to ast_app_getdata() on line 189
14:48.10dandreis there a backport of the 1.6 hint() function for asterisk1.4?
14:48.59[TK]D-FenderBCS-Satori: "core show function CALLERID"
14:51.25russellbdandre: no, but give me one sec ...
14:51.40[TK]D-Fender~devstate
14:51.41jbot[~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/
14:52.42russellbok, backport done ..
14:53.11russellbwait, that was the wrong module, lol
14:53.19warlock_mzadoes anyone know some kind of adaptor from rj45 -> rj11  voip->analog ?
14:53.29warlock_mzato avoid having so much fxs ?
14:53.33Rienzilla+ons
14:53.42RienzillaI have a linksys app which does that
14:53.46dandrerussellb: my problem is not directly related to the hint function:
14:53.48warlock_mzaand dont have to throw away all my phones ?
14:54.16dandreI have put here part of log and dialplan: http://pastebin.ca/947402
14:54.16dandrewhen 4321 is dialed, I was expected to try extension 13, then on timeout extension 15 and then 16.
14:54.16dandrebut I have a hungup just after 13. I don't understand why?
14:55.21dandreif I change the dial(...) by a stdexten macro call, this seem to work correctly so as I haven't found solution, I was trying something like
14:55.39*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
14:55.59dandreMacro(stdexten,13,${Hint(13@from-internal)})
14:56.11dandrebut I am in 1.4
14:56.50*** join/#asterisk bmg505 (n=leon@196-209-78-68-tbnb-esr-2.dynamic.isadsl.co.za)
14:56.53*** join/#asterisk twitchnln (n=raleigha@cpe-orncorp.dktc.atl.oneringnetworks.net)
14:57.13*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)
14:57.23twitchnlngood morning, how do I set the maximum channels on a per trunk basis?
14:57.37russellbwhat kind of trunk?
14:57.42twitchnlnsip
14:57.44russellband are you using freepbx?
14:57.46twitchnlnno
14:57.48russellbok
14:57.53russellbthen you can set call limits in sip.conf
14:58.04russellbsearch for limit in configs/sip.conf.sample
14:59.59tomfmasonfinally compiled zaptel on centOS 4
15:00.41_ShrikErussellb: where can I find that hint backport you just did :)
15:01.10russellbi didn't do the right module, heh
15:01.17russellbi did func_extstate
15:01.27twitchnlnrussellb: thanks
15:01.28_ShrikEYeah I see that.. 2 minutes
15:01.32russellbbut it's in ... svn co http://svncommunity.digium.com/svn/russell/asterisk-1.4
15:01.35_ShrikE2 minutes old that is
15:02.47*** join/#asterisk xacatecas (n=jkroon@dsl-241-143-116.telkomadsl.co.za)
15:03.17xacatecashi, has anybody played with the linksys wip330 phones?  specifically I'm unable to get WPA2 working.  Any pointers would be appreciated.
15:05.37dandrerussellb: do think a backport of the hint function could solve my problem?
15:05.37dandreI'd prefer understand why the Dial(Local/13@from-internal) doesn't return
15:06.02[TK]D-Fenderdandre: Might be a thought to put the "/n" in there so it doesn't bridge them .....
15:06.09[TK]D-Fender(hand-off)
15:06.22russellbdandre: i'm not sure .. i got busy with something else, sorry
15:06.23*** join/#asterisk jpeeler (n=jpeeler@216.207.245.1)
15:10.21xacatecasok, so i take it there are no-one using a wip330 here ...
15:11.25*** join/#asterisk P4C0 (n=jannet@200.124.22.34)
15:12.24dandreI have put the "/n" but the result is the same
15:12.30P4C0hello everyone, i bought two license for g729, it was working fine but now all of the suddent it doesn't and show translations shows - for all the g729... how can I check the status of the codec?
15:13.32Newbie___morning all, how do i do a dial command after hangup ?
15:14.11*** join/#asterisk op3r (n=Op3r@203.177.177.26)
15:15.46P4C0is there anyway to check the status of g729 codec?
15:15.54russellbshow g729
15:15.55russellbi think
15:15.57russellbat the *CLI>
15:17.42BobLutzWhat type of files can app_read.c use ?
15:17.58[TK]D-FenderBobLutz: "CORE SHOW MODULES LIKE FORMAT"
15:19.25russellbO.O
15:19.33russellb[TK]D-Fender: i didn't know you could yell at the CLI!
15:19.35russellbthat's cool
15:19.35BobLutzmodule show like format
15:20.11*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
15:20.11*** mode/#asterisk [+o anthm] by ChanServ
15:20.16BobLutzso app_read can use anything that `module show like format` returns ?
15:20.53BobLutzapp_read.c uses an API call to ast_app_getdata(), I assume the format type is transparent
15:21.00Kattyanyone know how to use DID numbers with FOP?
15:21.05twitchnlnquestion, will this fail over to the second trunk if i enable it?
15:21.05twitchnlnexten => _X.,1,Dial(SIP/trunk1/${EXTEN})
15:21.05twitchnlnexten => _X.,2,Dial(SIP/trunk2/${EXTEN})
15:21.05twitchnlnexten => _X.,3,Macro(out-congestion)
15:21.06*** join/#asterisk ccvp (n=ccvp@66.0.46.210)
15:21.23ccvpwhat's this caused by
15:21.24ccvpryan@myubuntu:~/Desktop/xlite$ ./xtensoftphone
15:21.25ccvp./xtensoftphone: error while loading shared libraries: libstdc++.so.5: cannot open shared object file: No such file or directory
15:21.35ccvpafter i chmod +x'd the binary
15:21.55shastainstall libstdc++ :-)
15:22.05twitchnlnccvp: looks like you need libstdc++
15:23.13ccvpE: Couldn't find package libstdc
15:23.14ccvpryan@myubuntu:~$
15:23.21ccvpwhts the package name?
15:23.24*** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net)
15:23.30BobLutzccvp: packages.ubuntu.com
15:23.42twitchnlnccvp: apt-get install libstdc++
15:23.44BobLutzIf I remember correctly..
15:23.51ccvpthats what i did twitch
15:23.58ccvpmust be named diff or osmething.
15:24.06P4C0russellb, it says that show command is invalid... i will try to reinstall the codec
15:24.17twitchnlnccvp: apt-cache search libstdc++
15:24.20shastaapt-cache search libstdc++
15:24.31shastabut this is definitely NOT related to Asterisk
15:25.29ccvpi know shasta, but cant find the info on google, searched past 35min
15:25.43ccvpapt-cache search libstdc++ gave back like 40 results
15:25.55BobLutz~ubuntu
15:26.12km-[TK]D-Fender: what's your opinion on the sangoma's hardware echo canceller?
15:26.19[TK]D-Fenderkm-: Great
15:26.37shastaccvp, Results 1 - 10 of about 948 for ubuntu libstdc++.so.5: cannot open shared object file. (0.28 seconds)
15:26.40km-[TK]D-Fender: gonna try popping two of those cards into a dell 2850 chassis and see what asterisk does with it.
15:26.41ccvphttp://pastebin.com/m95dd32
15:26.44ccvpthats my output
15:26.53[TK]D-Fenderkm-: which card?
15:29.28ccvptwitch, it runs now
15:29.33ccvpapt-get install libstdc++5 worked
15:30.19km-[TK]D-Fender: the sangoma 8-port. a108d?
15:30.31[TK]D-Fenderkm-: ok.
15:30.32km-[TK]D-Fender: apparently we use them elsewhere in our company too
15:32.15lirakis_awayis away (leaving..."the internets" are safe ... for now)
15:32.17*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
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15:38.09BobLutzrussellb: ast_write() writes a frame to a channel (i.e - you can essentially play a .wav over a channel), right ?
15:38.34russellbyeah, sort of
15:38.40russellbast_write() is the lowest level API call
15:38.46russellba frame is not a file
15:38.49dandreis there any reason why my Dial(Local/13@from-internal/n) doesn't return from 'from-internal' context?
15:38.54russellba frame is like a packet of audio ... 20 ms usually
15:40.00BobLutzrussellb: Is it possible to write a file to a channel?
15:40.00russellbyup
15:40.00russellblook in include/asterisk/app.h
15:40.00BobLutz:-D
15:40.00BobLutzdanke
15:40.00russellbtons of examples of usage in apps/
15:40.05russellbfor example, app_voicemail.c plays a ton of files ...
15:40.09russellbthough that's a pretty big app
15:40.16russellbapp_playback is a simpler one
15:40.17BobLutzgeusses app_playback.c might be good?
15:40.19BobLutzyea
15:40.20russellbnods
15:40.22BobLutzok great, thanks
15:40.24russellbnp
15:45.56*** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
15:50.13*** join/#asterisk patrick-- (i=patrick@devnull.biz)
15:51.02patrick--Hey, im using asterisk with mISDN and Beronet ISDN cards. when 2 phones are connected to a port can i set an outgoing number for both of them?
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15:55.44really_phuktLooking for respectable and reliable IAX providers for small and large business. Who would you recommend?
15:57.31dandreif I put 'return' just after the stdexten macro call and use gosub instead of dial(Local...) this works.
15:57.31dandreis this a good idea? russellb, [TK]D-Fender ?
15:58.45really_phuktIs *any*body here using IAX providers or yall pretty much stuck with SIP?
16:00.07really_phuktIs there anybody awake this morning? (or whatever time applies to you)
16:00.51cpmuses IAX provider(s)
16:01.07[TK]D-Fenderdandre: Perhaps you should do "/n" and not use "return"....
16:01.26cpmthe values for respectable and reliable don't have the same meaning in telcom as they once may have. THerefore I don't feel qualified to give an answer
16:01.55[TK]D-Fenderreally_phukt: Only people using IAX for ITSP's are those in dire need of bandwidth or with firewall issues.
16:02.17really_phukt[TK]D-Fender: ...firewall... that would be me :(
16:02.22dandre[TK]D-Fender: I have put Dial(Local/13@from-internal/n)
16:02.22dandreis this ok?
16:02.38dandrebecause that doesn't works
16:03.00[TK]D-Fenderdandre: pand the second part...
16:03.14[TK]D-Fenderreally_phukt: Is yours really screwed up?  What are you running?
16:03.18*** join/#asterisk svenna_ (n=svenna@p548D357E.dip0.t-ipconnect.de)
16:04.27really_phukt[TK]D-Fender: well, I am using SIP provider now (callcentric) and works OK. But IT guys is not happy with the amount of open holes in the firewall...
16:04.49[TK]D-Fenderreally_phukt: then downtune your RTP range and tell him to get over it.
16:05.09dandre[TK]D-Fender: I haven't understood
16:05.23[TK]D-Fenderdandre: "remove the "return""
16:05.33[TK]D-Fenderdandre: jsut let "s" run out like normal.
16:06.04dandrethat's what I have done
16:06.32really_phukt[TK]D-Fender: I wish I could. I've been fighting this battle for a while. Are there any major advantages of SIP over IAX that I can stress to try to convince him?
16:07.44*** part/#asterisk jivco (n=jivco@85.187.217.6)
16:07.49really_phuktfrom what I read IAX seems to be better for NAT environments and limited bandwidth
16:07.54[TK]D-Fenderdandre: and -- Local/13@from-internal-5273,1 answered Zap/2-1 <-- this is a real problem.  its counting the call as "answered"
16:08.11[TK]D-Fenderreally_phukt: If you don't need the BW then don't do it.
16:08.29[TK]D-Fenderreally_phukt: IAX2 has quality/stability issues.
16:09.59russellbit should be good in the later 1.4 releases
16:10.04russellbwe put a lot of work into it
16:10.13x86works great here
16:10.18russellbsee :-p
16:10.46x86I've got 7 asterisk boxes sprinkled across most of my branch offices and HQ, and I do IAX2 trunking between all of them for intra-office calls
16:10.53x86inter-office I mean
16:10.56x86works great
16:11.06x861.4.12.1 is what I'm running
16:11.28russellbyou're lucky then, a lot of stuff has been fixed since then :)
16:11.54russellbwell, 25 fixes to that module ... less than i thought
16:13.32dandreok [TK]D-Fender but I don't why this channel answers .
16:13.32dandreCan this:
16:13.32dandre<PROTECTED>
16:13.32dandrebe related to that?
16:13.58x86russellb: guess so ;)
16:14.06x86russellb: well "trunking" never worked for me
16:14.22x86russellb: I say I'm trunking, but it's not setup that way in iax.conf :)
16:17.38*** join/#asterisk esaym (n=user@72.183.198.134)
16:20.06*** join/#asterisk pfn (n=pfnguyen@hanhuy.com)
16:21.24Qwell~hold
16:21.24jbothold is probably a status flag to tell apt not to automatically upgrade a package.  apt will place packages on hold if they require packages that are not currently installable; you can 'apt-get install pkgname' to explicitly install the package.  To put a package on hold, 'echo pkgname hold | dpkg --set-selections' or use the '=' key on the package in dselect, or 'echo pkgname install | dpkg --set-selections' to remove the hold
16:21.28Qwell<3
16:25.22[TK]D-Fenderdandre: I'd also be looking at that voicemail line funny...
16:25.32[TK]D-Fenderdandre: Comment it out.
16:25.45[TK]D-Fenderdandre: its TRIED going in there.. that might have something to do with it.
16:25.50dandre[TK]D-Fender: that's it
16:26.01dandreI've just discored it now
16:26.23[TK]D-Fenderdandre: Because it did say "stopping sounds", but you never saw a reall attempt to play audio.
16:27.08dandrehow can I know wether a voicemail exists in the dialplan?
16:27.52flushwhat is said in the sound file "vm-login".. i dont understand the first 2 words.. "something mail"
16:28.06Qwellflush: you've already had like 3 people answer you
16:28.08fileComedian Mail.
16:28.09[TK]D-Fenderdandre: You are calling it in your maco.  Perhaps you should put some actual thought into your dialplan coding....
16:28.22flushComedian mail? hrm
16:29.28flushcan i boost a phone volume in zapata.conf file cause when i use hands-free volume is so low its almost ridiculous..
16:29.58[TK]D-Fenderflush: "lookup rxgain" & "txgain"
16:30.10flushcopy
16:35.28UserReg_CLhi... which is velue TOS for xlite ?
16:37.40*** join/#asterisk adjohn (n=adjohn@p7221-ipad87marunouchi.tokyo.ocn.ne.jp)
16:37.53*** join/#asterisk nephfl (n=none@wsip-70-168-186-225.ga.at.cox.net)
16:38.20nephflhello im trying to setup vtwhite for incoming and am having no success
16:39.13flushhey.. i have set voicemail its awesome i removed my bell message box and now saving money
16:39.30flushbut id like to know.. can i put an option or something to say theres a message, like a busy tone when you pick up the phone or something ?
16:39.43jblackNow move to high speed internet and remove local phone service. That'll really save you
16:40.08jblackflush: Some ATAs can do that, like the linksys line.
16:40.18flushim powered by the POTS
16:40.32jblack* can probably do it on FXS ports too. Check indications.conf
16:41.53*** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com)
16:43.24BobLutzrussellb: file: Are you guys around / Would I be able to talk with via PM for a quick second ?
16:43.56fileI don't accept private messages, anything said usually benefits all.
16:44.15UserReg_CLhi... wich is the value for bit TOS ?
16:44.23russellbsame as file
16:44.25nephflmaybe he wants to talk dirty
16:44.28russellband i'm quite busy right now
16:44.58BobLutzrussellb: file: very specific question, I'll ask later
16:45.00BobLutzthanks
16:45.06[TK]D-Fenderflush: Set the mailbox for your device.
16:45.14russellbBobLutz: if it's code related, just join #asterisk-dev and ask there
16:45.20russellbplenty of people around that can answer besides us
16:45.37BobLutzjoins #asterisk-dev
16:47.17*** join/#asterisk enjay5150 (n=chatzill@ip70-190-60-237.ph.ph.cox.net)
16:48.24enjay5150Is it possible to use '#' within the same context "multiple times" for validation (i.e. press # to confirm)
16:48.55enjay5150like in an IVR. Enter your name "press pound to continue" Enter your Address "press pound to continue" etc..
16:50.11[TK]D-Fenderenjay5150: "core show application read"
16:51.10*** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net)
16:51.19enjay5150thanks
16:57.51*** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com)
16:58.28enjay5150so exten => 1,n,Read(whatever,,#,,1) could work..
16:58.43enjay5150thanks D..
17:01.24rkeeneMan, Polycom's webpage sucks!
17:02.27rkeeneAll I want to do is download their crappy software, for the crappy phones which I have purchased
17:02.41*** join/#asterisk bkw_ (n=brian@adsl-71-153-171-225.dsl.tul2ok.sbcglobal.net)
17:02.50enjay5150lol what issues are you having with their phones?
17:03.23rkeeneWell, mine locks up on boot 75%+ of the time
17:03.30rkeeneAnd requires a hard-power-cycle to fix
17:03.41*** join/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com)
17:03.44enjay5150at the IP screen?
17:04.03*** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com)
17:04.10enjay5150what sip code are you currently running?
17:04.27rkeeneSoundPoint IP 550;  It locks up at a couple of different places.  At the point where it tells me my IP/MAC and says "Please wait ..." and at the main screen
17:04.46rkeene2.2.2.0084
17:05.59enjay5150I'll find out if theres an update sec..
17:06.00rkeene(The latest version I could try without creating an account... then I created an account ... eventually, their web page wouldn't let me register my organization because it thought someone already registered for me (which could be, since we are huge military organization, but I still want support).. and now it won't give me access to the firmware)
17:06.52[TK]D-Fenderrkeene: Just ask your reseller.
17:07.04rkeene[TK]D-Fender, Like I know who that is
17:07.11rkeeneI don't control who we buy crap from
17:07.15[TK]D-Fenderrkeene: You don' know where you bought them from?
17:07.23rkeeneNo, the orders get bid out
17:08.10rkeeneIt works like this, I tell someone in procurement what I want, they send it up to DC, who loses the paperwork, we resend, and they bid it out (even if it's on the GSA schedule) and 6 to 12 months later a product arrives
17:08.41rkeeneThat doesn't come to me, of course -- it goes to DC.  They package it up and send it to me.
17:08.52rkeene(sometimes)
17:09.23*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
17:09.38enjay51503.0.0 is out..
17:09.47*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
17:10.29rkeeneYes, but their portal at this point won't let me access it
17:10.42jblackIs there a traditional extension to dial for pickups?
17:11.09enjay5150depends on what you've defined in your features.conf
17:11.34jblackOh, of course. I was about to mis-use the PickUp() app
17:12.43*** join/#asterisk Rico29 (n=Rico@ARennes-358-1-104-253.w86-203.abo.wanadoo.fr)
17:13.01*** join/#asterisk The-Bat (n=The-Bat@203.199.114.33)
17:13.10rkeeneGrr!
17:13.14rkeenehates Polycom's webpage
17:13.40jblackhmmm. actually, I can understand why I didn't look there. I've always thought of features.conf for already established calls.
17:13.59*** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au)
17:20.16*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)
17:20.35RobHOk, time for a stupid question.  I have Asterisk 1.4 compiled and running on ubuntu, kernel 2.6.22-14-generic.  Also compiled are libpri-1.2.7 & zaptel-1.4.9.2.  Asterisk loads up, and it will accept the simple dial of my IAX2 softphone, and reads that it is playing back a sound, but I hear no sound on my phone.  They are on the exact same subnet, no firewall between them.  Any ideas of what I should check first?  (I have not had this
17:20.51RobH(Sorry for the long message, figured it best to throw out as much info as possible.)
17:21.01DefrazHey all, all of the sudden my linksys pap2 and pap2t atas sound bubbly in one direction? I use a polycom or softphone and it is clear as day.
17:21.28Defrazbut the 3 ATAs sound bubbly in one direction. I call from a land line and I hear it bad, but they hear me crystal clear.
17:21.55Defrazcould be a codec issue
17:22.07Defrazthat happens for me when my codec isn't right.
17:22.28RobHInteresting, ok, I am new at this, how would I troubleshoot the codecs?
17:22.50Defrazmake sure all your devices have the same codec set.
17:23.05RobHShouldnt Asterisk translate that for me automatically though?
17:23.46RobHHmm, I just confirmed I can call hardphone to softphone, so its working somewhat.
17:24.02Defrazdepends
17:24.13Defrazdo you have canreinvite=no?
17:24.25RobHcrap, that would do it huh?
17:24.38RobHi dont stop reinvites
17:24.43Defrazyep
17:24.48RobHbut one is on iax2 and one is on sip, so wouldnt it have to stay in asterisk?
17:24.48Defrazcould cause that.
17:24.57Rico29hi
17:25.29Rico29does anyone knows where I can find a softphone which supports h263 or h264 video ? (FOR LINUX)
17:25.31[TK]D-FenderRobH: Check your sound card, and fix your libpri while you're at it.
17:25.44[TK]D-FenderRico29:
17:25.46[TK]D-Fender~ekiga
17:25.46jbot[~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org
17:26.01Rico29thanks
17:26.05eric2anyone have success getting the intercom functionality to work with snom phones?
17:26.07RobHcheck sound card on the server?  It has no sound card.  libpri compiled ok.
17:26.15Rico29i thought that ekiga didn't support it
17:26.24eric2linksys phones.. no problem, got it working.. but snom is a beyatch!
17:27.02mpwizarderic2: It worked for me,  out of the box with Snom M3.
17:27.50eric2so from one snom phone you can dial another snom phone and it automatically conferences? withouth having to pick up the phone or press any buttons on the receiving end?
17:28.11eric2I have snom 300's
17:33.15DefrazCan't find an Ekiga binary for windows.
17:34.10Juggiehttp://snapshots.voxgratia.org/win32.php
17:35.43RobHOk, my polycom is forced to ulaw, and my zoiper to gsm.  They can call one another, reinvite is off, and they are staying bridged.  Asterisk translates from ulaw to gsm just fine.
17:35.52*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6 , 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies.
17:35.59RobHbut when I call a simple playback function, no audio is played.
17:36.23[TK]D-FenderRobH: thats often a zaptel issue.  unload ztdummy.
17:36.29Qwellrussellb: did you intentionally leave out 1.4.8.1?
17:36.33russellbno
17:36.34Qwell1.4.18.1 rather
17:36.39*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6 , 1.4.18.1, 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies.
17:37.47RobH[TK]D-Fender: That worked.  So continuing my line of novice questions.  How exactly can I troubleshoot the ztdummy issue.  I require that it work so I can use meetme for my company.
17:38.00RobHand thank you for the answer =]
17:38.12russellbQwell: and i'll have to update the web site after i get some food ...
17:39.53*** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6, 1.4.18.1, 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies.
17:39.56Qwellhmm
17:40.13*** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu)
17:40.16[TK]D-FenderRobH: I don't know the fine points about this personally.  I jsut know its usually the culprit.  Manxpower has most often spoken about this in here to my memory.
17:40.23b11d`whats the dCAP all about? Is it worth getting>?
17:40.41RobH[TK]D-Fender: Thank you, your name came up in bootcamp last week =]
17:40.57[TK]D-FenderRobH: LOL... can't be good news :p
17:41.04RobHI noted that a few months ago you helped me, Jared made a point of saying that you are one of the regulars who help folks
17:41.31[TK]D-FenderRobH: Yeah, I'm something of a constant around here (others would likely add to that statement...)
17:41.47RobHYes well, I am now an official lurker.
17:42.07RobHwe pushed our asterisk server live, but its 1.2 and my dialplan looks like vomit =P
17:43.03b11d`anyone? dCAP worth it?
17:43.05*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
17:43.26RobHb11d`: It puts your name on Digium's website as an asterisk person to contact.
17:43.47b11d`im more about the learning and the content than the name on a website..
17:43.50RobHSo if there are no others in your area, or a limited amount, it may generate you some business.
17:43.56b11d`hmm..  thats cool
17:44.01b11d`my employer would pay for me to take it..
17:44.03RobHbootcamp was good for the learning.
17:44.05b11d`so I was considering it
17:44.08RobHI just did it last week.
17:44.10b11d`cool
17:44.13b11d`where?
17:44.21RobHI went to the main Huntsville AL office
17:44.24b11d`I was thining Huntsville.. but would LOVE the Netherlands :)
17:44.29b11d`yeah Las Vegas sucks
17:44.29b11d`:)
17:44.30RobHmet all the Digium folks, it was very neat.
17:44.56RobHthe Vegas one is just in a training facility, at Huntsville it is in the main office, which is cool.
17:45.08RobHplus Huntsville has less distractions from study, heh.
17:45.13b11d`yeah.. i just hate vegas.. Huntsville sounded like the best place to go
17:45.15QwellRobH: when did you do it?
17:45.21Qwelloh, last week.  nevermind
17:45.21RobHJust last week =]
17:45.36Qwellahh, yeah, I heard about you ;)
17:45.39Rico29do you work with OPAL lib ?
17:45.39RobHJared was an excellent trainer too.  Everyone in our class took the dCAP
17:45.47QwellRobH: yes, Jared is awesome
17:45.48b11d`how many people were there?
17:45.52RobH10 folks total
17:45.53*** part/#asterisk enjay5150 (n=chatzill@ip70-190-60-237.ph.ph.cox.net)
17:45.55b11d`good
17:45.58RobHplus jared as instructor
17:46.27b11d`im really in it for the Orange Ice Digium Pen
17:46.32RobHNow, my company paid for it, so my viewpoint is kind of skewed, but it was well worth my time and my company's money
17:46.42Rico29do anyone work with OPAL libs here ?
17:47.03QwellRobH: so, how extensively do you guys use asterisk there?
17:47.30b11d`I am a FreeBSD Man.. not linux.. will I have much trouble with the dCAP?
17:47.44RobHOur main office phone system is using 1.2 with an iax2 connection to our voip provider
17:48.00Qwellahh..  gonna upgrade?
17:48.03RobHYes.
17:48.10Qwellhence the training, I assume
17:48.19RobHI am trying to make my test system, which is an exact copy of the main one, into 1.4 deployment
17:48.43RobHtraining was more for the fact I struggled solo to get 1.2 up and running, and it works, but its basic and doesnt do all we want it to do
17:49.11RobHgot the dCAP because it was only 300 bucks more and I was already there =]
17:49.18RobHdunno if I passed yet, heh.
17:49.33Qwellreally?  hmm..I thought they graded it while the people were still here
17:49.42Qwellmaybe not..  *shrug*
17:49.45filenot all of it.
17:49.46RobHWe got the practical graded
17:49.48Qwellahh
17:49.50RobHbut not the written
17:49.56Qwellmakes sense
17:49.59b11d`aye
17:50.05RobHI passed the practical easily, but the written has some obscure questions.
17:50.39RobHI think I passed, but I am often overconfident about my test taking abilities.
17:50.54fileyarrrr it's harsh
17:51.06RobHnow if i could figure out why ztdummy is being stupid...
17:51.11*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
17:51.42RobHI am almost to the point of cheating and just purchasing a hardware card to put in the server at the office for the timing...
17:52.03QwellRobH: nothing wrong with that
17:52.11alrsRobH: you can pull timing off a POS x100p card
17:52.36RobHThe problem is eventually we will have 4 servers deployed for this, I rather not have to put a card in all of them.  I rather get ztdummy working =]
17:52.42alrsjust don't try to load ztdummy and a real zaptel module at the same time?
17:53.11*** join/#asterisk Cresl1n (n=matt@216.207.245.1)
17:53.12*** mode/#asterisk [+o Cresl1n] by ChanServ
17:54.31DefrazI still see bridging even though I have canreivite=no
17:54.35DefrazDon't' know what is up.
17:54.58RobHalrs: when zaptel compiled, it saw no hardware and stated it was loading only the ztdummy.
17:55.21RobHsame thing when it starts the zaptel service
17:55.28RobHsees no hardware, loads ztdummy
17:55.37RobHQwell: You said you heard about me?  eh?
17:58.05RobHits ztdummy being stupid, i load the service and asterisk generated audio ceases to function
17:59.55*** join/#asterisk pud (n=jkatka@sea01-v502-nat.marchex.com)
18:09.01*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
18:10.20RobHcaves and installs the tdm411B
18:12.06*** join/#asterisk svenna_ (n=svenna@p548D357E.dip0.t-ipconnect.de)
18:16.00*** join/#asterisk implicit (n=implicit@200.12.227.181)
18:17.29*** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
18:18.17Rico29~xlite
18:18.18jbot[~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/
18:19.48*** join/#asterisk Bonix-BR (i=Bonix-BR@217-lo1.rt2.isimples.com.br)
18:19.50*** join/#asterisk kfb (n=kfb@c-76-116-250-235.hsd1.pa.comcast.net)
18:20.45really_phuktxlite is cool, but do u know how to add another account?
18:21.38_ShrikE~mini-mall
18:21.38jbotIt's just like, it's just like, a russell.. b
18:21.43_ShrikEha
18:22.38[TK]D-Fenderreally_phukt: You don't.  Its limited in that way
18:23.01*** join/#asterisk hohum (n=dcorbe@68.26.108.244)
18:23.26really_phuktI guess I gotta upgrade to eyebeam or bria
18:23.27*** join/#asterisk AJayMN (n=Me@75-134-29-194.dhcp.mdsn.wi.charter.com)
18:24.25RobHtry zoiper, lets you have 2 for free.
18:24.38RobHbut still limited.
18:25.11*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135)
18:25.19really_phuktI tried it some time ago, but it doesn't look as cool :)
18:25.39RobHTrue, x-lite is smooth looking.
18:25.41[TK]D-FenderCool... yeah, because thats what really important.... </sarcasm>
18:25.46RobHheh
18:25.48*** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller)
18:25.58RobHbut iax2 connection via softphone > sip phone that looks cool.
18:26.06RobHatleast for me.
18:26.18lmadsengrrrr... so when I do a transfer from the destination extension, I get a <ZOMBIE> channel (something useful to parse on), but when the source does the transfer, I don't get that <ZOMBIE> channel, thus I can't tell if it was a transfer or just a regular call in my dialplan. Also wonder why my TRANSFER_CONTEXT isn't working too (I'm just thinking out loud)
18:27.36really_phuktwhat do u guys think about ekiga?
18:27.53lmadsenI think you should try it and form your own opinion
18:28.20*** join/#asterisk stoffell (n=stoffell@d51A4DE51.access.telenet.be)
18:29.06really_phuktthank you lmadsen... not
18:29.30lmadsenthat joke is so 1992
18:29.35*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
18:30.34really_phuktI don't remember putting any smileys at the end of my statement
18:30.36*** join/#asterisk esaym (n=user@72.183.198.134)
18:31.02lmadsentouche
18:31.40lmadsenif ekiga didn't work for someone, then there wouldn't be active development on it -- the only way to get a real opinion of something is to try it in your own environment and see if it is useful. Taking a poll is hardly useful.
18:32.01lmadsenmight as well ask what distro is the best
18:32.38[TK]D-Fenderreally_phukt: No smiley?  Good... that means you should be prepared for further berating and disappointment.
18:34.04*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
18:34.47really_phuktI just wanted ask a question, as I figured someone here has experience with it. What's wrong with that?
18:35.42[TK]D-Fenderreally_phukt: Well, you did ask.  We answered and felt cheated.  Just letting you know its all part of the experience.
18:36.03*** join/#asterisk macli (n=macli@nmc.brc.ubc.ca)
18:36.34lirakisis away (leaving..."the internets" are safe ... for now)
18:36.38*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
18:36.50[TK]D-Fenderreally you*
18:36.51really_phukt[TK]D-Fender: ok, fair enough
18:36.56*** join/#asterisk ccvp (n=ccvp@66.0.46.210)
18:37.11[TK]D-Fenderreally_phukt: Excellent.
18:37.35jameswf-homeweeee.
18:38.18[TK]D-Fender~whee
18:38.18jbot[~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee
18:38.58AJayMNI need to rewrite the callerid ID from the person calling into asterisk..  I need to add a 1+callerid
18:40.01ccvpheh
18:40.17ccvp"really_phukt", is your name an irc indirectly reference to Really Fucked?
18:40.18ccvp:)
18:40.20cpmyeah, [TK]D-Fender, what the heck is wrong with you?
18:40.22ccvpbrb, coffee
18:41.18lmadsenAJayMN: Set(CALLERID(num)=1${CALLERID(num)})
18:41.31ccvpd-fender, i think i saw some guild last night in COD4 multiplayer all with {TK} clan tags
18:41.32really_phuktccvp, not it refers to you
18:41.48ccvpi instantly thought of d-fender, and Tha killaZ :)
18:42.03shido6can you use playback or background to play a blob from realtime?
18:42.12lmadsenshido6: sorry, no such luck
18:42.32lmadsenshido6: would be very cool though, Qwell and Corydon76-dig starting something for that, but didn't get a chance to finish it
18:42.39lmadsenafaik
18:42.42Qwellhuh?
18:42.47Qwelloh
18:42.55lmadsenthought you guys has a branch for playing stuff from ODBC blobs
18:43.01Qwellast_storage
18:43.02lmadsenor at least a framework started for it
18:43.05lmadsenya, that was the one
18:43.23RobHIntersting, I record my voicemail name, it says it saves itm but gives an error
18:43.32RobH[Mar 18 14:41:27] WARNING[5030]: app_directory.c:197 retrieve_file: Failed to obtain database object for 'asterisk'!
18:43.52RobHack ,thats the dir call
18:43.52RobH[Mar 18 14:41:06] WARNING[5025]: app_voicemail.c:1408 store_file: Failed to obtain database object for 'asterisk'!
18:44.39ccvpRobH
18:44.40ccvphttp://www.google.com/search?hl=en&q=+app_voicemail.c%3A1408+store_file&btnG=Search
18:45.06Rico29is it easy to set up an enum server ?
18:45.22RobHccvp: saying dont paste in channel?  My bad, sorry about that.
18:46.33ccvphuh?
18:46.38ccvpno heh
18:46.42RobHthe link goes to a blank entry page
18:46.43ccvpits just my first result from a google
18:46.46ccvpyes, just noticed
18:47.13*** join/#asterisk bdheeman (n=bsd@122.161.65.75)
18:47.53Rico29nobody knows about seting up an ENUM server ?
18:48.23ccvprico, we are in the darkness, only coming out at night , asteriskers
18:48.27RobHGive me a bit of credit, I tried googling a bit =]
18:48.34Rico29huh ?
18:48.35ccvpcan I lead you to the light, so you come out, and socialize in the daylight, ie: call manager express ? :)
18:49.05RobHI find lots of folks with the issue, but no answers, it looks like asterisk is trying to reach a non-existant relational database, rather than use the AstDB
18:49.14Rico29sorry but I don't understand everything, particullary the " ie: call manager express "
18:49.17BCS-SatoriIf i am using the exten = s,n,GotoIf($[${CALLERID(num)} = 4105551234]?reject:allow) feature to reject the phone number 4105551234, how would i go about adding a second, or even a thrid number to reject
18:49.22ccvpmen like the darkness, because their asterisk ways are kept secret, for if they come to call manager express, their asterisk sins will be exposed during the day
18:49.36*** join/#asterisk ACiDV (n=joel@122-205-229.dr.cgocable.ca)
18:49.54Rico29hu ?
18:49.59*** join/#asterisk razu__ (n=razu@195.222.7.33)
18:50.25Rico29in basic english ?
18:50.29Rico29:)
18:50.31*** part/#asterisk bdheeman (n=bsd@122.161.65.75)
18:50.36ACiDVWhat can explain, on CDR log, that a call has a duration/billsec in NEGative ? (ex. -21642 seconds)
18:50.43ACiDVdate/time change ?
18:50.43ccvpn/m rico, your french
18:50.53ccvpyou dont understand , is joking, i cannot help with your enum server issue.
18:50.59Rico29ok
18:50.59Rico29:)p
18:51.12*** join/#asterisk Greek-Boy (n=email@41.221.58.4)
18:51.14Rico29what means "m/n" ?
18:51.14*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
18:51.19ccvp"nevermind"
18:51.22Rico29k
18:51.24ccvpi shall teach you the ways of american
18:51.24Rico29ok
18:51.27ccvpinternet addiction slang
18:51.53Rico29hard for a frenchie to understang everything
18:51.57Rico29:p
18:52.15ACiDVah les francais :P
18:52.17ccvp<PROTECTED>
18:53.11Rico29ACiDV > mé euh
18:53.23Rico29huhu
18:53.38ACiDVishh ca la ete long avant que le la catch la 'Parlay voo fron say' ...
18:53.47ACiDVsorry ;)
18:53.48Rico29lol
18:54.02ccvpacid
18:54.04ccvpwtf did you say
18:54.07ccvpim being sarcastic
18:54.11ccvpwith my american spelling heh
18:54.13ccvp?
18:54.32ACiDV:D ccvp, just took me time to understand that ' Parlay voo fron say ' mean ' Parlez vous francais ' :P
18:54.43*** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com)
18:55.51RobHwell hell.  I was getting the error because I had ODBC storage set for voicemail.
18:56.01RobHNot sure how to remove that setting, so I just recompiled.  easy enough.
18:57.26*** join/#asterisk hmm-home (n=hmmhesay@24-119-176-74.cpe.cableone.net)
18:59.47[TK]D-Fenderccvp: Garde ta guele sinon tu l'perdrait  bien-tot ;)
19:00.42Rico29i'm french and I didn't understant a damn thing
19:00.45Rico29:D
19:00.50Rico29sorry [TK]D-Fender
19:00.58rkeeneGrr.
19:01.02rkeenekills Polycom
19:01.45rkeeneTheir webpage sucks.
19:02.30ACiDVRico29, j'ai mon traducteur francais -> anglais et francais -> quebecois a mes cote, tjrs utile
19:02.45Rico29:p
19:02.52Rico29ok huhu
19:02.56ccvpD-Fender, le garde-boue, quelle heure sont moi et vous allant avoir notre temps d'homme privé ensemble ce soir en arrière à votre appartement ? :)
19:03.14ACiDVdoh =)
19:04.26[TK]D-Fenderccvp: That is a positively HORRID translation...
19:05.28[TK]D-FenderACiDV: Je parles bilingue pour me sauver du temps ostie!
19:05.29Rico29ccvp > hahaha
19:06.08ACiDV[TK]D-Fender ... c'est pas Andrew ca ?
19:06.36[TK]D-FenderACiDV: Salut toi-la :)
19:07.00ACiDVJust to be sure =)
19:07.07[TK]D-FenderACiDV: Je t'ai pas reconnu de meme...  tiens-donc une seule nom!
19:07.53[TK]D-FenderACiDV: Just installed that framework you've been using at home with the basic kit.... cool stuff...
19:08.18[TK]D-FenderACiDV: And getting close to that re-install.  I have a who server rack to rebuild.
19:08.46ACiDVok, np =)
19:08.52ACiDVmust leave, ttyl here or on msn ;P
19:08.56[TK]D-Fenderwhole*
19:09.02[TK]D-FenderACiDV: later.
19:09.18*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
19:11.28*** join/#asterisk Hemos\ (n=cyberspa@80.104.208.187)
19:12.23ccvpwtf
19:12.26ccvptheonion.com is insane
19:12.28ccvp(BREAKING NEWS)(AP/REUTERS) - President Of The United States Barack Obama has been assassinated while in Manhattan, NY.  Vice President Hillary Clinton assumes the Oath of Office as President of the United States.  Details Soon.
19:15.14bkruseccvp: lol
19:15.23*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135)
19:23.26*** part/#asterisk really_phukt (n=chatzill@static-72-77-217-74.tampfl.dsl-w.verizon.net)
19:29.25*** join/#asterisk datachomper (n=russ@75.146.194.59)
19:30.11datachomperHow can I tell what DTMF mode, a call being sent to my asterisk box is in?
19:31.14datachompersipinfo, rfc2833, or inband?
19:35.09lirakisis away (leaving..."the internets" are safe ... for now)
19:39.39*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
19:48.54Greek-Boywill wav sounds sound better on ulaw and alaw calls?
19:49.38Greek-Boyif all sound file types are installed will asterisk playback the format most suitable for the channel?
19:51.41Kattyanyone know sendmail enough to give me a hand?
19:51.54Kattyi'm doing voicemail to email, and it isn't happy )=
19:54.15*** join/#asterisk PepOSX (n=angeldav@190.72.154.247)
19:54.26x86Katty: what ever happened with that PRI?
19:55.01Kattyx86: it works!
19:55.12Kattyx86: tho, FOP doesn't like DID numbers much.
19:58.19datachomperHola Katty
20:02.57*** join/#asterisk jamessan (n=jamessan@debian/developer/jamessan)
20:03.51Kattydatachomper: hola! como estas? (=
20:03.52*** join/#asterisk shtoom (n=godson@123.176.41.46)
20:04.05jamessanis it possible to get * to add a Route field to the Via header without having seen a Record-Route header first?
20:04.22x86Katty: what was the problem with the PRI?
20:04.48Kattyx86: i had a line in zaptel wrong...
20:04.57Kattyx86: sangoma fixed me right up tho (=
20:05.11Qwellthat'll teach ya
20:05.13JunK-YKatty: which line?
20:05.22Kattysec, lemme me find my blog post
20:05.51RobHGreek-Boy: someone answer you?
20:06.10x86Katty: yeah they are great :)
20:06.19RobHGreek-Boy: Asterisk will play back whatever soundfile it has that requires the least cpu power to transcode (so if the caller is on ulaw, it will choose the ulaw sound file, as it takes no cpu)
20:06.29QwellGreek-Boy: no phones really support wav, so it'll have to be transcoded - and in some cases will sound "worse"
20:07.08RobHYea, if you have a ton of HDD space to burn, just install all the possible codecs you will allow to connect, and remember to convert your custom recordings into the various formats.
20:07.33Kattyx86: JunK-Y: http://angela.sleekgeek.org/2008/03/14/sangoma-pri-card-setup-a101/ <- see note from sangoma.
20:08.35JunK-YKatty: so it was just a bug with ur config, right?
20:08.40KattyJunK-Y: ya
20:09.38JayTee52Katty, I just read the blog
20:09.42x86Katty: ah, see, I told you I've not done much with PRI's...
20:11.16JayTee52Katty, I'm wondering if I can take a context that has all of our extensions in it and add it to my [from-pstn] context as an include and then have an additional line that would be executed if none of the extensions in the included context match
20:14.10Greek-Boythanks RobH and Qwell
20:14.17RobHhope it helps =]
20:15.14Greek-BoyI'm sure it will, I'm just going to go ahead and install all the sound formats
20:15.36RobHif you have the space to kill, it cannot hurt
20:16.06Greek-BoyI noticed on the asterisk web site there is a format called sln16 available for download
20:16.31RobHI dunno that off the top of my head, I reference this:  http://www.voip-info.org/wiki-Asterisk+codecs
20:17.05RobHI do not force many codecs on my system though, I just let folks use gsm, ulaw, and alaw in that order.
20:17.42RobHit all goes through a single voip provider which lets me trunk iax connections, so I am good as far as bandwidth.
20:17.57JayTee52does Asterisk handle which codec to use automatically or does that have to be configured in the musiconhold.conf or another config file?
20:18.26RobHits automatic, in regards to cpu utilization, not bandwidth.
20:18.38RobH(if I am saying something incorrect, someone jump in and tell me.)
20:19.16RobHdo a core show translation in the CLI, it shows you cpu loads to transcode codecs (in miliseconds)  its just kinda nea.t
20:19.18RobHneat even
20:19.51RobHbut then again, I am a giant freakin nerd who stares at a terminal for 10 hours a day ;_;
20:22.21*** join/#asterisk harlequin516 (n=sham@stewart.styk.net)
20:22.41harlequin516how can i find out fromthe asterisk console what is the soundfile path?
20:25.09*** join/#asterisk mfedyk (n=mfedyk@adsl-71-134-153-204.dsl.irvnca.pacbell.net)
20:25.44harlequin516Hello, anyone here?
20:25.50harlequin516helo?
20:25.58RobHfolks are here, but they are lurking and working, gotta be patient.
20:26.58codefreezeharlequin516: There's 265 lurkers in here. The walls are literally plastered with eyes!
20:27.03harlequin516Okay, sorry.
20:27.13mfedykanyone know how to use two or more subexpressions to extract data into variables?
20:27.20mfedykin the dialplan
20:27.34mfedykI have been able to get one expression and extract it
20:27.42mfedykbut two or more just return the first
20:28.19codefreezeharlequin516: I can't think of a cli command to show that; you may have to consult the source. Usually it's /var/lib/asterisk/sounds
20:28.41mfedyk;extract dialed number
20:28.42mfedykexten => s,n,Set(dialed_num=$[ "${MACRO_EXTEN}" =~ "(.*)\\*" ])
20:28.42mfedyk;extract user specified callerid
20:28.42mfedykexten => s,n,Set(callerid_num_custom=$[ "${MACRO_EXTEN}" =~ "\\*(.*)" ])
20:28.55mfedykI currently use one line per part I want to extract
20:29.11mfedykthat's ok for two, but if I wanted to extract more, it'd be a lotta lines
20:29.38*** join/#asterisk jmesquita (n=jmesquit@201.7.117.114)
20:29.58*** join/#asterisk ManxPower (n=manxpowe@107.sub-70-221-153.myvzw.com)
20:30.29harlequin516codefreeze: thanks
20:31.40*** join/#asterisk implicit_ (n=implicit@200.12.227.181)
20:33.22Greek-Boyyeah Rob you're right, gotta be patient :)
20:33.34Greek-BoyRobH thanks for your time and help
20:33.48RobHanytime
20:34.11RobHwhen i answer questions i just end up understanding things better, like anyone else.
20:39.41*** join/#asterisk adrin_ (n=adrin@chello083144070043.chello.pl)
20:39.57adrin_hello
20:40.17RobHOk, a question for me, whats the asterisk application to call to add to a number (may jut be expression)  I have the pattern match if they dial a 7 digit number, it means its local to the office area code, so I want asterisk to slap the area code in front of the EXTEN before passing it to the outbound.
20:40.57Greek-Boywhat is the app ivrdemo?
20:43.52adrin_Hello, i have a question: is there a way to allow a calling user to select, after connecting, an extension (like 9x) and after a period of inactivity to redirect him (connect) to a default extension lets say 90?
20:44.02*** part/#asterisk km- (n=pgrace@aeneas.fierymoon.com)
20:44.08russellbGreek-Boy: it's an example for an internal IVR API ...
20:44.17russellbwhich never has been used outside of that demo, heh
20:46.09_ShrikEHey russelb, not to nag, but were you able to backport 1.6 hint?  I was excited to hear you were doing it :)
20:46.23russellbno, i haven't done it, sorry
20:46.29russellbgot very busy making releases
20:46.41_ShrikEthats OK.
20:55.24*** join/#asterisk akira2014 (n=chatzill@173.Red-88-17-52.dynamicIP.rima-tde.net)
20:55.30akira2014hello
20:56.08akira2014can some one tell me how to dial a iax extension in the dialplan
20:56.10akira2014?
20:56.15akira2014thk's
20:56.40*** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org)
20:56.56Maliutadial(IAX/etn)
20:57.06MaliutaRTFM FFS
20:57.41*** join/#asterisk ManxPower (n=manxpowe@74.sub-70-221-32.myvzw.com)
20:58.24*** join/#asterisk kamaji (n=kamaji@resnet-186224.resnet.bris.ac.uk)
21:00.16*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
21:01.23akira2014Maliuta: i don't need to put IAX2
21:01.25akira2014?
21:01.59MaliutaRead The F*@!ing Manual
21:02.03*** join/#asterisk rpyne (n=richard@69.77.169.14)
21:02.15*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
21:03.04rpyneDoes nayone know how to startup the tftp server on Asterisk 1.4. I'm running CentOS
21:03.21Kattywibbles
21:03.24russellbMaliuta: an RTFM attitude is not welcome here.
21:03.37russellbrpyne: asterisk has no tftp server ...
21:03.43russellbthat's a completely different application
21:03.54alrsisn't there a tftp server going in to 1.6?
21:04.01russellbno
21:04.09Qwellalrs: there's a web server
21:04.13Qwell(in 1.4 too though)
21:04.17russellba very minimal HTTP server, yes
21:05.50kamajiCan I run asterisk and a SIP cilent on the same computer or does SIP have to use a certain port
21:05.53rpyneHow do I get a TFTP server up and running on my CentOS so that I can manage some 400 aastra sip phones
21:06.00[TK]D-FenderQwell, What is the advantage of having an HTTP server directly as part of * again?
21:06.26[TK]D-Fenderkamaji, Set your client to use another port so it doesn't interfere with * binding to the primery.
21:06.38kamaji[TK]D-Fender: mkay, thanks
21:06.55[TK]D-Fenderrpyne, "man tftpd.conf"
21:07.42rpyneThanks much [TK]D-Fender
21:07.56Kattyhow do i use asterisk?????????!!11
21:08.02Kattyoh, wait
21:08.02rpyneexit
21:08.05Qwell~nowwhat
21:08.06jbotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E
21:08.07QwellKatty: ^^
21:08.09Kattyhow to use asteriskk plzzzzzz
21:08.11Kattyetc.
21:08.13*** part/#asterisk rpyne (n=richard@69.77.169.14)
21:08.26Kattyexim4 doesn't like me.
21:08.28filetickles Katty
21:08.33Kattyi shall beat it with sticks until its attitude improves.
21:08.43Kattyhugs file
21:10.37*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135)
21:10.45*** join/#asterisk bronson (n=bronson@adsl-68-122-117-135.dsl.pltn13.pacbell.net)
21:12.00lirakis_awayis away (leaving..."the internets" are safe ... for now)
21:12.19Kattyfile: i guess exim4 does love me afterall
21:12.27Kattyfile: i'm just not loving it back with my firewall :/
21:12.52file:(
21:13.06*** join/#asterisk WhipsMcGee (n=barney@c-71-199-49-168.hsd1.co.comcast.net)
21:14.09WhipsMcGeecan someone walk me through get my a sip device working.  my server is behind a firewall and so is the sip device.  I've had it working before with opening a few ports and something with externip but I'm setting up a new server and can't remember what I did.
21:14.53ManxPowerWhipsMcGee: externip/externhost and localnet
21:15.40kamajidoes anyone know how to change Ekiga's default SIP port by any chance? people in #ekiga are asleep
21:16.42*** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320)
21:17.06Yourname`Hello. I was wondering if some good soul would be kind enough to help me figure out why Asterisk isn't using all the cores..  or if it is and htop is fux0red (which looks like it may be, thanks to mvanbaak) how can i find out in top?
21:17.13putnopvutkamaji: I've done that before, and you don't actually change it in ekiga itself, oddly enough. I used gconf-editor to change it the time that I used it.
21:17.13Yourname`nods to russellb
21:17.17WhipsMcGeelocalnet should be 192.168.1.0/255.255.255.0 right?
21:17.28putnopvutkamaji: unfortunately, it's been a while so I don't know the exact steps involved.
21:17.52WhipsMcGeeif my asterisk server is on 192.168.1.200
21:18.04kamajiputnopvut: okay, that's bizarre ^^ thanks
21:18.50jameswf-homelmao I just copied the asterisk "critical update" email to the trix forums... now sit back and wait for an outcry of people demanding they update....
21:19.09*** join/#asterisk toddejohnson (n=toddejoh@63-252-82-2.ip.mcleodusa.net)
21:19.46[TK]D-FenderWhipsMcGee, Yes.  And go read this :
21:19.47[TK]D-Fender~sipnat
21:19.48jbot[~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
21:19.49putnopvutkamaji: inside gconf-editor, apps->ekiga->protocols->sip->listen_port
21:20.25adrin_Hello, i have a question: is there a way to allow a calling user to select, after connecting, an extension (like 9x) and after a period of inactivity to redirect him (connect) to a default extension lets say 90?
21:21.00kamajiputnopvut: i haven't got gnome installed so I think i'll just switch to kiax
21:21.07kamajiputnopvut: thanks anyway though :>
21:21.45putnopvutkamaji: no prob. it took me forever to figure it out when I was using ekiga last year.
21:21.57Greek-Boywhy is it that most of the formats core sound files in http://downloads.digium.com/pub/telephony/sounds/ have an abnormally small file size?
21:22.19QwellGreek-Boy: because they are small
21:22.21*** join/#asterisk jicksta_ (n=jicksta@dsl093-128-144.sfo4.dsl.speakeasy.net)
21:22.27putnopvutThat was easy.
21:22.34Yourname`So easy.
21:22.37Qwelloh, those
21:22.48Qwell55 bytes for the tar is abnormally small :D
21:22.55filesymlinks?
21:23.13Qwellyeah...but some aren't
21:23.45*** join/#asterisk Synoptic (i=Synoptic@modemcable105.136-203-24.mc.videotron.ca)
21:24.04Synoptichi all
21:24.11*** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net)
21:24.37putnopvutadrin_: the WaitExten application sounds like what you want.
21:24.52Synoptici'mhaving trouble compiling the latest zaptel source. I'm using a base system running gentoo with kernel 2.6.24. I have configured my kernel myself, and maybe i forgot to include some option in it. can someone help me debug this please ?
21:24.56putnopvutadrin_: it will wait for the user to type an extension, and once it has been input, it will go to that extension.
21:27.02adrin_ok thanks
21:28.15adrin_i will see it
21:28.26adrin_:beer:
21:28.26*** part/#asterisk jamessan (n=jamessan@debian/developer/jamessan)
21:29.06SynopticHmm.. what is the kernel config to allow module loading regardless of kernel version ?
21:29.31*** join/#asterisk Cazper (n=cazper@85.196.120.126)
21:31.27SynopticI think I found it
21:31.31Synopticrecompiling
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21:34.27Greek-BoyQwell so whats up with those sounds?
21:35.09tzafrirSynoptic, what errors do you get?
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21:36.48tzafrirSynoptic, hmm.. why would you like to override module versions?
21:37.30tzafrirmodule versions can save you from nice panics when external modules are involved
21:39.51Synoptictzafrir : Well, actually, i did not enable the options.. i was speaking out-loud.
21:40.08Synoptichere is the error when make module_install : wctc4xxp.ko needs unknown symbol release_firmware
21:40.14Synopticthis is one of the many error..
21:40.16WhipsMcGeedo I just do amportal restart after changing externip or do I need to restart other services
21:40.21Synopticwhat can cause an unknown symbol ?
21:41.25tzafrirWhat version of Zaptel do you use? 1.4.9.2?
21:41.32Synopticya
21:41.33Synopticbut
21:41.39SynopticI havent enabled osme kerbnel option
21:41.43Synopticlet me reboot hte new kernel
21:41.48Synopticand try to recompile zaptel
21:41.51*** join/#asterisk implicit (n=implicit@200.12.227.181)
21:42.13tzafrirhmmm... is the detection of firmware loading broken there? hmmm....
21:42.21tzafrirdo you have any Zaptel hardware?
21:42.42Greek-BoyI installed unixodb and myodbc packages on debian but menuselect still doesnt allow cdr obc and func_odbc?
21:43.00tzafrirSynoptic, to build vs. a specific kernel version or tree, use KVERS (and maybe also KSRC)
21:43.03putnopvutGreek-Boy: make distclean && ./configure
21:43.14*** join/#asterisk RoyK (n=roy@ip-77-54-149-91.dialup.ice.no)
21:43.39Synoptictzafrir: I have a TDM400P
21:44.04tzafrirso you don't actually need firmware. We can afford faking
21:44.38*** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp)
21:44.53Synopticok
21:46.18tzafrirSynoptic, what is the value of CONFIG_FW_LOADER in your .config ?
21:46.18Synoptictzafrir : WARNING: "crc_ccitt_table" [/root/download/zaptel-1.4.9.2/kernel/zaptel.ko] undefined!
21:46.51Synopticlet me check
21:47.16SynopticNOT SET
21:47.20tzafrirhttp://zaptel.tzafrir.org.il/#_kernel_configuration
21:47.24Greek-Boyputnopvut: unfortunately, make distclean did not do the trick :(
21:48.53Synoptictzafrir : readin your site, thanks
21:49.31tzafrirSynoptic, it's not "my site". It's the README file in the zaptel source tree, formatted with asciidoc
21:49.47tzafrircredit should go to the authors of asciidoc
21:50.17putnopvutGreek-Boy: did you also rerun configure?
21:50.31tzafrirSynoptic, try 'make README.html'
21:51.05Synoptictzafrir : ok. btw, what is the name of the CONFIG_FW_LOADER in menuconfig ?
21:51.34Synoptici know i dont need it since I'm not using firmware hardware
21:52.38Greek-Boyputnopvut: yes I did
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21:53.44putnopvutGreek-Boy: hrmmm...what does menuselect say the dependencies for those modules are?
21:53.52putnopvutGreek-Boy: oh oh!!
21:53.56putnopvutunixodbc-dev!
21:54.10putnopvutNot just unixodbc.
21:54.18lmadsenindeed
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21:54.54Greek-Boybrb
21:55.50WhipsMcGeeOK, so I've read a few documents that all say I need to setup externip and localnet, forward the ports and that should be it.
21:56.15Yourname`A call is made by Asterisk, and when connected, it's sent via sip to another Asterisk install where there's a queue of agents. When agents rcv the call, the agent cannot hear the other side in about 10-12 calls in an hour out of the 60-70 calls they get in an hour. What could be the issue here?
21:56.36Yourname`I know NAT isn't. All agents have no firewalls on their XP PCs, and they are on direct IPs.
21:57.30Yourname`My guess, enroute to the queue box, the caller hangs up, but Asterisk still carrys on the call to the agent.
22:01.31ManxPowerWhipsMcGee: Great, so it's working?
22:01.44WhipsMcGeeno
22:02.02ManxPowerwhat ports did you forward?
22:02.22WhipsMcGee5060 and 10000-13000 and I set /etc/asterisk/rtp.conf to be the same and it's also the same on the phone.
22:02.34ManxPoweryou didn't do something silly like put a hostname in for the value of externip, did you?
22:02.48ManxPowerWhipsMcGee: the phone is behind a different NAT, I assume.
22:03.17WhipsMcGeeyeah, I've tried both externip= with my external ip and externhost= with a dynamic host that resolves to my IP.  neither works
22:03.26WhipsMcGeeyeah, the phone is behind my nat at home.
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22:08.49Synoptictzafrir : I'm getting nowhere, though errors are different now !
22:10.09Synoptictzafrir : no more module error.. which is good, but the executable file zaptel is nowhere to be found, and my /dev/ctl/zap is non existen.
22:10.43Synoptictzafrir :a hh nevermind
22:10.51Synopticit seems to work after a depmod -a
22:10.55Synopticlet me reboot to see
22:12.31Synoptictzafrir : hmm, /dev/zap is non existent..
22:12.40Synoptici'musing udev, is that a problem ?
22:13.32*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135)
22:13.41tzafrirSynoptic, for starters, what is the output of: cat /proc/zaptel/*
22:14.22Synoptictzafrir : befire I answer your question, I noticed that my zaptel modules are not loaded at boot, but they do work with a modprobe
22:14.38Synopticmodprobe zaptel and my /dev/zap appears
22:14.49Synopticno I need them to load at boot
22:15.23tzafrirSynoptic, gentoo has a file (/etc/modules-2.6 ?) for a list of modules to load at startup, IIRC
22:15.28Synopticyes
22:15.30Synoptichold on
22:15.37tzafrirBut I wonder why the device is not hotplugged
22:15.37*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
22:18.16Synopticbecause it is absent for the moment
22:18.41Synopticit's in my live tribox box (which I'm going to REMOVE cause I dislike it a lot)
22:19.16Synopticnow
22:19.24Synopticlet's assume them odule loading problem is solved
22:19.35Synopticthe make config isn't working
22:20.00Synopticmake: zaptel: Command not found
22:21.13WhipsMcGeeMaybe it's something else.  If I put both my asterisk server and my sip in DMZ I still get nothing
22:22.17*** join/#asterisk craigk (n=craigk@58.174.150.119)
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22:24.10Synoptictzafrir : ok, i'm getting it to work
22:24.21Synoptichad to tweak small things in my systems
22:24.23Synopticrebooting
22:24.51tzafrir<Synoptic> make: zaptel: Command not found   ---  where exactly?
22:25.48tzafrirany chance you have no perl on your system?
22:25.50Synoptictzafrir : well, command is found, it's the exec file functions it calls that was not found. it was named functions.sh. it is fixed now
22:26.32Synoptictzafrir : maybe i have no perl, but now I get the no telephony device message, which is good, and true. I have a X100p clone that I can throw-in for test purpose since I cannot remove my TDM400p from thelive system
22:27.33tzafrirSynoptic, kernel/xpp/utils/zaptel_hardware
22:28.00tzafrir(unless you have no perl)
22:29.04r0d3nt<SecNews> Title: (Critical Updates) Asterisk 1.2.27, 1.4.18.1, 1.4.19-rc3, 1.6.0-beta6 Released
22:29.04r0d3nt<SecNews> Link: http://www.asterisk.org/node/48466
22:29.04r0d3nt<SecNews> Description: The Asterisk.org development team has released four new versions of Asterisk to address critical security vulnerabilities.
22:29.07r0d3ntopps
22:29.18*** join/#asterisk CVirus (n=GoD@82.201.222.52)
22:33.01*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
22:33.19Synoptictzafrir : i have no perl. my x100p is in, but zttool reports no x100p
22:33.56Synoptictzafrir : i have to go play my game of badminton. I'll try to catch ya later, ok ?
22:34.14tzafrirSynoptic, modprobe wcfxo
22:34.33tzafriror just install perl...
22:34.37Synoptictzafrir : I do have perl..
22:34.45SynopticModule Size Used by
22:34.45Synopticzttranscode 8840 0
22:34.46Synopticztdynamic 11472 0
22:34.46Synopticztdummy 5032 0
22:34.46Synopticwctdm 37952 0
22:34.46Synopticwcfxo 12576 0
22:34.48Synopticzaptel 193156 5 zttranscode,ztdynamic,ztdummy,wctdm,wcfxo
22:34.50Synopticsorry for the flood
22:34.58Synopticdev-lang/perl-5.8.8-r4
22:35.09Synopticwcfxo is loaded but not used
22:35.57tzafrirso what's the output of zaptel_hardware ?
22:36.10Synoptici dont have that executable file.
22:36.19Synopticbtw my zaptel.conf is still untouched.
22:36.26tzafrir(a pastable lspci)  kernel/xpp/utils/zaptel_hardware
22:36.57Synoptic01:00.0 Communication controller: Motorola Wildcard X100P
22:37.12Greek-Boyputnopvut you were right. most dependencies required the dev package too. thank you
22:37.15Synopticmy clone card (which works btw.. poor quality, but it does work for the purpose of installing zxaptel)
22:37.28putnopvutGreek-Boy: glad to help
22:37.40*** join/#asterisk mvicha (n=someaddr@201.234.95.162)
22:37.44Synoptic01:00.0 Communication controller: Motorola Wildcard X100P
22:37.44SynopticKernel modules: wcfxo
22:37.48Synopticit does use it
22:39.01mvichahello everybody, can ne1 help me configure a digium te120p?
22:39.54Nuggetdigium can.
22:40.22Greek-Boynow I'm trying to find the debian package for imap_tk
22:41.39Synopticasterisk kernel # zaptel_hardware
22:41.39Synopticpci:0000:01:00.0 wcfxo- 1057:5608 Wildcard X100P
22:41.46SynopticI just found how to compile the tool :)
22:41.46Synopticok
22:41.47Synopticgtg
22:42.04Synoptici'll catch ya later, thabnx for your patience tzafrir
22:42.39seanbrighthow do you cancel an attended transfer?
22:44.32*** join/#asterisk mvicha (n=someaddr@201.234.95.162)
22:44.38mvichahello everybody, can ne1 help me configure 01:01.0 Ethernet controller: Digium, Inc. Wildcard TE120P single-span T1/E1/J1 card (rev 11)
22:45.40mvichaor at least how to test if it's configured correctly?
22:46.31russellbmvicha: support@digium.com
22:48.24mvicharussellb, that's too much for just a simple test I think... I need just a basic configure :s
22:48.52Greek-Boywhere do I get myself astersik core sounds since the ones on the asterisk site are only 55bytes in size and are obviously not intact?
22:49.15QwellGreek-Boy: make menuselect, edit the sounds options, then make install
22:49.50Greek-Boygreat
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23:00.22djc123hrm.. voicemail.conf... externnotify.. how can my script distinguish between the cases 'a voicemail was just left' and 'voicemail was checked'..
23:00.54djc123it would be more useful to have a script only called when a new vm is left, and pass it the # of the message, or even the filename to it
23:01.31djc123i want to email new vm's to myself, but I want mp3.. since it doesnt support mp3 directly, i will have to run a script to use sox.. BUT.. I only want it to email me a message when a new one is left, not anytime I might check it manually
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23:07.18djc123to be honest, id almost like to suppress the entire normal voicemail system, and *just* have new voicemails sent as email attachments in mp3 format
23:07.54djc123visual voicemail for iphones that dont have ATT = setup seperate pop account, have voicemails come to that mailbox as mp3..
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23:19.43Rico29for realtime mode, if I add fields in the database, like "bandwith" or anything else which is not in http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip database, will they be ... used by asterisk ?
23:20.32*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
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23:40.52mfedykRico29: some yes, some no.  useragent, no.  trustrpid, yes
23:40.52mfedykRico29: try it and see
23:41.09Rico29ok
23:41.10Rico29thx
23:45.45Rico29looks like it works with bandwith
23:45.49*** join/#asterisk talntid (n=swarm@66.208.251.174)
23:45.53talntidHi guys!
23:45.58jameswf-homesee it before its moderated off :)) http://www.trixbox.org/forums/trixbox-pro/trixbox-pro-help/paid-telephone-support-really-sucks
23:46.07Rico29good night : 1am here.
23:46.10Rico29:]
23:47.24talntidSo I have a Nortel BCM 400 call box... and want to switch to Asterisk... possible?
23:47.48talntidI would like to eliminate the BCM 400, and use Asterisk to perform its duties..
23:48.02talntidThat is what Asterisk is for, right?
23:48.21Rico29with realtime, y dont see my peers with a "sip show peers". How can I see them ?
23:48.40Rico29tainted_ > dont know, sorry
23:49.12Rico29talntid :]
23:49.51fujinmm, app_page and rick astley = win
23:49.58jameswf-homeassuming Nortel BCM 400 is a PBX yes talntid
23:50.12talntidIt is.
23:50.18jameswf-home~rickroll
23:50.18jbotfrom memory, rickroll is http://www.internetisseriousbusiness.com, or http://www.xkcd.com/396/
23:50.33fujinor http://www.mylazysunday.com
23:51.02jameswf-homedont forget the flea market
23:51.06jameswf-home~fleamarket
23:51.07jbotFleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E
23:51.07talntidHmm...
23:51.15talntidSo which one of you wants to come set me up on Asterisk? :)
23:51.36jameswf-homethat would depend on your budget
23:51.42jameswf-homeI am not a cheap date
23:51.43Rico29sip show peer <peer-name> load
23:51.58talntidI'll pay ya $1700 to come do it :)
23:52.05Rico29can I make the dynamic peers viewable by default ?
23:52.07jameswf-home+ expenses
23:52.08talntidplus pizza ;P
23:52.24Rico29pleae
23:52.26Rico29please
23:52.47talntidyou pretty good at Asterisk, james? :P
23:53.08jblackman. That Obama speech is incredible. I think it's one of the top 10 american speeches of all time.
23:53.22jameswf-homeplays with it 10 hours a day
23:53.26talntideep.
23:53.41Rico29jameswf-home > any solution tu my little problem ?
23:53.46Rico29to*
23:53.51talntidcan ya answer some Q's for me? ones that I could probably answer myself by googling for a few hours...
23:54.08talntiddon't want to impose, but you could probably answer them in 5 minutes
23:54.14jameswf-home[TK]D-Fender: is the google proxy
23:54.19jblackI'm not as good as jameswf-home, but I'll work for that
23:54.41jameswf-home~[TK]D-Fender
23:54.42jbot[TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy.
23:54.49jameswf-homesee
23:54.52jameswf-home:)
23:55.23talntidI run an outbound call center. I employ 30 sales reps who make primarily outbound calls.
23:55.41jblackok
23:55.48talntidI currently have a BCM 400, without the ability to record calls ($13,000), and recently got the RSI telecost software to get call details
23:55.57Rico29rtcachefriends=no in sip.conf doesn't change anything
23:56.26talntidI gotta say, I really dislike the BCM 400. I hate having to pay $13,000 to record calls.
23:56.33jblackThat's silly.
23:56.35talntidso I won't pay that much.
23:56.41jblackI record every call I get
23:56.43talntidDoes asterisk allow for that?
23:56.48talntidok, so obviously it does
23:56.48jameswf-homeAsterisk will record and log  calls........
23:57.09talntidAny downsides?
23:57.22jameswf-hometalntid: you may wanna try elastix... give you all the features and easy for a novice to admin
23:57.25jblackand numerous things are logged by default, and many other things can be logged too. And if you want even more data than that, you can even dump more data into an sql server.
23:57.27talntidI am a very DIYer...
23:57.34talntidI can figure stuff out pretty well... :)
23:57.53talntidWill it work with my current phones? and How?
23:58.05jameswf-homewhat are your current phones
23:58.13jblackThat's as far as I'm going until I get a deposit towards that promised 1700. :)
23:58.36talntidelastix only has support for 12 concurrent calls.. i need more ;)
23:59.04jblackthat's no problem.
23:59.07jameswf-homesays who?
23:59.19talntidsays elastix
23:59.27talntidElastix Appliance ELX-025
23:59.27jameswf-homeconcurrent calls 100% hardware dependent up to like 400
23:59.42talntidsorry, was just reading main page
23:59.45jameswf-homeno no buy a server install elastix
23:59.53talntidok. I have a server

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