IRC log for #asterisk on 20090205

00:00.53Gopher_77[TK]D-Fender: ok so I found one from a call; I didn't realize it got buried in so much ping junk. Here's from a call: http://nopaste.com/p/atEW5wSBU
00:01.25*** join/#asterisk mosty (n=mosty@eth1426.vic.adsl.internode.on.net)
00:02.20StanManCanFender: It's too bad there isn't somebody actually helpful here, every time I've came in for help you've beene nothing but a sarcastic dick.
00:02.34[TK]D-FenderGopher_77: Reliably Transmitting (NAT) to 194.0.210.251:5060: <- they are out on the public internet
00:02.52Gopher_77StanManCan: he may be a dick sometimes but he's like House: he knows his stuff
00:02.54[TK]D-FenderGopher_77: Contact: <sip:s@192.168.1.48> <-- and you have not set your system up properly to work from behind NAT
00:03.02*** join/#asterisk matthew___i (n=matthew@adsl-163-41-84.hsv.bellsouth.net)
00:03.25[TK]D-FenderStanManCan: Numbers matter because precice patterns matter.  If they don't match, it doesn't work and you're hiding the EVIDENCE
00:03.27Gopher_77[TK]D-Fender: so can you tell me what's wrong with my sip.conf?
00:03.32[TK]D-FenderStanManCan: and being an IDIOT about it it
00:03.41[TK]D-FenderStanManCan: We see you kind in here all the time.
00:03.52[TK]D-FenderStanManCan: No, never MY fault.... and show NOTHING.
00:03.58[TK]D-FenderStanManCan: You will not get far like this
00:04.17StanManCanFender: The configs came directory from my provider, VOIP.ms
00:04.30[TK]D-FenderGopher_77: You did not set your sip.conf up properly to work behind NAT.  Please follow the guide.
00:04.32[TK]D-Fender~sipnat
00:04.33jbot[~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
00:04.44[TK]D-FenderStanManCan: And that claim means absolutely nothing.
00:04.49*** join/#asterisk saftsack (n=oliver@g228071052.adsl.alicedsl.de)
00:04.53[TK]D-FenderStanManCan: I've seen plenty of providers give wrong configs
00:05.07[TK]D-FenderStanManCan: I corrected a Broadvoice one just last week.
00:06.00[TK]D-FenderStanManCan: I do not care about "stories".  You want to fix it, you need to look at the ACTUAL debug and configs
00:06.09[TK]D-FenderStanManCan: everything else is a sad joke of an attempt
00:07.30[TK]D-FenderGopher_77:   -- Executing [913309318902@internal:1] Dial("DAHDI/4-1", "SIP/voipuser/0013309318902") in new stack <- * didn't even try to dial.  Lets look at your peer config, and PB up "sip show peers" and "sip show peer voipuser" as well.
00:07.37[TK]D-FenderGopher_77: Mask only passwords
00:07.42joesuffcerenARRRGH!!!*sigh* *pulls out hair*
00:07.45joesuffcerenDoes anyone know of a tapi driver for asterisk that will work on a 64 bit windows OS? I've found only one that claims to be able to do that (xtelsio), but I can't get it to install on either of the 64 bit machines I've tried.
00:08.31*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.90)
00:10.35[8none1]64bit windows == death destruction and mayhem
00:11.17[TK]D-Fenderjoesuffceren: Best of luck with that combination...
00:11.29*** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM]
00:11.32[TK]D-Fenderjoesuffceren: maybe you could pay them to develop it.
00:11.44joesuffcerenthem being digium?
00:11.49voxterCan any of you guys think of an easier way to manage building a meetme out of people that you are inviting, rather than calling them individually and transferring them to your meetme ext? (or using a DID)
00:11.58[TK]D-Fenderjoesuffceren: No, Xtelsio clearly
00:12.18[TK]D-Fenderjoesuffceren: they should only have to do a few adaptations probably.  Did you like their 32bit ver?
00:12.21joesuffcerengotcha, well, xtelsio already has a 64 bit version...lol. it just doesn't work
00:12.35[TK]D-Fenderjoesuffceren: Sit back and wait or keep looking.
00:12.35frogonwheelsvoxter: I've seen a few solutions around.. adding extra commands to meetme to get them in more easily.
00:13.00frogonwheelsvoxter: you could use .call files :)
00:13.00voxterfrogonwheels: yeah i just thought of adding a custom feature code like *5<number> which will generate a call file to the number you specified, and auto-dump them into a meetme.
00:13.40voxterfrogonwheels: this customer of mine insists on being able to create >3 user meetme's by calling the people from their phone directly
00:13.43frogonwheelsvoxter: there's also implementations that use  features.conf stuff as well.
00:13.47cesau2anyone have a link to a good example sip.conf of using asterisk solely as a sip client?
00:13.51voxterfrogonwheels: yeah thats what i meatn
00:14.03joesuffceren*sigh* that was the conclusion I came to as well. my other thought was to run the tapi on a 32 bit box and use the ms provided remote tapiprovider on the 64 bit box, but it was developed in the days where any phone line was good enough, so there's no way to distinguish which extension the calls go to
00:14.09joesuffcerenso... worthless
00:14.10[TK]D-Fendercesau2: take ANY ITSP's config samples.  There you go
00:14.24joesuffcerenokay, I guess I'll just keep bugging xtelsio to get it to work.
00:14.26joesuffcerenthanks
00:15.32*** join/#asterisk prxtien (n=proleone@ppp121-45-69-101.lns10.adl6.internode.on.net)
00:15.37frogonwheelsvoxter:  http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
00:15.45voxterfrogonwheels: just found that.. thanks!
00:16.12prxtienhey all... ive got what i think is a RTP/firewall issue... when i make a call in one direction it works fine, both users can hear sound, but when the remote site calls me, only I can hear sound, they cant hear anything
00:17.05[TK]D-Fenderprxtien: read up :
00:17.07[TK]D-Fender~sipnat
00:17.08jbot[~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
00:17.22Gopher_77~sip.conf
00:17.25prxtienokay ill check that out
00:17.53[8none1]frogonwheels, nice solution. I haven't seen that one before.
00:18.41frogonwheels[8none1]: I'd seen it b4 but not used it yet.
00:19.10*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)
00:20.52*** join/#asterisk ipguy (n=ipguy@129.94.190.121)
00:20.55ipguyhi all
00:21.19prxtien[TK]D-Fender, im not sure nat applies as they are communicating over a vpn tunnel
00:21.22*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
00:21.23ipguyis it possible to NOT allow a specific user to make TRUNK calls ?
00:21.26prxtienso as far as the phones are concerned, they are local network
00:21.43[TK]D-Fenderprxtien: show us a complete call including the failure with SIP DEBUG
00:21.49*** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM]
00:21.50[TK]D-Fenderprxtien: PASTEBIN it all up
00:21.52[TK]D-Fender~pb
00:21.53jbot[~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, or , http://bin.cakephp.org/
00:22.08[TK]D-Fenderipguy: Sure... YOU control what CONTEXT they point to.
00:23.14prxtien[TK]D-Fender, i cant do that right now, but i will post again when i can, thanks
00:24.23*** part/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
00:25.08*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
00:25.24Gopher_77[TK]D-Fender: here is the configuration as I understood it should be from the first page: http://nopaste.com/p/arFYgwVjI and the results of a call: http://nopaste.com/p/aQqQq21BI
00:25.41Gopher_77[TK]D-Fender: still not working, but different messages
00:27.11[TK]D-FenderGopher_77: Contact: <sip:s@192.168.1.48> <---- stll bad
00:27.40[TK]D-FenderGopher_77: and AGAIN * didn't even TRY to call them.
00:28.10[TK]D-FenderGopher_77: externip=jelutz.homeip.net ; this is our router's WAN IP.
00:28.17[TK]D-FenderGopher_77: that is not an **IP**
00:28.30Gopher_77[TK]D-Fender: I was told it would work with my dyndns
00:28.35Gopher_77[TK]D-Fender: that was wrong?
00:28.44[TK]D-FenderGopher_77: for a HOST you need to use "externhost=thehosthere" and "externrefresh=120" (or whatever in seconds...
00:29.04frogonwheelsipguy: sure.. you can give them a different starting context..
00:29.12frogonwheelsipguy: and you can do what you like..
00:29.14*** part/#asterisk slashdotfx (n=slashdot@ip-252-29.mitra.net.id)
00:29.30[TK]D-FenderGopher_77: and [voipuser] has no HOST!  There is NOWHERE to send the call to!
00:29.31*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
00:29.32frogonwheelsipguy: oh. sorry.. [TK] replied.
00:29.51Gopher_77[TK]D-Fender: somehow it got my IP correctly before, the 77 one
00:30.05[TK]D-Fenderfrogonwheels: :)
00:30.20Gopher_77[TK]D-Fender: That tutorial you refered me to didn't mention a HOST, is it a bad tutorial?
00:30.30Gopher_77[TK]D-Fender: sorry, but I don't know these options at all
00:30.49[TK]D-FenderGopher_77: No, it is a GREAT tutorial, and does EXACTLY what it says.  You were the one who took "liberties" with what you read
00:31.02[TK]D-Fender:p
00:31.46Gopher_77[TK]D-Fender: the only liberty I took was putting the register line in there and changing out the fillers with real info, and assuming that I needed nat=yes for this provider
00:32.18[TK]D-FenderGopher_77: OH, and the guides SPECIFICALLY told you that for providers you DON'T put "nat=yes"
00:32.36Gopher_77[TK]D-Fender: so then I need all the information in the register line also in separate lines in the entry?
00:32.38[TK]D-FenderGopher_77: You've missed the mark a little here...
00:32.51mostyi think i have spotted a regression between 1.4.23-rc2 and 1.4.23.1 - hold music restarts with each Dia(...|m) command in 1.4.23.1, whereas it previously continued from where it left off. are there any settings which may have caused this on my part, or should i report this as a bug?
00:33.14[TK]D-FenderGopher_77: And as also said you did not put ANY host line for your PEER.  You CANNOT call them.  Ever.
00:33.23[TK]D-FenderGopher_77: * has nowhere to contact
00:35.06harry_vTK, have you ever seen a polycom phone not log into a asterisk dhcp server when there is another dhcp server on the same network?
00:35.08Gopher_77[TK]D-Fender: So I just read the tutorial again... did you send me the wrong one?
00:35.13harry_vI mean there would be a conflict.
00:35.21[TK]D-FenderGopher_77: NO
00:35.37Gopher_77[TK]D-Fender: oh, it was jbot
00:35.38[TK]D-Fenderharry_v: 2 DHCP servers on the same subnet?
00:35.39*** join/#asterisk s0uls (n=s@port-87-234-30-24.static.qsc.de)
00:35.49[TK]D-FenderGopher_77: No, the same FIRST link of the two
00:36.02s0ulshiho everybody
00:36.05[TK]D-FenderGopher_77: And I prompted jbot to link you those
00:36.19s0ulsanyone here by chance using a cisco 7940 with skinny/sccp in an asterisk?
00:36.28Gopher_77[TK]D-Fender: it doesn't have any of the HOST information there, or peer (only friend), etc
00:37.09[TK]D-FenderGopher_77: Every peer requires a HOST LINE!
00:37.25harry_vi dont think the phones have the ability to select a given dhcp. there was a case where a company did not want to put in new cabeling and thay had a dhcp network running.
00:37.30[TK]D-FenderGopher_77: this points to your PROVIDER
00:37.41[TK]D-FenderGopher_77: I can't tell you the address of THEIR SERVER you know
00:38.09joesuffcerenharry_v: you cannot have two dhcp servers on the same subnet
00:38.28[TK]D-Fenderjoesuffceren: I know...
00:38.34harry_vive always know that but that but wanted to know if there was any other way
00:38.37[TK]D-Fenderjoesuffceren: I jsut wanted to see him say "yes' :p
00:38.47[TK]D-Fenderharry_v: immensly stupid <-
00:38.52s0ulsi am having some trouble with the SEPMac.cnf.xml file for the 7940, or maybe i dont understand something correctly
00:38.54joesuffceren[TK]D-Fender: sorry to ruin your fun
00:39.02harry_vno im not. your just very critical
00:39.10s0ulsanyone here that might have some clue or a sample file?
00:39.24joesuffcerenharry_v: you can have your asterisk box serve tftp for the phones to boot from, but you'll need to tell the existing dhcp server to hand out the asterisk server's ip as a tftp server
00:39.25bmoracahas anyone here used an SPA8000 before?
00:39.31[TK]D-Fenderharry_v: That action... did not apply to you as a whole.
00:39.36[TK]D-Fenderharry_v: 1 very bad idea
00:40.01joesuffcerenharry_v: what are you using for a dhcp server
00:40.10harry_vasterisk box.
00:40.13[TK]D-Fenderjoesuffceren: Forgot the (s) :p
00:40.30joesuffceren[TK]D-Fender: lol
00:40.34joesuffcerenharry_v: the other one
00:40.37joesuffcerenthe one you want to keep
00:40.53harry_vthat is the clients network and is controled country wide.
00:41.06[TK]D-Fenderharry_v: You cannot do that.  1 DHCP server, period.
00:41.07joesuffcerenin other words, you have no access to it?
00:41.13harry_vI know
00:41.14harry_v:)
00:41.21bmoracayou'll need to have them set DHCP option 66 then.  if they won't, then you need to manually config the phones.
00:41.28[TK]D-Fenderyup
00:41.35joesuffcerenthen, you're up a creek unless you put in a new switch that will do vlans maybe?
00:42.03harry_vmanually configing the phones might be the easiest way to get around this issue.
00:42.14bmoracaharry_v:  you can manually config the phones to look at a TFTP server...you don't need to manually enter the entire config into the phone.  that's what i meant by manually configuring them.
00:42.38bmoracawell, you can do that with most phones anyway
00:42.55harry_vjust ways where I can meet the customers buget constraints
00:42.59bmoracathat's what i've done in the past when the client had either a crappy DHCP server that didn't support it or was being too much of an ass to let me do it right.
00:43.31harry_vstatic asterisk box with boot server on it
00:43.52Gopher_77[TK]D-Fender: I have the feeling that you've been a dick with me but I got something valuable from you: that registering and calling out aren't necessarily related, correct?
00:44.17[TK]D-FenderGopher_77: Nope, not at all, and I have not been a "dick"  withyou
00:44.29[TK]D-FenderGopher_77: I took it that you had a sane peer to call your host.
00:45.03[TK]D-FenderGopher_77: And you don't unfortunately.  You should know what to enter for this from them.
00:45.27[TK]D-FenderGopher_77: And the guide was very explicit with all of the parameters related to your NAT issues
00:47.27[TK]D-FenderGopher_77: http://www.aocomputing.net/?p=3#more-3 <- there... has "host" now.
00:48.13Gopher_77[TK]D-Fender: just "dynamic"?
00:48.35[TK]D-FenderGopher_77: this is a SAMPLE.  this is NOT the value you will use for YOUR ITSP
00:50.09*** join/#asterisk saftsack (n=oliver@g227070167.adsl.alicedsl.de)
00:50.11harry_vTK, a vlan would take care of the dhcp issue that I have talked about right?
00:50.17*** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net)
00:50.34[TK]D-Fenderharry_v: If the rest of their infrastructure supports it, probably
00:50.38Gopher_77[TK]D-Fender: aha now I'm getting 403 call denied
00:51.09[TK]D-FenderGopher_77: Good... now they hear you..... and just don't LIKE you :)
00:51.13Gopher_77[TK]D-Fender: status only "CONGESTION"
00:51.21Gopher_77:)
00:51.25[TK]D-FenderGopher_77: *'s message is meaningless.
00:51.32[TK]D-FenderGopher_77: the 403 is progress
00:51.40[TK]D-FenderGopher_77: Now fix your AUTH
00:52.09Gopher_77[TK]D-Fender: AUTH?
00:52.34[TK]D-FenderGopher_77: Authorization information.  User, password, encryption, etc
00:52.58[TK]D-FenderGopher_77: You aren't identifying yourself to them properly
00:53.17[TK]D-FenderGopher_77: [voipuser] doesn't even have a secret set
00:53.37[TK]D-FenderGopher_77: I think you seriously need to sit down with the book and look for a setup sample for your provider
00:54.02*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-016b1eeaa5c63298)
00:54.19voxterah another final question that you might know fender..
00:54.22citywokanybody have a recommended cordless sip phone?
00:54.50voxterdoes polycom have the ability to do blf on a speed dial key? ie it will light up to tell me so and so is on the phone, but if i press it (while lit) it will dial an extension that I can set to 'call barge' ?
00:54.56Gopher_77[TK]D-Fender: it would help if any information on the net was correct/clear
00:55.04saftsacki won't buy a snom m3 because of the bad accumulator
00:55.40[TK]D-Fendervoxter: A polycom speed-dial on a line key certainly supports presence (except IP 301), but will only dial ONE number, not something conditional that is state-based
00:55.47saftsackcitywok, in which country do you live?
00:55.55voxter[TK]D-Fender: ok so i have to check state in the dialplan then.
00:56.02citywoksaftsack: i live in the US, but the phone would be used in the Philippines
00:56.15voxter[TK]D-Fender: can you subscribe to watch, say, extension 100, but if you press it, it dials **100 ?
00:56.19[TK]D-Fendervoxter: Yup.  very easy in 1.6, minor trickery in 1.4-
00:56.25saftsackdo you plan to use it in an area that is bigger than a common dect range?
00:56.45frogonwheelssaftsack:  oh? I was thinking of getting an M3..  accumulator?
00:57.11saftsackthe battery
00:57.38citywokits a 2bdrm apartment
00:57.43saftsacki bought a snom m3 and compared to analog siemens gigaset handsets the battery lifetime was very poor
00:57.47citywokmaybe 1200sqft
00:58.05saftsackcitywok, that's ok for one base station
00:58.40citywokyea i would assume so, what are the "reliable" phones in the cordless arena? are they as crappy as every other cordless house phone i've ever used?
00:58.59saftsackif that works i would take a common dect handset with a m3 basestation if this is possible
00:59.28saftsackcrappy? here in germany we use telephones called siemens gigaset. they are as reliable as wired phones
00:59.39*** join/#asterisk ingenius (n=alektro@host253.190-30-205.telecom.net.ar)
00:59.42s0ulshehhe, DECT
00:59.46s0ulswho wants DECT these days ..
01:00.10saftsacks0uls, in my opinion dect is better if battery is important
01:00.28voxters0uls: what else would you pick?
01:00.32saftsacki dont think that wifi phones have the same battery lifetime
01:00.48s0ulsmobile phone?
01:00.52frogonwheelsyeah, going on the way the wifi sucks the juice out of my mobile :)
01:00.54Gopher_77[TK]D-Fender: looks like "setup" was the key word to search for to get help from my provider :)
01:01.00[TK]D-FenderI have an Aastra 57i CT covering a 60,000sqft warehouse quite well
01:01.01s0ulsi was only referring to the security issues of DECT actually, i would wanna use that anymore
01:01.06Gopher_77[TK]D-Fender: I've been searching for days
01:01.15florzhowever, DECT is basically broken, while WPA2 still is not ...
01:01.17[TK]D-FenderGopher_77: Bet you're kicking yourself now :)
01:01.19voxterwifi phones are a horrible idea are you kidding me? battery lifetime will always suffer to do wireless beacon times.. and wifi is also far too sensitive to common frequencies cuasing interference
01:01.23saftsacks0uls, you are talking from com on air cards? :)
01:01.31Gopher_77[TK]D-Fender: nah
01:01.39[TK]D-FenderGopher_77: If you're having trouble, I'll gladly lend a hand...
01:01.44s0ulssaftsack: i am more talking about the DECT implemantation, the com on air is only a piece of hardware
01:01.44[TK]D-Fenders/hand/foot/
01:01.47[TK]D-Fender:p
01:02.05Gopher_77[TK]D-Fender: :P
01:02.18saftsacks0uls, yes but this is the hardware which is commonly used by basement kiddies which are "hacking" dect
01:02.31s0ulswhat basement kiddies hack DECT?
01:02.36saftsackvoxter, true
01:03.05saftsackso i would prefer an established dect phone
01:03.10s0ulsthe people that broke DECT are all acknowledged academics
01:03.28[TK]D-FenderGopher_77: http://www.aocomputing.net/?p=3 <- there updated for "host=" AND for the "externhost" bits
01:03.46s0ulsanyways, seriously no one running SCCP/Skinny with cisco 79xx?
01:03.50saftsacks0uls, yeah that was university of darmstadt but everybody can use it
01:03.50*** join/#asterisk squish102 (n=squish10@cpe-098-024-067-184.carolina.res.rr.com)
01:04.01s0ulssaftsack: actually thats only propaganda
01:04.07s0ulsjust one guy was from darmstadt
01:04.15s0ulsand its not like he was the head behind it
01:04.30saftsacki was born in darmstadt so it's nice for me to see is as that
01:04.46s0ulswell, its a shame they claim all the fame
01:04.49s0ulsactually
01:04.50s0uls:)
01:04.53saftsack;)
01:04.59*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
01:05.14s0ulsas its really not like he did the major work on it
01:05.19saftsackso back to topic. whatÄs about citywok s phone?
01:05.30citywokevery m3 base station i've found is the snom one
01:06.14citywoki'm just going to connect it to * a few blocks away
01:06.25saftsackyes but maybe it can be used with another handset
01:07.35squish102being sort of a noob to asterisk/voip what is the best approch (other than becoming an expert in PBX overnight) asterisknow or freepbx or trixbox?
01:07.44saftsackso at all the m3 works and the speech quality is ok, but it is not a premium phone
01:07.54[TK]D-Fendersquish102: Approah to what?
01:08.07citywokhow about the aastra 420d?
01:08.18squish102getting it up and running and a voip trunk set up
01:08.27s0ulssaftsack: what battery lifetime are you looking for?
01:08.28squish102no hardware yet
01:08.45saftsacki dont search for a phone or what do you mean? :)
01:09.19citywokif the m3 isn't a good phone, i saw TK mention he uses an aastra and it works pretty well, i need something reliable that i dont have to worry about from 7000 miles away
01:09.22s0ulssorry, must have misread the context
01:09.47saftsacks0uls, ok. maybe 10 days standby and 10 hours talking time
01:09.51[TK]D-Fendersquish102: Elastix.  At least they don't own your ass like Trixbox does
01:10.08s0ulsok, havent seen a wireless one yet that does that :)
01:10.14squish102home use of pbx, with a grandcentral -> gizmo, a voipbuster, an attepmt at sype gateway.. and maybe a magicjack
01:10.27[TK]D-Fendersquish102: Just realize that any GUI approach means you're starting off backed into a corner and almost NOTHING you "learn" from them accounts for anything
01:10.51squish102ok thanks [TK]D-Fender
01:10.52saftsacks0uls, we have an old gigaset and this comes very near to this
01:11.12[TK]D-Fendersquish102: Ah! Every cheap-skate stupid little scam.... ALL AT ONCE!  Well why didn't you say so!
01:11.22squish102the end result is that i will be trying to get it working on my mythtvbox
01:11.46*** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com)
01:11.46saftsacks0uls, whats then? handset -> basestation -sip-> ???
01:11.48squish102for cheapest home phone use...
01:12.00saftsackhow do you connect to the net?
01:12.08squish102unless there is a better product aimed at me
01:12.18[TK]D-Fendersquish102: Good luck with that.... especially with a GUI running the show.
01:12.29[TK]D-Fendersquish102: There is no product aimed at you.
01:13.04saftsacks0uls, this is this old phone. the battery lifetime is impossible long http://rudi02.de/bilder/t-sinus700m.jpg
01:13.10squish102no home users that want to tinker around in voip?
01:13.12[TK]D-Fendersquish102: they guy who wants it all to "just work" without any learning curve and do EVERYTHING that requires "getting down & dirty" in order to cheat the system.
01:13.28[TK]D-Fendersquish102: Sure there are... the ones who LEARN Asterisk
01:13.48[TK]D-Fendersquish102: But you expect to get it all up and running quick.  THAT is not compatible
01:14.13s0ulssaftsack: hm?
01:14.51saftsackjust an example for long battery lifetime and quality
01:14.52[TK]D-Fender"Hi, I'm not an aeronautics engineer, but do you have a 'SAMS  Learn Aeronautics in 24 Hours' book laying around anywhere?"
01:15.16saftsacks0uls, take the m3. its ok
01:15.31saftsackif you need better quality take the aastra. nothing easier as that ;)
01:15.36squish102[TK]D-Fender, not to piss on anyones battery, but sometimes i wonder why everything that i install in linux requires you to understand indepth things that add no value... this is after learning much more than i care to about mythtv
01:15.42[TK]D-Fenders0uls: Specific warning : Those whose opinions I trust shat all over the M3
01:15.54s0ulsuhm ..
01:16.02s0ulsfolks, i am not looking for a phone at all
01:16.07s0ulsi got tons and am quite happy :)
01:16.11[TK]D-Fendersquish102: Because maybe this time they're RIGHT
01:16.33s0ulsi am actually just looking for someone familiar with SCCP/Skinny config of asterisk
01:16.49s0ulsor lets say .cnf.xml files for connections of cisco phones to asterisk
01:16.58saftsacks0uls, thory i wanted to talk to citywok
01:17.00[TK]D-Fendersquish102: All of the GUI's out there are for churning out dumb-toaster PBX's.  They were not made for the kind of psycho dialplan trickery to mix in 3 things like that with the kind of scripted cheating they'll require
01:17.00s0ulsi am somewhat not getting anywhere anymore here and dont get it
01:17.23*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
01:17.24squish102ahhh, ok i get it now
01:17.30citywoksaftsack: i think i like the aastra 420d and am gonna pitch that as an idea
01:17.52squish102[TK]D-Fender, i guess i have a lot of reading and wiki searching ahead of me
01:18.04[TK]D-Fendersquish102: Here's a START for you :
01:18.07[TK]D-Fender~jerjerguide
01:18.08jbot[~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * :  http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/
01:18.20squish102thanks
01:18.30[TK]D-Fendersquish102: use this as INSPIRATION for some of the minimal basics and you can grow stuff onto that basis.
01:18.45[TK]D-Fendersquish102: because the install process is a little differnt now.
01:19.01[TK]D-Fendersquish102: But its a good basis for a small setup
01:19.27saftsackdoes one of you take part on the digium beta test for the skype connection?
01:19.33[TK]D-Fendersquish102: You jsut want to make calls... taht means a couple of funky contexts because of GC's call-back crap (I said trickery, didn't I?)
01:20.07[TK]D-Fendersquish102: And you are looking to integrate 3 different services... that means you'll need some logic to help in choosing which to use & when
01:20.48[TK]D-Fendersaftsack: The first rule of SkypeForAsterisk is you do not talk about SkypeForAsterisk
01:21.03[TK]D-Fendersaftsack: The second rule of SkypeForAsterisk is you do NOT talk about SkypeForAsterisk
01:21.21icebrew54lol
01:21.32[TK]D-FenderHIS NAME IS ROBERT PAULSON!
01:21.32icebrew54I wish I could be part of that....it's going to be a nice add-on for skype
01:21.41icebrew54err asterisk :P
01:21.42saftsackafter clicking on the nda button :-P
01:21.50icebrew54bleh NDA this, NDA that
01:21.52[TK]D-FenderGreat.... Asterisk.... is a great add-on... FOR SKYPE
01:22.04icebrew54sorry I mispoke, long day for me
01:22.18[TK]D-Fendericebrew54: You're sense of prioritization is mighty FUBAR'd there :p
01:23.16icebrew54yeah I agree....I like asterisk far more than skype
01:23.23icebrew54skype would be a nice module to add to asterisk though
01:23.34saftsackicebrew54, full ack
01:23.37icebrew54very good "partnership" for both parties
01:23.50icebrew54asterisk will get plenty more face time...
01:24.02icebrew54does any other PBX feature a skype add-on?
01:24.07saftsackand it is called skype for asterisk and not asterisk for skype
01:24.09sipyTK and the rest: What are your thoughts about the gui's with asterisk. Does it hamper the operation at all?
01:24.23icebrew54saftsack: sue me man, I said skype when I meant asterisk
01:24.25sipyseru=iously
01:24.27icebrew54:P
01:24.38saftsack^^
01:24.39[TK]D-Fendersipy: *-GUI is not a "complete solution and well...
01:24.42[TK]D-Fender~users.conf
01:24.43jbotusers.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system.
01:24.43icebrew54sipy: I used web-gui and I regret it
01:25.04[TK]D-Fendersipy: As for FreePBX..... its completely owns your ass... but at least it does the whole job :)
01:25.12sipyIS it safe? ok for run of the mill small 10-25 phone systems?
01:25.29icebrew54yeah for a standard phone system, it seems to be a solid solution
01:25.39icebrew54if you want to start doing stuff out of the box, that's where I experienced failure
01:25.42[TK]D-Fendersipy: Of course its SAFE... *-GUI isn't COMPLETE and you have to actually code a whoel pile of stuff outside the GUI.
01:25.58sipyok..... without it, how do you get users to add a phone or do any simple programming?
01:26.02[TK]D-Fendersipy: And yes, FreePBX is safe, and can do a pretty complete job all within the GUI
01:26.37[TK]D-Fendersipy: If you want to know the nitty-gritty, you're just going to have to get off your ass and TRY IT
01:27.09sipyI ran * before freepbx.... I know the nitty gritty to some extent.
01:27.36sipyI learned it without the gui stuff years ago.
01:27.49Gopher_77~sip_notify.conf
01:27.50[TK]D-Fendersipy: I'm talking about learning just what parts you have to do yourself for *-GUI
01:27.58[TK]D-Fendersipy: thats the one you can't just leave with people
01:28.04saftsackwhat's with sipx? *duck*
01:28.07frogonwheelssipy: sias
01:28.09frogonwheels~sias
01:28.10jbotSuck It And See
01:28.47frogonwheelssipy: Find a gui that you can let your users play with - but that gives you some ability to tinker when it doesn't quite deliver.
01:30.02sipydoes it exist?
01:30.03VoipForces[TK]D-Fender: Strange I find freepbx fairly flexible. In fact there are hooks to every parts of it and you can recode the parts that you need. There is a custom hooks to every contexts
01:30.29sipyHow is PIAF? Does it write all the confs from the db?
01:30.40[TK]D-Fendersipy: FreePBX <---
01:30.57harry_vsipy, do you have your 4 soft keys configured?
01:31.32VoipForcessipy: I would recomment freepbx all the way. it gives the user the flexibility and give you room to write your own code as well
01:32.01harry_vhow is freepbx different then asterisk
01:32.05sipyharry, yes, depends on which menu item I am in, but basically yes.
01:32.07VoipForcesI have freepbx running install of over 160 phones.
01:32.16[TK]D-Fender"dialparties.agi" <- the ink you sign your soul away with
01:32.17VoipForcesharry_v: freepbx is just a web gui for asterisk
01:33.07harry_vokay
01:33.07VoipForces[TK]D-Fender: But you can modify it, or as I did on some "custom" install skip large part or it.
01:33.12harry_vsounds like php
01:33.30VoipForcesharry_v, yes freepbx is coded in php
01:33.46[TK]D-FenderVoipForces: But would you WANT to?  yuck
01:33.52sipyif you use the template for creating sets, do they show up on the freepbx page for extensions?
01:33.58VoipForcesharry_v: all it does is write for you the asterisk config files.
01:34.11harry_vI figured that was the case
01:34.17[TK]D-FenderVoipForces: For SMB chump installs, FreePBX is fine, just seriously avoid any thought of "crastivity"
01:34.18harry_vsame with trixbox and others.
01:34.25[TK]D-Fendercreativity*
01:34.38[TK]D-Fenderharry_v: Same thing
01:34.41[TK]D-Fenderonly forked
01:34.54harry_vTK, unless you modify the php code to allow creativity.
01:35.01sipyCan you do templates with the extensions?
01:35.07sipywith freePBX
01:35.09VoipForces[TK]D-Fender: crastivity???
01:35.28[TK]D-FenderVoipForces: I fixed the typo immediatly following... please catch up :p
01:35.36[TK]D-Fender(and am making NEW ones)
01:36.01VoipForces[TK]D-Fender: Sorry had one of my kids cmae down to ask for a glass of milk beforegoing to bed...
01:36.11[TK]D-Fendersipy: GO TRY IT
01:36.21harry_vyea, dont create a serurios topic you may get burned for it :)
01:36.26VoipForcessipy: There are no extension templates.
01:36.37harry_vor get heat for the spelling errors
01:36.38harry_v:)
01:36.52sipywill it break if you do your own extension conf with templates?
01:37.07[TK]D-Fenderharry_v>yea, dont create a serurios topic you may get burned for it :) <- not to fear, I'll take you all down with me :p
01:37.22[TK]D-Fendersipy: yes
01:37.24VoipForcessipy: all custom stuff must be done in the _custom.conf files. so there is a sip_custom.conf
01:37.38sipygreat!
01:37.51[TK]D-Fendersipy: you don't get to template "how some guy's phone gets dialed" for your funky rules
01:38.01VoipForcessipy: But those will not show in the gui configuration screens
01:38.58VoipForcessipy: Go and download then new asterisknow or trixbox, they are build with freepbx.
01:39.09harry_vmy other keyboard is new and has a sticky keyboard then people yell at me and say "Hey moron, stop with the spelling issues!" really though, if I do mispell something it is not because it was more a type then bad spelling :)
01:39.51harry_vnow, bad spelling can reall $&$%^ your dial plan :)
01:40.01sipyI have been playing with trixbox. I didn't like it cause you can't mess with the files. I was changing shit, reloading and it would go away.
01:40.02[TK]D-Fenderharry_v: In my defense, my spelling is only just short of perfect.  My TYPING on the other hand is a complete roll of the dice :)
01:40.17[TK]D-Fendersipy: that is FreePBX <----
01:40.43sipyhence the freePBX logo in the upper left?
01:40.46sipyhaha
01:41.04harry_vwould it be better to have a php on a laptop to make the files then transfer it to a asterisk box?
01:41.36harry_vno need to have a web server running on the asterisk box.
01:41.39sipyI ended up writing the files, copying them elsewhere and wrote a script to coppy them back when * restarted.
01:41.50sipythat didn't work to well though....:-)
01:41.56[TK]D-Fenderharry_v: Does it seriously matter?
01:42.06[TK]D-Fenderharry_v: You aren't running GOOGLE off it....
01:42.16sipyI didn't know what was going on until I said F&*k it and went into the forums and started using the gui.
01:42.44[TK]D-Fendersipy: Yup, it STILL looks DB stuff up (AstDB, etc) and nails you one way or another.
01:43.00harry_vastdb has issues?
01:43.26harry_vAnyway, need to get. gym time.
01:43.54harry_vsipy we will talk another time when you have time.
01:44.33sipyharry look me up in the other channel when you get back. I'll send you my phone configs and apps
01:44.39*** join/#asterisk HermesNeto (n=HermesNe@189.71.50.127)
01:45.25VoipForcessipy: that is because all the _additional.conf files are written by the guy.
01:45.34sipyIts pretty cool though if you don't need to dothe crazy shit some of you are doing.
01:45.42VoipForcessipy: you needed to write your custom stuff in the _custom.conf files.
01:46.17*** join/#asterisk dlewis (i=44c6d5a7@about/security/staff/dlewis)
01:46.40sipyyeah but you should be able to customize the conf files and hae it dynamic on the extension php page.
01:46.43VoipForcessipy: I have done some crazy stuff arounf freepbx, including wake-up calls, 911 calls from sip phones that would call 911 but would also call the front desk to say that a room had called 911,...
01:46.48harry_vokay
01:47.12sipyHow come I am getting conflicting stories here then....:-)
01:47.42sipyOne person says you can't customize and do crazy shit, now someone else says I do crazy shit with it.
01:48.06sipyI will play with it and see what I can get away with.
01:48.17VoipForcessipy: You just have to understand how freepbx config files are written. Hevery context in the _additional.conf files has a include context_custom. It's that custom context that allow you to overwrite with your own code.
01:48.27sipyThere is overhead though running it instead of just *
01:48.48VoipForcessipy: The only overhead is dialparties.agi.
01:49.15VoipForcessipy: That script takes care of all the DND, CallForward and ring group things.
01:49.17sipyweb server, modules, databases...and the list goes on.
01:49.59VoipForcessipy: It's just plain asterisk configuration files with 1 (maybe 2 agi script). That's it. All other stuff is just dialplan and astdb.
01:50.05sipywon't really make a diff though in a small office pbx though
01:50.24VoipForcessipy: FreePBX is also very modular, so you just install/enable modules you need/want.
01:51.00VoipForcessipy: No it won't. I have installs doint high call volume with it on multiple PRI without a glitch.
01:51.16sipyWhat would be cool is just an extensions module so the customer can add phone sets and do minor stuff. Is there access levels?
01:51.39sipyVoipForces: thats good to know
01:51.57VoipForcessipy: You you can create users with acess to only the extension modules.
01:52.00sipyyou running just frePBX? or other gui stuff?
01:52.11VoipForcessipy: You can ever restrict the extension range they are allowed to play in.
01:52.37VoipForcessipy: Just freepbx. minimalist install with only what's needed to run a corporate pbx.
01:52.48sipyI didn't see that part, That works. I didn't add users thats probably why I didn't see it.
01:53.10VoipForcessipy: Trixbox is not the best implementation
01:53.18*** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net)
01:53.18*** mode/#asterisk [+o Deeewayne] by ChanServ
01:53.19ricko73vim is my gui
01:53.33VoipForcessipy: They tried to put too mush useless stuff in it.
01:53.33Pryonemacs 4 lyfe yo
01:53.41icebrew54ricko73: gui? what is that?
01:54.54VoipForcesWell, tell that to a 10 person office that want to take care of their oen PBX. direct config file edit is great for us developers, but for your common mortal it sucks and it's just like having an other Nortel PBX they don't have control on.
01:55.02sipyhaha F vim
01:55.10sipypico por ever
01:55.22VoipForcesWith a GUI yuo can tell them, you have full control over 95% of what you need to do for your corporate pbx usage.
01:55.41ricko73Then when they break it, they can still call you to fix their mess
01:56.03VoipForcessipy: Try the new asterisknow with freepbx, it's centos based and much more minimalist as an installation than trixbox.
01:56.03ricko73end users aren't trained to work on phone systems
01:56.25sipyI've allready decided to ace trixbox
01:56.35VoipForcesriko73: I give a basic training and manual.
01:57.07VoipForcesAnd I give the administrator training to a single person who is at least a bit tech savvy.
01:57.24sipyI just created a PIAF, and will look at AsteriskNow. I just want to give the end user something they can program phones and do minor shit with.
01:57.33[TK]D-Fendersipy: Not sure if this point has struck home, but pretty much ALL of the GUI'd *'s out there use F-ING FREEPBX
01:57.38[TK]D-Fendersipy: PIAF = FreePBX
01:57.42VoipForcesFor one I would not want to buy a phone system where I would have enev less control over than a nortel...
01:57.45[TK]D-Fendersipy: Elastix = FreePBX
01:57.54sipyI know
01:58.03[TK]D-Fendersipy: AsteriskNOW .... Comes with it TOO in addition to their fractured GUI
01:58.19[TK]D-Fendersipy: Same shit, different smell.  Everything you dislike ... is the SAME
01:58.33sipyhehe,
01:58.36sipylovely!
01:59.30sipyWell, for the clients its nto that bad because they won't need nor even understand half the shit this is capable of.
02:00.15sipyIts like VoipForces said. end user will want to have some control on it.
02:00.39*** join/#asterisk highlevelcode (n=elmatado@adsl-75-51-95-24.dsl.lsan03.sbcglobal.net)
02:00.51sipyricko73 : some corps s=spend plenty of money on coice training.
02:01.15ricko73coice?
02:01.25sipyTyco sent me all together probably 1 year total training for lucent G3's now avaya and whoever it is now.
02:01.38sipyvoice sorry
02:01.46*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
02:02.08[TK]D-Fendersipy: Perfect because GUI's won't realize more than a tiny fraction of what * is capable of
02:02.14ricko73that's great, but for most companies who don't have someone dedicated onsite, they are usually better off calling you than messing with it on their own
02:02.40ricko73that's your choice though
02:03.33VoipForcesricko73: I rather do interesting stuff then being called to reset a voicemail password or change the name on a sip phone.
02:04.23sipyUsusally they will do what you train them to do. Add voicemail, change music, the small stuff.
02:04.44sipyand if they mess it up, they pay you to go fix what they messed up.
02:04.45ricko73right, and because they do it once every 2 months how much do they remember?
02:04.51sipywhats wrong with that?
02:04.57ricko73nothing
02:05.12ricko73I said that it's your choice
02:05.17VoipForcesWell, if you give them proper documentation it helps.
02:05.50VoipForcesI give a 50 page manuel on freepbx
02:07.01sipyI remember saying back when I first started using *, we had IDNX boxes compressing voice/hardware echo telabs cancelers,  about 3 million dollars wiorth of pbx in 8 offices across the country, I told him he could put 20,000.00 worth of hardware running * to replace that 3 mills worth of ?BS
02:08.23sipy$25,000.00 voicemail Audix..... JEEZ!!
02:08.38sipyand that was limited to 200 users
02:09.25VoipForcesLOL, i can't believe some vendors still limit the number of extensions or the number of voicemail boxes...
02:09.30VoipForcesThat's so lame.
02:09.46[TK]D-Fendersipy: $25,000 can buy a complete 200 phone * installation <-
02:10.23saftsackthats true
02:10.42*** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net)
02:11.09sipyI remember lucent/avaya downtalking linux saying it wasn't up to corporate standards. They were running linux in their freaking g3's
02:11.58saftsackwhat is g3?
02:12.03*** join/#asterisk adam000 (n=adam@h199.163.141.67.dynamic.ip.windstream.net)
02:12.36ricko73apple pc
02:12.48sipywhere are you going to get 200 phone sets for 25,000 that you would trust?
02:13.09[TK]D-Fendersipy: POLYCOM <-
02:13.12saftsackbuy 200 snom or polycom phones, one big server
02:13.23saftsacka second as backup
02:13.23VoipForcessipy: Aastra
02:13.27saftsackthere you are
02:13.45sipyAastra's for 179.00 each = $$$
02:13.50sipy200 of them
02:14.01saftsackpolycoms are much cheaper. snom too
02:14.17ricko73which models?
02:14.21saftsackif you calculate 100$ per phone you have 5000$ left for two servers
02:14.22VoipForcesSNOM I would use, Polycom, never again.
02:14.34ricko73Why not plcm?
02:14.41saftsackpolycom can be very nice if you use the provisioning function
02:14.44VoipForcesBut if you want quality phones Aastra is the way.
02:14.56saftsackand i would recommend provisioning by an environment of more than 20 phones
02:15.07*** join/#asterisk Khratos (n=Khratos@190.166.113.91)
02:15.09VoipForcesricko73: Bad experience, theyr configuration file format sucks.
02:15.12sipycan Aastra compete with polycom's quality in speaker phone?
02:15.22ricko73VoipForces: not if you really understand it
02:15.45VoipForcessipy: Yes.
02:15.57sipyI never used Aastra
02:16.12[TK]D-FenderVoipForces: What bad experience?
02:16.13ricko73Aastra's are good phones, but the speakers are not exactly as good as Polycom
02:16.22ricko73good, but not as good
02:16.31saftsacklet us stop talking about phone brands. everybody has another subjective opinion
02:16.53saftsackbut at all you can say, that aastra, polycom and snom are all good brands for phones
02:16.55icebrew54saftsack: question: should asterisk develop an open-source phone?
02:16.59VoipForcesricko73: well, true, I spend the last 4 years working with Aastra so I know them. Also have good engineering contact with Aastra. + I'm canadian and Aastra is Canadian, as well as Sangoma :-)
02:17.02ricko73http://www.kfife.com/voip/
02:17.04sipyshall we switch to which asterisk dist? isn't everything about opinions?
02:17.17ricko73Check that link for a good way to do Polycom configs
02:17.29ricko73not my page, but makes it easy to upgrade
02:17.41[TK]D-Fenderricko73: Yeah... its not like it isn't all documented left right & center...
02:17.44saftsackyes but there are discussions about more or less founded questions
02:18.00VoipForcesricko73: Thanks.
02:18.58saftsackbut for 5000$ you cannot buy a pots connection system for 200 phones + 2 servers so for 25000$ you just can use a sip/iax2 provider
02:19.15[TK]D-Fendersaftsack: OH?  Really?
02:19.18VoipForcesicebrew54: SNOM is as good a opensource phone as you will get.
02:19.26[TK]D-Fendersaftsack: I beg to differ...
02:19.31icebrew54openmoko?
02:19.49saftsack[TK]D-Fender, i would calculate 60 to 90 channels for 200 phones.
02:19.53VoipForcessaftsack: would you trust only pure voip trunks for a 200 seat firm?
02:20.02saftsackNO i wont
02:20.07[TK]D-FenderVoipForces: I don't personally care if my phones firmware is OSS... I care that its GOOD
02:20.14sipydialback copper
02:20.19saftsackthats because i beg that 25000$ arent realistic
02:20.26[TK]D-Fendersaftsack: for 200 phones I sure a shit wouldn't use POTS <-
02:20.30[TK]D-Fendersaftsack: PRI <-
02:20.35[TK]D-Fenderonly way to fly
02:20.42saftsackyes. you need three E1s
02:20.54[TK]D-Fendersaftsack: and <$1000 for the car.  that leaves $4000 for the server
02:21.13saftsackcar. = carrier?
02:21.23[TK]D-Fendersaftsack: 3 E1?   Who said you need to support 100% concurrency?
02:21.35VoipForcesTK: PRI indeed. mee too that's why I stick with Aastra.
02:21.46[TK]D-Fendersaftsack: How many PBX's do you see sold to non-call-centers that aren't 4-to-1 AGAINST?
02:22.02sipyno car = 1000 VW to get all that equipment to site.
02:22.11VoipForcesOk, need to go code (well, first do the dishes, then code) so see you guys around.
02:22.13saftsackwith 4 to 1 you mean 50 out/inbound lines for 200 users?
02:22.27[TK]D-Fendersaftsack: Yes, look at every key system.
02:22.42[TK]D-Fendersaftsack: Basics : Nortel 8x24, 6x16
02:22.49[TK]D-Fendersaftsack: Big multiples against
02:23.05saftsackso you would by 2 t1s for 200 users? ^^
02:23.10[TK]D-Fendersaftsack: Sure call centers have a lot more concurrency, but thats specialized and not a good assumption
02:23.27[TK]D-Fendersaftsack: Well I never pass 10 for my 40 users.
02:23.37[TK]D-Fendersaftsack: Ask around...
02:24.44saftsackok. i had to do this in a past as i built up an * for a distribution department. but you are right in another company it was 6 to 1
02:24.44*** join/#asterisk kank (n=anil@59.95.18.149)
02:25.08*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
02:26.26saftsackbut the 25000$ calculation is interesting ...i would buy 200 phones 100$ each, two desktop grade servers with ssds and a pri gateway with at least 2 ports. so 25000$ should be ok. you are right
02:26.28[TK]D-Fendersaftsack: Keep in mind, your telco can't support 50% concurrency :)
02:26.48[TK]D-Fendersaftsack: Yes... BARELY right, but I was squeezing the point by :)
02:26.49Gopher_77[TK]D-Fender: yes, I got it working!
02:26.57[TK]D-FenderGopher_77: Glad to hear
02:27.00Gopher_77[TK]D-Fender: voipuser's setup instructions were wrong
02:27.14[TK]D-FenderGopher_77: Say it ain't so!
02:27.36Gopher_77[TK]D-Fender: it was the difference between voipuser.org and sip.voipuser.org
02:35.23*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
02:35.29*** mode/#asterisk [+o russellb] by ChanServ
02:38.28*** join/#asterisk matthew___i (n=matthew@adsl-163-35-202.hsv.bellsouth.net)
02:42.40*** join/#asterisk keebler (n=keebler@h149.148.20.98.dynamic.ip.windstream.net)
02:42.45keeblerd
02:44.09*** join/#asterisk bmoraca (n=bmoraca@75.12.126.173)
02:46.53keeblerWhat's that url for the basic VOIP config with Asterisk. Johnny something?
02:47.46bmoracaexten => _.,1,Hangup() is the only one you'll need :D
02:47.48*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
02:48.37saftsackhehe
02:49.26Gopher_77[TK]D-Fender: great, now all the sudden without changing my sip.conf I'm getting the same stuff as before...
02:50.25*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)
02:50.46[TK]D-Fenderkeebler: ...
02:50.48[TK]D-Fender~book
02:50.49jbotwell, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
02:50.50[TK]D-Fender^^^ there!
02:51.14keeblerHeh, I'm reading the "bible" now
02:52.10Gopher_77~playback
02:52.31keeblerI can't remember the website. But its fine, I found what i need with the book from oreilly.
02:53.33*** part/#asterisk ipguy (n=ipguy@129.94.190.121)
03:01.14*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
03:04.24keeblerHhahaha Page 85 of the Asterisk handbook. Typically, it?s at 4 A.M. on the morning of a critical 8 A.M. meeting, after having worked all weekend. Red
03:04.28keeblerBull is probably the most popular drink of the Asterisk developers. Dr. Pepper would be a close second. Red
03:04.31keeblerBull, anyone?
03:04.43keeblerstares at his wall of Dr Pepper cans behind him.
03:04.55[TK]D-Fenderdoes not require such things.
03:05.14[TK]D-Fenderis already NASA's first classified caffeine-based life-form
03:06.08keeblerHahah
03:16.52*** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net)
03:23.03*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
03:28.48*** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis)
03:37.48xacatecasok, not sure what i'm doing wrong, i've tried both http and tftp but I seem to be unable to push firmware 1.0.1.10 to my gs gxw4104.  it downloads the files but doesn't update.
03:38.13*** join/#asterisk kadath_ (n=kadath@rrcs-96-11-226-10.central.biz.rr.com)
03:38.38*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
03:39.56*** join/#asterisk LemensTS (n=customgt@adsl-70-238-154-243.dsl.stlsmo.sbcglobal.net)
03:40.57LemensTSAny AGI/php programmer that can figure out http://forums.digium.com/viewtopic.php?t=66943 ill paypal you $25
03:41.33*** join/#asterisk TrentCreek (i=kvirc@adsl-70-254-123-95.dsl.hrlntx.swbell.net)
03:41.57xacatecasnm, grandstream screwed up:  ERROR Failed to write firmware gxw4100b.bin to Flash: Data MD5 fail(4105)
03:42.08xacatecasor the ram on the device is screwed.
03:45.04[TK]D-FenderLemensTS: "Does not work with global variables. Does not work with some variables that are generated by modules."
03:47.03[TK]D-FenderLemensTS: This is the point where I might suspect that the var created here might be one of those
03:47.11*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
03:48.33*** join/#asterisk ScribbleJ (n=sj@c-67-172-6-141.hsd1.il.comcast.net)
03:51.12*** part/#asterisk Khratos (n=Khratos@190.166.113.91)
03:54.07*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
03:56.26LemensTSTK: Yea ive seen that a few times. Just odd that it will show the result but not save it to the variable. http://www.voip-info.org/wiki/view/Asterisk+AGI+php   wondering if those stdin scripts would be better
03:57.45[TK]D-FenderLemensTS: What ver of PHPAGI?
03:59.24LemensTSTK: Version 2
04:01.43[TK]D-FenderLemensTS: Do make sure to try the latest
04:07.04*** join/#asterisk rene- (n=rene@189.149.133.181)
04:07.28rene-hello
04:08.31*** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis)
04:08.47rene-when a pots line is shared between a DSL modem, asterisk via tdm voice card, and an external pots device such as a fax machine, where should adsl microfilters be placed?
04:10.39*** join/#asterisk dr0ck (n=dr0ck@nat/digium/x-0928e69be378ec1f)
04:11.02mostybetween everything except the adsl modem and the line
04:13.50[TK]D-Fenderrene-: Before everything except the modem of course
04:14.46rene-thanks, i had it all wrong
04:16.52LemensTSTK: i just downloaded the latest 2.14 and tried....same thing
04:17.48*** join/#asterisk joako (n=joako@99-153-162-33.lightspeed.miamfl.sbcglobal.net)
04:18.00joakoWhat is the root password for switchvox?
04:18.50*** join/#asterisk johnakabean (n=none@pool-72-82-111-56.nrflva.east.verizon.net)
04:19.01johnakabeanhow do you keep asterisk from starting automatically on boot?
04:19.35mostyjohnakabean, depends on your linux distribution, and how you installed asterisk
04:19.57johnakabeansource
04:20.04johnakabeani installed free though
04:20.06johnakabeanfreepbx
04:20.20johnakabeanso i use amportal to start and restart
04:20.38[TK]D-Fenderjohnakabean: Well I guess you don't tell amportal to start then
04:20.58johnakabeani have encrypted the /var/lib/mysql and /var/spool/asterisk directories and they cannot start until i enter the passphrase
04:21.13drmessanolol
04:21.32[TK]D-Fenderenters 1.....2....3....4...5...
04:21.33johnakabeanyou don't think you should encrypt call recordings, messano?
04:21.47johnakabeanand the mysql databases?
04:22.32drmessanoEncrypting directories on a PBX, that prohibit it from starting until a passphrase is entered?  Yeah.. Thats absolutely brilliant
04:22.54johnakabeani did this too late, messano, otherwise the whole partition would be encrypted
04:23.21drmessanoYou have nevergodownunlessiamthere set in grub.conf?
04:23.47johnakabeanit won't go down with 4 battery upss
04:23.54drmessanoLOL
04:23.54johnakabeanraid mirror
04:24.02drmessanoUPS's never go down
04:24.13drmessanokernels never panic
04:24.13joakoWhat is the root password for switchvox?
04:24.19johnakabeanif one goes down there are 3 more and a loud BEEP
04:24.44drmessano~glwt
04:24.44mostybad luck if the power goes out on the weekend, even with 4 ups'
04:24.52drmessano~gflwt
04:24.55drmessano:(
04:25.07johnakabeanmessano, you didn't answer the question you just asked why I encrypted it
04:25.15johnakabeanthat is irrelevant
04:25.35drmessanoand?
04:25.46johnakabeanand the question was how to keep amportal from starting on boot
04:25.53prg3Does anyone know of a good set of building blocks for a asterisk queue dialplan?  Specifically, I have external users calling into Zap lines to register their number as an agent, and then a set of zap lines take incoming callers, and forward them to registered agents on a zap trunk outbound?
04:26.00johnakabean.. /etc/local.d?
04:26.02drmessano[TK]D-Fender answered you
04:26.26johnakabeanasterisk starts on boot without me typing amportal start
04:26.50johnakabeanchkconfig safe_asterisk off doesn't work
04:27.38drmessanoDunno what to tell you.. when I set a freepbx box up, I purposely and knowingly set amportal to run at startup, which sets off the rest of the process.  No telling what the heck you've done.
04:27.50carrarjoako, You are not allowed to login to switchvox
04:28.02carrarvia ssh
04:28.07carraror console
04:28.13keeblerBesides TelephonyDepot.com Is there another recommended  site that sells both 802.11b/g SIP phones and ATAs? I see the list on voip-info.org , but any reviews/recommendations?
04:28.24johnakabeanthanks for the help /etc/rc.local
04:28.52drmessanojohnakabean: Whatever
04:29.21johnakabean"Dunno what to tell you.. when I set a freepbx box up, I purposely and knowingly set amportal to run at startup, which sets off the rest of the process.  No telling what the heck you've done." I think it automatically set it for it to run for you; that's why you had no clue what file to put it in.
04:29.24drmessanokeebler: I would avoid 802.11 SIP phones
04:29.36drmessanojohnakabean: Negative
04:29.39*** part/#asterisk johnakabean (n=none@pool-72-82-111-56.nrflva.east.verizon.net)
04:29.47keeblerdrmessano: I have a client that wants to walk around the site.
04:29.51carrarkeebler: http://tinyurl.com/bxcywc
04:29.59drmessanoBattery lufe is going to be horrible at best
04:30.08*** join/#asterisk highlevelcode (n=highleve@adsl-75-51-95-24.dsl.lsan03.sbcglobal.net)
04:30.13*** part/#asterisk highlevelcode (n=highleve@adsl-75-51-95-24.dsl.lsan03.sbcglobal.net)
04:30.23keeblercarrar: I was hoping for REVIEWS of who to avoid.
04:30.23drmessanolife*
04:30.29keebleror who to recommend.
04:30.30carrarheh
04:30.33carrarI couldn't resist
04:30.48keeblerLike, VOIPSupply looks promising.
04:30.52drmessanoEWWW
04:31.04drmessanotelephonydepot is pretty much the best option
04:31.17keeblerdrmessano: Yeah, i bought some ATAs from them this week.
04:31.33keeblerThey arrived yesterday.
04:31.34keebler:)
04:31.36carrardepends what you are buying I suppose
04:31.38carrarebay works great
04:31.45keeblerNeeds to be new. :/
04:31.58carrarYou can buy new on ebay
04:32.00keeblerAnd I was hoping to get more ATAs and Phones from the same provider.
04:32.06drmessanoewwww
04:32.20keeblererm.. distributor
04:32.24drmessanoebay auctions end up OVERpriced
04:32.46keeblerHaven't really used ebay in over a year.
04:33.00keeblerAnd I need guaranteed 2nd day.
04:33.01drmessanokeebler: So whats the problem?  You bought from telephonydepot, got the stuff.. buy more from them..
04:33.32carrarhttp://www.voip-info.org/wiki/view/Asterisk+system+vendors
04:33.37keeblerdrmessano: Just trying to avoid multiple purchases from different vendors. is all. But I'll probably just end up doing it anyway.
04:33.43keeblercarrar: .....
04:33.58keebler22:28:16 < keebler> Besides TelephonyDepot.com Is there another recommended  site that sells both 802.11b/g SIP phones and ATAs? I see the list on voip-info.org , but any reviews/recommendations?
04:34.09drmessanoDo they not sell what you need?
04:34.14keeblerdrmessano: Nope
04:34.22drmessanoWhat is it you need?
04:34.26keeblerdrmessano: Their IP phone selection is limited.
04:34.39carrardrmessano, need to repast what already been asked :)
04:34.44carrarrepaste
04:34.59keeblerDecent/stylish 802.11bg SIP phone. :( The client is vain and cares about crap like that.
04:35.00drmessanoTheir IP phone selection seems to include the most reasonable items
04:35.08joakocarrar: So I installed Switchvox, the Zaptel interface does not work. I just give up with Switchvox then?
04:35.08drmessanokeebler: There is no such thing
04:35.21*** join/#asterisk adnc (n=adnc@unaffiliated/adnc)
04:35.22carrarjoako, call their free support
04:35.23drmessanokeebler: 802.11 phones suck, are ugly, and not worth it
04:35.35joakocarrar: There is free support for the free version?
04:35.40*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
04:35.45drmessanokeebler: Hence the availability from someone selling products they expect to make money on
04:36.09keeblerdrmessano: Ah. I might be able to get away with this. http://www.voipsupply.com/cp-7921g-a-k9 ...
04:36.11carrarjoako, call them and ask if Zaptel is part of the free version
04:36.18carrarI kinda doubt it
04:36.26keeblerAlthough, A lot of my clients have iPhones and Blackberry Bolds.
04:36.42drmessanoWell, good luck with it
04:36.50carrarSwitchVox has great phone support
04:36.52joakocarrar: It is... site says free version allows analog cards
04:37.10joakoBut I don't expect them to provide free support for the free version.....
04:37.16keeblerdrmessano: I think I might just setup a cell number with a VOIP Service provider and have them call the exstention and use their cell phone.
04:37.19carrartry
04:37.25keeblerextension
04:37.30carrarthen join the switchvox channel and ask
04:37.39keeblerdrmessano: That sound reeasonable?
04:37.42keeblerreasonable/
04:37.43keebler?
04:37.53adncis there a way to map voipuser@myhost.mydomain.tld to voipuser@mydomain.tld if yes how can this be done?
04:38.12drmessanoIm not sure what you mean.. "setup a cell number with a VOIP Service provider"
04:38.29keeblerdrmessano: I mean, 3G internet access.
04:38.42keeblerdrmessano: Then get a DID from someone like vonage or whatnot.
04:38.44joakoWell then obviously SwitchVox is crap... they can't even make a CD. I got the disc and it had the ISO on it!
04:38.56carrarSwitchVox is nice
04:39.03carrarI use it for specific customers
04:39.05carrarand works great
04:39.15carrar4.0 will even be better
04:39.19[TK]D-Fendercarrar>SwitchVox has great phone support <- which he'd use.... if his phones were working :p
04:39.24carrarhaah
04:39.27carraryeah
04:39.31[TK]D-Fendercarrar: ... or wouldn't need :)
04:39.31carrarcatch 22
04:39.35drmessano"Set up a SIP client on their 3G phone as an extension on the PBX"  <--- Yes
04:39.43[TK]D-Fenderpasses the ball back for another inning
04:39.49sipyhaha: joaka, did you make an iso of that cd with an iso on it?
04:39.59keeblerdrmessano: I provide satellite internet and VOIP phones on the rig right now. But, the clients refuse to use the Sat phones because of the 2 second delay.
04:40.12carrarjoako, I found the Zaptel config 100% self intuitive
04:40.21carrarerr NOT 100% self intuitive
04:40.26carrarso call them
04:40.36carraror read the admin guide
04:40.45[TK]D-FenderYou sold your soul... accept the payment!
04:41.29adncmhh
04:41.45carrarIf you are nice on the phone they will even login and do it for you maybe
04:41.53carrarthey are that awesome
04:41.54*** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-e8c4e185dbf64dd5)
04:42.32drmessano[TK]D-Fender: johnakabean says "Hi"
04:42.41*** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net)
04:42.41drmessano[TK]D-Fender: I think he's still bitter
04:42.59carrarjoako, join #switchvox
04:43.01carrar*** #switchvox           10
04:43.14[TK]D-Fenderdrmessano: Gotta love that johnakabean.... renames root, is an iptables MASTAH, is runni g**SOFTWARE** RAID (MIRROR only, not even Raid 5, no spares....), and can't figure the &^#%$ING STARTUP for a damn service :p
04:43.19*** join/#asterisk eppigy (n=Dave@plasticlobster.com)
04:43.21eppigyhello
04:43.23eppigyi am dave
04:43.27[TK]D-Fendereppigy: you are dave
04:43.31eppigyyes
04:43.33carrarHi dave
04:43.34joakocarrar: Noone there. Yes, the webinterface is very nice... but I just boot into single-user mode and see that /etc/zaptel.conf is WRONG
04:43.35eppigyhi
04:43.36[TK]D-Fendereppigy: I've channeled you in your absence :)
04:43.40eppigyoh good
04:43.58eppigysomeone needs to carry the torch
04:44.05[TK]D-FenderBURNS people with it
04:44.10adncmy subhost with asterisk is somewhere else located than my domain pointing to. is it still possible to redirect them?
04:44.26drmessano[TK]D-Fender: Yeah, I love how he gets pissed off when I wont help him.. and then tries his little digs "well, sounds like you just didnt know..."
04:44.27[TK]D-Fenderadnc: ... pardon?
04:44.36drmessanoNo, sounds like I want you to die in a fire
04:44.40drmessanoahem..
04:44.43[TK]D-Fenderdrmessano: I bet YOU don't know...
04:44.50[TK]D-Fenderwaits for drmessano to take the bait.
04:44.54carrarjoako, if you do stuff not using the interface you will only screw yourself over
04:45.14[TK]D-Fenderdrmessano: Seriously... thats &#^$ing kindergarten tactics
04:45.14drmessanoZOMGWTFBBQ FSCK YOU R U SERISU GRRRRR BZZZZZZZZ AACCCCCKKKKKK FIGHT
04:45.20drmessanoIndeed
04:45.35eppigyRAGE FILLIGN EVERY CAPILLARY
04:45.41carrarw00t!
04:45.42drmessanoI mean
04:45.43Gopher_77lol
04:45.44[TK]D-Fenderdrmessano: He's clearly advancing in his social strategies ;)
04:45.57carrarDr. Phil can help you with RAGE ISSUES
04:46.04drmessanoMy BP is 160/90 right now.. Does he think he can raise it?  Doubtful.. IT CANT GO ANY HIGHER
04:46.15[TK]D-Fendercarrar: <drphil. How's that working out for you?
04:46.15adnc[TK]D-Fender: my subdomain.domain.tld is pointing to my dynamic ip where my asterisk is behind, but i would like to use names@domain.tld  is there a way to redirect if there is a sip request
04:46.30carrarTK, I'm drinking, it's working out great
04:46.44[TK]D-Fenderadnc: * is not a sip proxy and does not care about "domains".
04:46.54carrarDrinking seems to be the best way to really enjoy this channel
04:46.59[TK]D-Fenderadnc: either the packets make it there, or they don't
04:47.15[TK]D-Fendercarrar: Salvia <-
04:47.28frogonwheelsadnc: SRV records could be what you're after.
04:47.31carrarheh
04:47.33eppigycarrar: try smoking purp by the pound
04:47.41carrarnaw I don't smoke
04:47.50eppigynigga you aint real
04:47.52adncfrogonwheels: srv records? i'll search for it, thank you
04:47.59[TK]D-Fendercarrar: http://tinyurl.com/cv8sky
04:47.59joakoeppigy: Agreed
04:48.00prg3I'm having CID problems with my Zap lines.. basically, I don't get any CID data when calling in.  http://pastebin.com/m63a228d4 has a bunch of details
04:48.00carrar1 bad habbit is enough
04:48.13frogonwheelsadnc  http://www.voip-info.org/wiki-DNS+SRV
04:48.16carrarhahah
04:48.24drmessanocarrar: People come in here SOBER?
04:48.25adncfrogonwheels: thank you
04:48.31eppigyi pretty much like to get really chill
04:48.32drmessanocarrar: U R SHITEN ME
04:48.41eppigyand choke the life out of something
04:48.44carrarTK, http://www.osburn.com/E14T6863.jpg
04:48.47frogonwheelsadnc: of course, you have to have a good domain host to let you set them.. but that's gonna be another story.
04:49.17drmessanocarrar: I bet YOU didnt know the answer to my question, hence you not answering.  Thought so.  Told you.
04:49.19adncfrogonwheels: i use joker.com which was ok till now. lets see hopefully it can handle this aswell
04:49.23[TK]D-Fendercarrar: Lots'o'toys
04:49.29carrardrmessano, I missed the question
04:49.31drmessanocarrar: I win.  You lose.  i am rubber.  You are glue.
04:49.39drmessanocarrar: Thats not important now.
04:49.40*** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de)
04:49.48carrarok
04:49.57drmessanocarrar: We're past the question, and to childish antics and finger pointing
04:49.59carraroh
04:50.02carrarI see the Q
04:50.04carrarI hope not
04:50.06frogonwheelsadnc: I haven't quite got round to doing it myself - on my todo list.
04:50.07drmessanocarrar: The "question" is so 3 mins ago
04:50.12carrarheh
04:50.13carrar@#($*&!@)
04:51.00drmessano()%$@*%* you too, and, I still think you didnt know, ZOMG, WTF, I have more hitpoints than you.
04:51.17carrarthats MAD regex
04:51.27adncfrogonwheels: yes, thank you, thats what i was looking for
04:51.32adnc[TK]D-Fender: thank you to you aswell
04:51.56drmessanocarrar: Then theres the ultimate troll question.. "What level dungeonmaster are you?"
04:52.00*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
04:52.05carrarheh
04:52.09eppigythis is intense
04:52.14drmessanoDUNGEONMASTERS DONT HAVE LEVELS, DUH
04:52.23drmessano^^^^ I win, the interweb
04:52.32carrarIRC is the only game I have time for
04:52.37drmessanoBut life, lose miserably
04:52.52ruben23hi...i have this on my asterisk CLI>>Feb  5 12:40:51 NOTICE[16149]: chan_sip.c:10057 handle_response_peerpoke: Peer 'Prime' is now TOO LAGGED! (1262ms / 1000ms)
04:53.01drmessanocarrar: Kissing real women is so overrated.  Gaming 4eva
04:53.08carrarheh
04:53.58drmessanocarrar: Seen a real naked woman?  No.  CGI'ed one with HUUUUUGE amounts of gold to help my clan in WoW?  HELL YESH
04:54.26carrarI found one crazy enough to marry me
04:54.38keeblerHas anyone used one of these? Cisco WBP54G Wireless-G Bridge
04:54.53keeblerhttp://www.telephonydepot.com/Catalog/Linksys-Phones/Linksys-WBP54G-Wireless-Bridge-for-Phone-Adapters
04:55.13carrarLinksys
04:55.14carrarheh
04:55.16drmessanoI found two.. the first one will hopefully keep taking her meds and marry someone born with chainmail covering every inch of their body.  The second one, well, I need two more signatures for her.
04:55.18ruben23after that i have this Feb  5 12:43:04 NOTICE[16443]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'sip' (cause 3 - No route to destination)
04:55.42mostyruben23, because the sip peer is too lagged
04:56.08Gopher_77I uninstalled and re-installed my dahdi to solve my dial tone problem, and now something's seriously wrong with dahdi; here's some of what I see: http://nopaste.com/p/aMnwkcbIl Can someone help me get this working again?
04:56.34prg3ruben23: can you ping the phone? (or device..)
04:56.54adncdoes a user agent always do a SRV lookup when placing a call?
04:57.04ruben23mosty:my asterisk server is behind NAT thus this affects the channel line..?
04:57.11mostyadnc, depends on the user agent i suppose
04:57.39adncmosty: i suppose if not then it is doing normal dns lookup, right?
04:57.40carrarSRV needs work in Asterisk
04:57.49drmessanosrvlookup=yes
04:57.58drmessanofor Asterisk anyway
04:58.07drmessanoDevices should have their own option
04:58.22drmessanouse_dns_srv for linksys devices
04:58.25mostyadnc, sip clients can do DNS lookups however they want
04:58.29adncbut my asterisk is behind and my server, so the user agents have todo this srv lookup, which would be ok even if asterisk needed some work with srv
04:58.39carrarruben23: read http://svn.digium.com/svn/dahdi/linux/tags/2.0.0/UPGRADE.txt
04:58.49carrarcheck out some of those tools
04:59.43ruben23<PROTECTED>
04:59.48carraryes I know
05:00.01carraroh
05:00.08carrardid I use the wrong nic?
05:00.09ruben23carrar:those tool can help...?
05:00.15prg3Zaptel not picking up Callerid, anyone help? http://pastebin.com/m63a228d4
05:00.19carraroh
05:00.21carrarthat was for Gopher_77
05:00.22carrar:)
05:00.30carrarI blame drmessano
05:00.39Gopher_77carrar: what was for me?
05:00.41carraryeah
05:00.42drmessanoI BLAME BURMA
05:00.48drmessanooh nm
05:01.02carrarthere are some tools you can use to view more details of your T1/PRI
05:01.18drmessanoYou find most of the tools on IRC
05:01.33ruben23prg3:my asterisk server is behind NAT thus this affects the channel line..?
05:01.35carrartype /sign <your sign here>
05:01.37Gopher_77carrar: I only have a 4 port card
05:01.51mostyruben23, http://www.aocomputing.net/?p=3
05:02.11[TK]D-Fenderis burning CentOS 5.2 ISO's for tomorrow....
05:02.12carrarI have a few dozen of digium 4 ports also
05:02.19carrarw00t
05:02.20prg3ruben23: nat does bad things in general.. no route to host indicates that one machine or the other doesn't know how to talk to the other
05:02.25carrarwelcome to almost yesterday TK!!
05:02.29drmessanolol
05:02.31carrar5.3 will be out sooN!
05:02.43[TK]D-Fendercarrar: I know...
05:02.57[TK]D-Fendercarrar: but this is for a pretty boring work server.. SAMBA only really.
05:03.00drmessanocarrar any day^h^h^h^h^hweek now!
05:03.01carrarah
05:03.15carrarI do all my 5.2 installs over PXE boot NFS installs
05:03.21[TK]D-Fendercarrar: I want the Novell on it TRASHED so I can panic head-office :)
05:03.21carrarfast!
05:03.32drmessanoLOL
05:03.40drmessanoTrashing Novell is fun
05:03.44carrarheh
05:03.51[TK]D-Fenderdrmessano: The look on their faces.... PRICELES
05:03.59[TK]D-Fenderdrmessano: The look on their faces.... PRICELESS
05:04.07[TK]D-FenderWorthy of full correction
05:04.09prg3Just run Windows if you want to run windows.. SuSE is a bad windows implementation..
05:04.15carrarNovel 10base2
05:04.21carrarLL
05:05.41ruben23mosty:yes i already read that when i set it my sip.conf....it worsen the case....no chennel available always..
05:06.23carrarruben23, befor dialing a SIP device do a ChanIsAvail beforing dialing the SIP device
05:06.35carrarcheck to see if it's there before calling it
05:06.52carrarwon't solve your issues of it not being there however
05:07.26mostycan you ping between the two machines?
05:07.58Gopher_77I ahven't heard about 10base2 in a while
05:08.21carrarI admin a Novell 4.11 server once
05:08.35carrarthat was fun for about a week
05:08.38adncdoes someone know of a comparison of sip phones anywhere?
05:08.44prg3Gopher_77: have you ever seen 10Base5?
05:08.52Gopher_77prg3: no
05:08.57carraradnc, use polycom
05:09.08prg310base2's big ugly brother
05:09.12carrarPolycome, Aastra, Cisco or Linksys
05:09.22carrar-e
05:09.46adncdo the linksys ones work ok with asterisk? do they also do iax2?
05:09.48prg3Gopher_77: to connect to it, a pin punches a hole in the cable to make a connection
05:09.50frogonwheels(carrar, next time try  s/e//   :) )
05:09.52carrarSIP
05:09.59Gopher_77prg3: vampire connection?
05:10.03prg3Yup
05:10.05carrars/Polycome/Polycom/
05:10.10mostyadnc, no phones do iax2 (none that are worth using, anyway)
05:10.12carrar:)
05:10.12frogonwheelstoo late :)
05:10.14carrarheh
05:10.17carrarPolycome, Aastra, Cisco or Linksys
05:10.18carrars/Polycome/Polycom/
05:10.24carrarheh
05:10.25mostysnom are ok too
05:10.36carrarif you say so
05:10.40carrarI don't like em
05:11.01frogonwheelssomebody was talking about Siemens Gigaset ?? Anybody got an opinion?
05:11.26mostycarrar: they're easy to use and administer, and have good BLF functionality with asterisk
05:11.38mostypity about the lack of backlit lcd
05:11.53carrarhttp://pics.osburn.com/photo/32428/original
05:11.58carrarthere is a snom in that pic
05:12.56adnccarrar: can you recommend a particular one from polycome or linksys?
05:13.03carrar430,601
05:13.11carrar(Polycom)
05:13.20carraror their newer phones
05:13.29carrarlinksys 942 are decent
05:13.48carrarPolycom is the best I think
05:13.52adncthank you
05:14.02carrarCisco 79XX has the best feeling in the hand phone
05:14.11mostypolycom are almost unusable without a provisioning server
05:14.12carrarnot so in the config though
05:14.14mostynice hardware though
05:14.28Gopher_77the human network?
05:14.39carrarI wrote perl scripts to general all my ftp and tftp config for my phones
05:14.44adncmosty: the cisco or polycom?
05:14.45carrargenerate
05:15.02carrarand Aastra are ok
05:15.03mostyadnc, polycom
05:15.13carrarPolycom would the best I think
05:16.01carrarPOlycom has some nice HD Audio phones too
05:16.05carrarg.722
05:17.02carrarIhave 7941/61 and 7970's working great with asterisk too
05:17.06carrar(cisco)
05:17.33carrarthey are really "show" phones more then anything else
05:17.49*** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178)
05:19.57*** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com)
05:21.27carrarI'm pretty sure if the FBI raided you they would take your lousy quit message also
05:21.53*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
05:25.07adncwhat about the 7911 from cisco, does it work with asterisk?
05:25.32carrarI have not used those
05:26.00carrarif they are anything like the 7905, (they kinda look like it) you have to compile the config PITFA
05:26.30adnci just see that these phones are available with license and without, what sort of license are they talking about
05:26.36carrarBut Asterisk can support SCCP
05:27.04carrarI like to install the Cisco SIP Code on the their phones and not use the SCCP protocol
05:28.49carrarJust call them and ask what firmware is on the phone
05:29.05mostyadnc, i would never recommend cisco phones unless you want a pure cisco setup
05:29.12*** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net)
05:29.18carraras a basic phone they work great
05:29.23carrarfeel nice in the hand
05:29.26carrarand look cool
05:29.44mostythe linksys phones look like cisco phones these days, and should be cheaper and easier to setup
05:29.47carrarI use Cisco as my main phone at home
05:30.14carrarThe linkSys have more configurability then the Cisco 7900 but also feel "cheaper"
05:30.27[TK]D-FenderPolycom > ALL
05:30.29carraryes
05:30.32adncmosty: what is the most compatible to asterisk
05:30.40carrarthey all do SIP
05:30.48carrarAKA compatible
05:31.07carrarJust what do you want to do beyond SIP is what you are looking for
05:31.13frogonwheels..still for example - I want to make sure it supports  answer-after=0 or similar for Paging
05:31.22carrarPolycom
05:31.33frogonwheelstoo expensive :)
05:31.33mostyadnc: snom probably has the best asterisk integration, polycom and linksys would be next, i haven't tried much aastra gear but that's supposed to be good
05:31.34carrarThey all do that also
05:31.42carrarLInksys does auto answer nicely
05:31.48carrarcontroled from Asterisk
05:31.58carrarCisco only does it 'per phone'
05:32.13carrarall or nothing per registration
05:32.30carrarYou could assign a line on a cisco to auto answer
05:33.00adncahh
05:33.23carrarCisco has some cool XML services thats easy to write code for
05:33.30carrarLinksys doesn't really
05:33.45carrarhaven't tried with polycom
05:34.38carrarYou can write a perl script to send XML to Cisco that polls your exchange server or any server to get a live directory
05:34.44carraror to get the current weather
05:34.46carrarwhatever
05:35.02carrarinteractive menus
05:35.27adnccarrar: via agi
05:35.45carrarvia XML
05:36.05frogonwheelscarrar: presumably you just send the xml straight to the phone..
05:36.16frogonwheelscarrar: asterisk (presumably) isn't involved.
05:36.23carrarthe phone needs to do the inital poll
05:36.33frogonwheelsermm.. adnc ^^ (sorry carrar)
05:36.40adncahh, i see
05:36.44carrarto any xml server (apache)
05:36.50frogonwheelsahh..
05:36.52frogonwheels:)
05:36.55adncso pressing one button on the phone will display a directory
05:37.03carrarcan
05:37.14adncsounds nice
05:37.15carrarI have it display a "main" menu
05:37.22carrardirecotry, weather, stocks
05:37.28adnccool
05:37.31carrarphear my spelling
05:37.40adnccarrar: can linksys do this aswell?
05:37.41*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
05:37.52carrarand I use PHP to do the XML and db queries
05:38.01carrarno
05:38.11adnconly cisco
05:38.11carrarleast not the SPA-94X
05:38.23adncif i'm not wrong then linksys is part of cisco
05:38.34carrarand I have not researched Polycom to see if they suppose XML services
05:38.38carrarsupport
05:40.48carrarPolycom 600/601/650 support XML Browser
05:40.55carrarso probably yes
05:41.17carrarerr XHTML
05:41.35carrarjust need to hack it out
05:42.14*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
05:42.49adnccarrar: this one aswell http://www.sipgate.de/voipshop/polycom/soundpoint_ip320
05:43.27carrarI lived in Berlin for a year
05:43.42adnccool
05:43.48Gopher_77carrar: I see that lsdahdi is broken, but dahdi_scan seems to work fine
05:43.49carrarOnly German I know I forgot
05:43.52carrarheh
05:43.56SwK_the 500's support the xhtml browser too
05:44.08carrarI haven't used 320
05:44.19carrarused a lot of their 4000'
05:44.21carrars
05:44.41carrarI would think they work fine
05:45.13carrarPolycom just seems to work nicely
05:45.25carrarreading the admin guide is required however
05:45.49carrarLots of details in their firmware, app code etc...
05:45.57carrarthat you need to know
05:47.01*** join/#asterisk n_akram (n=nakram1@119.152.14.135)
05:47.01*** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com)
05:47.12n_akramHello room
05:47.41n_akramI am confused by the sheer difference between SIPPeers and IAXPeers AMI command
05:48.05carrarOne uses SIP protocol and one uses IAX protocol?
05:48.14carraroh AMI
05:48.16n_akramin response to SIPPeers, Asterisk will send me Events
05:48.38n_akrambut for IAXPeers, * sends list in a tabular format
05:49.52n_akramthis is not good...
05:50.21n_akramIt means that I'll be adding new logic to process IAXPeers... that sounds real bad...
05:51.34[TK]D-Fendern_akram: You've got the source like the rest of us.... Feel free to write a patch
05:53.01n_akramhmmm... well... I will do it... @[TK]D-Fender
05:53.12carrarNoting
05:53.13n_akrambut not right now, its a good idea... i will do it myself...
05:53.54carrarI'm gonna re-write asterisk
05:53.58carrarjust not right now
05:54.19carrar:)
05:55.11carrarcrazy people on cnn
05:56.19[TK]D-FenderNow via hologram!
05:56.25carrarhaha
05:56.25n_akrambuddy i will do it... but right now my boss is on my back... just yesterday he called me in and said "blah blah blah blah" :p
05:56.29carrarthat was so lame
05:56.48Gopher_77carrar: interesting, now lsdahdi is working
05:57.05[TK]D-Fendercarrar: n_akram OMG, your boss ounds like that too?
05:57.14carrarheh
05:57.27carrarAll bosses are the same at the end of the day
05:57.32Gopher_77I think most bosses sound like that
05:57.45Gopher_77either that or wok wok wok wok wok
05:57.52*** join/#asterisk frk2 (n=frk2@zivios/member/fkhan)
05:57.54Gopher_77but mostly that's teachers
05:57.54carrarSelf employement
05:57.55[TK]D-Fendermine actually supports most of what I want to do.....
05:58.11[TK]D-Fenderis happy in his long-term employment stability.
05:58.50frogonwheels[TK]D-Fender: .. so tell us the truth, just what _do_ you have over him :)  :)
05:59.09carrarPasswords!
05:59.09[TK]D-Fenderfrogonwheels: 10 years seniority :)
05:59.14[TK]D-Fendercarrar: and that :)
05:59.17carrarheh
05:59.31frogonwheels[TK]D-Fender:  yep- that'd do it..
05:59.36n_akram[TK]D-Fender: so you've been working for 10 years
05:59.39carrarAll the design/architecture in your head
05:59.50[TK]D-FenderTo be he'll always be boss #6 :)
06:00.11[TK]D-Fendern_akram: No, ... 10 more than HIM :p
06:00.24[TK]D-Fender(13.5 FYI)
06:01.00carrarMost Bosses just want to know they can spend 3 weeks in Hawaii and not be bothered and that you can handle everything
06:01.07n_akram[TK]D-Fender: wow... that's cool... so that's why you know all the asnwers...
06:02.40carrarTK is our online voip-info.org resource
06:02.50carrar(thats up todate)
06:02.58carrarheh
06:03.31carrar++
06:03.36eppigyi get crunk at work all day
06:03.50eppigystr8 wilin
06:03.51carrar~crunk
06:04.16carrarCrunk is a type of music, which originated from southern hip hop and EDM in the early 1990's.
06:04.41[TK]D-Fender~[TK]D-Fender
06:04.42jbot[TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy.
06:04.48carrarheh
06:04.50carrarhahah
06:05.09carrarTK, can you google something form e
06:05.11carrarme
06:05.16eppigy~consensual sex
06:05.25Gopher_77~sex
06:05.26jbotupdatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/
06:05.49eppigyinteresting
06:05.56carrarhttp://sqzm14.ust.hk/sexman.html
06:06.01Gopher_77bots are great
06:06.04Gopher_77~pirate
06:06.05jbotSo forgive me my rough mood unaccustomed to sue; I woo not, perhaps, as your landlubbers do.
06:06.28eppigyTAKE IT
06:06.55Gopher_77~fag
06:06.56jbotfag is, like, a impolite way of refering to bkw
06:06.58[TK]D-FenderNO WE AIN'T GONNA TAKE IT!
06:07.05[TK]D-FenderWE'RE NOT GONNA TAKE IT... ANYMORE!!!!!!!!!!
06:07.07Gopher_77~bkw
06:07.08jbotbkw is probably wants to eat file's muffin
06:07.08eppigyIRON EAGLE
06:07.27eppigy~eppigy
06:07.31[TK]D-FendermAN... THAT IS HARSH...
06:07.36eppigyW
06:07.38eppigyO
06:07.38Gopher_77~silence
06:07.39eppigyW
06:07.47eppigy~dave
06:07.48jbothmm... dave is the renowned inventor of the FR-clapper
06:07.55eppigyFACT
06:07.58Gopher_77~muffin
06:07.59jbotrumour has it, muffin is http://leifmadsen.com/images/muffin.jpg, or a filtering proxy server for the World Wide Web written entirely in Java. URL: http://muffin.doit.org/
06:08.52Gopher_77~dave
06:08.53jbotsomebody said dave was the renowned inventor of the FR-clapper
06:08.53carrarthat url doesn't work
06:09.00*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
06:09.20Gopher_77~FR-clapper
06:09.31eppigy4rl
06:09.35eppigywhat in the hell is that
06:09.39eppigyand when did i inventit
06:09.40*** join/#asterisk contrabanda (n=contr@92.241.69.134)
06:10.09contrabandahi all
06:10.19Gopher_77must have been last weekend
06:10.34*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)
06:10.35eppigyi guess so
06:10.39contrabandaGopher_77: Hello man
06:10.48Gopher_77~hello
06:10.48jbotHowdy Bub
06:10.55eppigyALLO
06:11.02joelsolankihello all
06:11.10Gopher_77~howdy
06:11.11jbothowdy right back at ya'!
06:11.12joelsolankiin this line i want to trim first 3 digits how can i do that ?
06:11.13joelsolankiexten => 311,n,SayDigits,${CALLERID(num)}
06:11.32eppigy:3
06:11.38joelsolankiwhere exactly ?
06:11.47[TK]D-Fenderexten => 311,n,SayDigits(${CALLERID(num)}:3)
06:11.48eppigy${CALLERID(num):3}
06:11.49carrar${CALLERID(num):3}
06:11.57[TK]D-Fenderexten => 311,n,SayDigits(${CALLERID(num):3})
06:11.59carrargotta be right
06:12.03eppigydang dog
06:12.15carrarOH
06:12.18carrar${CALLERID(number):3}
06:12.21carrarHAH
06:12.23[TK]D-Fenderjoelsolanki: No go read CHANNELVARIABLES.TXT  this is 101-grade material
06:12.26[TK]D-Fendernow*
06:12.31carrarheh
06:12.33eppigy4rl
06:12.41eppigytake ur tired ass home
06:12.43eppigywith that
06:12.46Gopher_77~channelvariables.txt
06:12.52Gopher_77~variable
06:12.56[TK]D-Fenderok, time for my 7-8 minutes of sleep
06:13.08eppigy~transcendtal numbers
06:13.16eppigy~transcendental numbers
06:13.19joelsolankilet me try
06:13.23Gopher_77~joel
06:13.29joelsolankiyes plz
06:13.35Gopher_77~contraband
06:13.37joelsolankiyes
06:13.38eppigy~let me have a wack at it
06:13.43Gopher_77~whack
06:14.01Gopher_77~transcendental
06:14.08Gopher_77~windows
06:14.09jbot[windows] either a 64bit hack on the 32 bit hack or just a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs, or bloatware, or an operating system
06:14.33Gopher_77lol
06:14.47joelsolankiyes that worked. thanks
06:14.59eppigymna
06:15.03eppigythat is pretty funny
06:15.19frogonwheelsyup. old. but funny.
06:15.24eppigyur old
06:15.44Gopher_77I've used windows mostly as a world of warcraft bootloader for the past couple years :)
06:15.55eppigywow is the devil
06:16.36Gopher_77he keeps the evil empire running
06:18.49Gopher_77so now it seems my main problem is no dial tone, which is minor
06:25.31*** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
06:29.22*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f79ecf52270c1b2d)
06:31.34drmessanojbot: windows is also an OS that people actually USE
06:31.35jbotACTION installs Windows ME on is also an OS that people actually USE's PC.
06:31.43drmessanoROFL
06:31.53*** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net)
06:31.56Gopher_77is there some sort of trick to getting * to act as a SIP server?
06:32.17Gopher_77I mean, so that I can use a SIP softphone with it
06:32.19frogonwheels~windows
06:32.20jbotwell, windows is either a 64bit hack on the 32 bit hack or just a 32 bit hack on a 16 bit operating system, originally designed for an 8 bit CPU, with a 4 bit system bus, made by a 2 bit company that can't stand 1 bit of competition... or the World of Warcraft bootloader, or the most important collection of bugs, or bloatware, or an operating system
06:32.20ScribbleJInstall it, mostly.
06:32.31ScribbleJNot sure how 'tricky' that is.
06:32.53drmessanoGopher_77: Asterisk _IS_ a SIP server
06:33.16drmessanoIt doesnt need to act as one, no more than apache needs to "act" as an HTTPd
06:33.18Gopher_77drmessano: yes, but I don't know how to set it up
06:33.46frogonwheelsGopher_77: it aint that hard.  read  sip.conf
06:33.54drmessanoTheres no trick other than "doing it"
06:34.57frogonwheelsGopher_77: there're a few tricks to getting certain SIP clients to talk to it.  but you don't sound like you've got that far.
06:35.44sipyit allways makes agents acknoledge
06:35.50Gopher_77frogonwheels: I've used X-lite with my SIP provider, but it doesn't get a response from my setup of *
06:36.13sipyeven with ackcall=no
06:36.20sipywtf
06:37.36Gopher_77frogonwheels: ok I found a sample configuration to communicate with X-lite
06:37.53frogonwheelssometimes it's tricky things on the client too - not all asterisk :)
06:39.11*** join/#asterisk fiddur (i=fiddur@c042.rit.se)
06:39.55carrarGopher_77, xlite config for * should be pretty basic
06:42.12harry_vx-lite is only as good as your microphone
06:42.22harry_vother then that its great
06:42.48harry_vcarrer it is basic
06:43.05*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
06:44.01carrarUSB headset or bust!
06:45.10Gopher_77carrar: yeah, it was easy to set up for my provider
06:46.16*** join/#asterisk denon (i=denon@synapse.subneural.net)
06:46.16*** mode/#asterisk [+o denon] by ChanServ
06:47.27Gopher_77registration error 408 - timeout
06:47.41Gopher_77I think there was a setting for incoming tcp somewhere?
06:54.12carrartcp eh?
06:55.04drmessano....
06:55.17*** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net)
06:56.56drmessanoSo let me get this right.. All the pics I get emailed of Obama are photoshops of his head on thugs bodies, but they're not meant to be racist?
06:57.12drmessanoOk, good.. I was beginning to wonder
06:57.54drmessanolooks for a belltower to express his disgust of racist a**holes
06:58.10harry_vnight
07:01.37TrentCreekwhat "race" is Obama?
07:01.46*** join/#asterisk sergee (n=serg@voip1.west-call.com)
07:02.09*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
07:02.50drmessanoAccording to 99% of the emails I get, hes black.. and a muslim.. and plays a lot of basketball and rap music.
07:03.05jameswftrixbox down to just andrew?
07:03.11drmessanoyeah lol
07:03.28drmessanoAndrew and andrew
07:03.36GameGamer43and drew
07:03.44jameswfcan i get a side of andrew with that
07:03.46TrentCreekAccording to Miriam Webster he is "mulatto"
07:04.17ScribbleJ"Race" doesn't seem to be a distinction with any meaning these days.
07:05.23TrentCreekBecause now day everything you do is "racist" when if affects (so they claim) certain ethic groups
07:05.44ScribbleJI don't think that's the only reason really.
07:05.53TrentCreekso it is over used and has no meaning , and the word is no longer being used in its proper meaning
07:05.54jameswfI hate purple people personaly....
07:06.28TrentCreekRacism is defined as the belief your own race is superior to others
07:06.29ScribbleJdown with greenies!
07:06.48TrentCreekHang Jolly Green Giant!
07:06.59ScribbleJOh wait, no, he's cool.
07:07.05ScribbleJMaybe down with some greenies, then.
07:07.29TrentCreekAnd the Green Lantern!
07:07.37TrentCreekand Green Hornet! Get them all!
07:08.51ScribbleJNow that you mention them, I guess greenies ain't so bad.
07:08.55ScribbleJBut screw plaid people.
07:08.59TrentCreekEspecially those GReen whores from Orion!
07:09.02jameswftrixbox is green
07:09.28jameswfthey see me trollin.... they hatin
07:09.35TrentCreekYeah! We got one plaid guy! No more Grunge Music!
07:09.46ScribbleJhah
07:10.07drmessano<TrentCreek> Racism is defined as the belief your own race is superior to others  <-- Which says nothing about my comments.. If youre attempting to counter, then try harder
07:10.15TrentCreekremeber: Dead men don't wear plaid
07:10.35TrentCreekdrmessano: no need
07:11.04ScribbleJdrmessano, people just react to change, I think.  They are still amused by the novelty of "a black president!"  How quickly we all forget Clinton.
07:11.22ScribbleJBut seriously, I think folks'll generally calm down and forget about it after a while.
07:11.39TrentCreekScribbleJ: Well, I will be amused when we actually get a real black preisdent
07:11.48drmessanoScribbleJ: I live in the south.. its more than adapting to change.. It's bigotry that will exist long after most of us are gone
07:12.46ScribbleJdrmessano, yeah, there's some folks who are bigots, plain and simple; but right now it rises to a higher level because there's those folks doing what they do anyhow, and a lot of folks who /don't/ propogate that kind of thing normally who are because of the novelty.
07:12.54*** join/#asterisk zeeesh (n=zeeesh@203.215.179.43)
07:13.00ScribbleJWhen the second class calms down things'll seem more usual I expect.
07:13.11TrentCreekIt's was the people in the North who rioted after the Supreme Court forced bussing if black students...ghee..after all those Abolitionists up there
07:13.15drmessanoImagine if he was black AND republican.. the old men wouldnt know what to do with themselves
07:13.23Gopher_77*sigh* still no SIP debug messages when I try to sign in with my X-Lite
07:13.47TrentCreekGopher_77: what is going on?
07:13.58drmessanoTrentCreek: Racism is overwhelmingly prevalent in the south.  Fact of life.
07:14.09Gopher_77TrentCreek: I'm trying to hook up a softphone to *
07:14.23TrentCreekdrmessano: Yeah and it don;t mostly come from whites
07:14.33drmessanoTrentCreek: Thats not true at all
07:14.38TrentCreekGopher_77: yes,  i know, but I asked what is going on
07:14.43drmessanoTrentCreek: Ever heard of the KKK?
07:14.51TrentCreekdrmessano: BULL SHIT
07:15.06TrentCreekblacks and asians in the south are just as racist
07:15.25frogonwheelsTrentCreek: key to that sentence 'just as'
07:15.31Gopher_77KKK was put together for self-defense
07:15.32drmessanoTrentCreek: Go put your head back up your ass.  Im not gonna argue with you about this.. Whites are far more racist than anyone else.
07:15.36drmessanoLOL
07:15.42ScribbleJI'm not agreeing or disagreeing, Trent, but if they are,w ouldn't we be able to name their national orginaztions they use to spread hate, like the KK for whitey?
07:15.50TrentCreekdrmessano: We the PC bull shit states :only whites can be racist"
07:16.11carrarWOAH
07:16.12ScribbleJOh, I know plenty of racists of every color.  I'm not sure you can argue who's worse though.
07:16.15TrentCreekSure we can....NAACP
07:16.18carrarKids are in here
07:16.25drmessanoNo one said only whites can be racist.. now youre being ignorant
07:16.29TrentCreekno
07:16.37ScribbleJNAACP a hate group like the KKK?
07:16.38TrentCreekWhy is there no "Miss White America"?
07:16.42drmessanoWhich is why its impossible to have an intelligent conversation with you
07:16.43ScribbleJWhere have /I/ been?
07:16.46Gopher_77nah, that was the Black Panthers
07:16.50TrentCreek"White Entertainment Channel?"
07:16.52Gopher_77same members, different group
07:17.10drmessanoScribbleJ: yes, the NAACP burns crosses and drags white people behind trucks
07:17.17TrentCreek"White College Fund?"
07:17.21drmessanoScribbleJ: Read the news, dude
07:17.42ScribbleJdrmessano, I see it all the time, man.  I just had some NAACP dudes lynch my mom.
07:17.53drmessano<TrentCreek> "White Entertainment Channel?"  <-- The GOLF channel, dumbass
07:17.58ScribbleJHahaha
07:18.21carrarGolf is now what I think 'white' entertainment is
07:18.22TrentCreekdrmessano: Oh..is that why we only see Tiger Woods on it?
07:18.24ScribbleJI don't think I agree with you generally, Trent, but I'm all for a "White Entertainment Channel" as well as a "Straight Pride Parade"
07:18.29Gopher_77~golf
07:18.30jbotgolf is probably not a sport, it's a recreational activity that requires no physical exertion.
07:18.32carrarnow=NOT
07:18.59Gopher_77~naacp
07:19.11carrarlynch mobbing is more 'white entertainment'
07:19.19drmessanoSo having pride in ones race is RACISM... just like lynching?
07:19.27Gopher_77yeah, look at the romans
07:19.33carraroh did I say that outloud
07:19.44drmessanoSo the united negro college fund and the KKK have "the same idea, different execution"?
07:20.14carrarEvery Kingdom in Power falls
07:20.26ScribbleJdrmessano, if you're helping yourself, then obviously you're starving me of that aid you might have given to me instead.  Thus by being proud of yourself, you are keeping me down, man.
07:20.27Gopher_77having pride is one thing, playing the victim for it is another, and lynching is completely different
07:20.32TrentCreekIs this racism or what? http://www.hsf.net/Scholarships.aspx?id=460
07:20.45TrentCreekBe of Hispanic heritage
07:20.46ScribbleJOr something.
07:21.03Gopher_77how about "Other"?
07:21.15TrentCreekWhat is I put put a  White Scholarship and sai dyou have to be of Ango heritage?
07:21.24drmessanoScribbleJ: If I oppress you, and you attempt to rise from that, you deserve to eradicated, you racist pig... now get back out in the fields and pick me some lettuce
07:21.26Gopher_77who could ever prove that my mother wasn't impregnated by aliens?
07:21.42carrarhablo no spanishoo
07:21.52Gopher_77~spanishoo
07:21.57Gopher_77~spanish
07:21.57jbotHispanohablantes: Por favor usen #debian-es, alli obtendran mas ayuda. Spanish speakers: please go to #debian-es, you will get a lot more help there.
07:22.10ScribbleJI think it's terrible nonsense, Trent.  I figure if it's supported by private money, though, then it doesn't hurt anyone and those ignorant fools can throw their dollars wherever they like.
07:22.25drmessanoTrentCreek: You are exactly the reason ignorance is so prevelant in the south.  You have the same mindset that keeps the south in the 1850s
07:22.44Gopher_77~trent
07:22.52Gopher_77~jbot
07:22.53jbotextra, extra, read all about it, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch, or suck, or a pain in the ass
07:22.54carrarIs this a language battle?
07:23.14carrarI love battles
07:23.15drmessanoPor Favor
07:23.24carrartime battles rock
07:23.26Gopher_77~hack
07:23.27carrarWed Feb  4 23:23:27 PST 2009
07:23.41drmessanocarrar: you like battles?  well, I dont.. so you're racist
07:23.45carrarWhat
07:23.49carrarI live in the USA
07:23.55carrarour constitution isin English
07:24.03carrarthanks by
07:24.04drmessanocarrar: Im going by TrentCreeks line of thinking
07:24.08ScribbleJIT's 1:20, I know this because that's when the system backups kick off, draining all resources and making my pager go nuts.  Every night.  At 1:20.  Am.
07:24.18TrentCreekdrmessano: Oh..making everyone on equal terms is "igonorant?" Now that was a moronic statement
07:25.26drmessanoTrentCreek: No, you're attempting to find racism and bigotry where there is none, attempting to prove a point that other races are racist too, the only purpose being to support your way of thinking.. Which is moronic.
07:25.33ScribbleJSo on an unrelated topic, Perl rotted my brain, it's true, I've spent a whole day now basically chasing down memory allocation problems in my C code.
07:25.37contrabandais it correct : exten => _4XXXXX,n,Playback(vm-isunavail.gsm)
07:25.40TrentCreekIt's all this going around spouting and forcing differences is what is causing the problems...look at everything the same amnd all the problems go away
07:25.49contrabanda?
07:25.56carrarexten => _4XXXXX,n,Playback(vm-isunavail)
07:26.03contrabandathanks
07:26.25carrarload all the codecs in
07:26.26drmessanoTrentCreek: EXACTLY.. if everyone acted like us white people, we wouldnt have any problems, right?  Screw difference.. Screw having pride in ones race, gender, or religion
07:26.29Gopher_77carrar: so it assumes an extension?
07:26.35carrarwell
07:26.38Gopher_77carrar: and a path?
07:26.42carrarI assume 4XXXX
07:26.45drmessanoTrentCreek: Now go back and analyze your statement, and you'll know why I call you ignorant
07:26.51Gopher_77carrar: I mean file extension
07:26.59carrarthe sound path
07:27.05carrarno extension
07:27.14carrarit will trascode if it doesn;t have a match
07:27.39carrarI select and install the variations
07:27.43TrentCreekdrmessano: Oh you mean like when National Council of La raza is screaming it is "racist" to enforce our immigration laws just because 90% of illegals are Mexican?? ghee.
07:27.48carrarso it doesn't have to transcode
07:27.58Gopher_77TrentCreek: that's a good one
07:28.45TrentCreekyeah...that's enjoying their :heratige"
07:28.55drmessanoTrentCreek: Thats not even relevant to the conversation.. Just another attempt to point out that "Everyone else is doing it, so why dont we"
07:28.55Gopher_77swimming?
07:29.12drmessanoTrentCreek: Which actually supports my statement about your ignorance.. so keep going
07:29.20TrentCreekI am pointing out the freak hypocrisy
07:29.24carrarTo migrate to America, we hope you are insired enough by our freedom to also learn the language that the country was founded upon
07:29.46TrentCreekdrmessano: pioint out facts is not "igornance"
07:30.04drmessanoTrentCreek: Your lack of understand here is mind blowing.
07:30.08TrentCreekcarrar: nope..it's now "racist" to make people leanr English
07:30.14Gopher_77TrentCreek: I've tried that argument before and he ignored it
07:30.18carrarI dissagree
07:30.20carrar100%
07:30.22TrentCreekyour feeble mind is mind blowing
07:30.50*** join/#asterisk botox93 (n=botox93@213.221.82.242)
07:31.13TrentCreekIt's all okay to bash whitey in the name of "equality"
07:31.15Gopher_77our constitution and all our laws are written in the language, so it should be our "official" language, whether it's required of the citizens or not
07:31.16drmessanoTrentCreek: Its racist to tie people to trees because they're black, you fucking idiot.. and you dare say that because they have a fucking college fund that is limited to those of their own color, they are racist just the same?  What the fuck is wrong with you?
07:31.34drmessanoYou are the most ignorant person I have run into in IRC this week.
07:32.02TrentCreekwell, I may be ignorant, but not a dumb ass like you
07:32.02drmessanoYou are absolutely clueless
07:32.09drmessanoNo, you are both
07:32.22carrarI think it's ok to come in not knowing English, but it's beneficial to learn english if you want to grow in this country
07:32.25drmessanoand clearly a bigot just the same, since you can justify it
07:32.25Gopher_77google: drmessano ignorant
07:33.10drmessanoGopher_77: Stay out of this, or I will make you cry like I did the last time you tried to cross me
07:33.23carrarhaha
07:33.26TrentCreekdrmessano: yeah..just like every other "Civil Rights" organization is not bigots
07:33.37carrarWhat are talking about again?
07:33.43drmessanoIts good to see you and TrentCreek have found each other.. you're cut from the same cloth
07:33.47Gopher_77lol
07:33.47carrarextensions?
07:33.48Gopher_77cry?
07:33.50Gopher_77far from it
07:34.04drmessanoGopher_77: Oh, most of us remember how you emobitched out
07:34.06TrentCreekPeople immigratign an dnot learning English because it is "racist"
07:34.21carrarlearn english if you move to America
07:34.28carrarEnd of story
07:34.51drmessano"Civil rights" is about bigotry.. I love it
07:35.01drmessano*cough* ignorant *cough*
07:35.29fiddurcarrar: Not if you're moving to Argentine... and that's also America.
07:35.33Gopher_77where did that come from?
07:35.39*** join/#asterisk op3r (n=op3r@114.108.201.50)
07:35.44carrarAmerica != Argentine
07:35.46Gopher_77civil rights about bigotry
07:36.00carrarwell I should say USA
07:36.04TrentCreekFunny I sure don't see a "White College" anywhere in the US
07:36.18fiddurcarrar: Yep... just one country among many in america...
07:36.24TrentCreekbut gosh..how many Black Universities are there??? hmmmmmm
07:36.36drmessano[02:34] <TrentCreek> drmessano: yeah..just like every other "Civil Rights" organization is not bigots
07:36.38carrarI lvoe foreign people, I married one
07:36.49carrarSo I know all about language
07:37.03ScribbleJOn an asterisk-note, can anyone tell me whether LumenVox returns an hypothesis per frame?
07:37.04drmessanoGopher_77: If youre gonna troll me, at least follow the convo
07:37.17TrentCreekHoly crap! There are hundreds!
07:37.18TrentCreekhttp://en.wikipedia.org/wiki/List_of_historically_black_colleges_of_the_United_States
07:38.01Gopher_77hey, ther IS a white college
07:38.07ScribbleJI didn't ask that question well but I suppose there's no one here with a ton of LumenVox experience right now anyhow.
07:38.11*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
07:38.16TrentCreeknow that would be "racist"
07:38.18Gopher_77oh, that's about white college students, hmmm...
07:38.21carrarcivial rights really isn;t about language
07:38.22drmessanoScribbleJ: jaytee has done a lit with it
07:38.27drmessanolot*
07:38.38TrentCreekcarrar: well it is now...
07:38.43drmessanoScribbleJ: But hes long since fallen asleep
07:38.45ScribbleJdrmessano, sure, nicks who are here have, of course.
07:38.55carrarthen you are off definitions
07:38.55ScribbleJdrmessano, but I don't think any of them are present presently.
07:39.06ScribbleJdrmessano, er, what you said.
07:39.08drmessanoNope
07:40.04TrentCreekGopher_77: did you fix your x-lite?
07:40.10carrarLyndon B. Johnson kicked ass on Civial rights
07:40.18Gopher_77TrentCreek: no
07:40.29Gopher_77TrentCreek: my X-Lite is still out
07:40.30TrentCreekwhat;s it doing, or not doing?
07:40.31Gopher_77:(
07:40.38carrarhe was touch on civial rights then Obama is so far on the stimulus
07:40.43carrartougher
07:41.02TrentCreeklike Strom Theurman? ;-)
07:41.25carrarirconic too
07:41.32carrarironic
07:43.11carraralso
07:43.19carrarif you are gonna move to a new country
07:43.25Gopher_77Registration Error: 408 - Request Timeout
07:43.27TrentCreekGopher_77: maybe you need to check the registration box
07:43.28carrarI would want to learn it's history
07:43.40carrarfunny me
07:43.45TrentCreekAnd speak it's language
07:43.48Gopher_77TrentCreek: you mean the *?
07:43.52carrarwell that come later
07:43.56TrentCreekx-lite
07:43.58carrarlearning takes time
07:44.10carrarbut "want too"
07:44.18Gopher_77TrentCreek: you mean to register for incoming calls
07:44.34carrarFreedom *IS EVERYTHING*
07:44.34TrentCreekGopher_77: yes
07:44.43Gopher_77TrentCreek: yes, it's been checked the whole time
07:45.00drmessanocarrar: *THIRST* is everything.  I saw that on TV
07:45.02Gopher_77TrentCreek: still no SIP debug info on *
07:45.27TrentCreekfunny, when I go to Mexico they make no attempt to speak to me in English, or provide me anythign in English. Is that a form of Racism? According to "cilvil rights" groups here inthe US that would be
07:45.34*** join/#asterisk xrmx__ (n=rm@host73-252-dynamic.8-87-r.retail.telecomitalia.it)
07:45.38carrarand they shouldn't
07:45.44Gopher_77TrentCreek: there should be some debug info if it's trying to connect, right?
07:45.50TrentCreekno
07:45.52carrarunless they want your money
07:46.16TrentCreekGopher_77: it will be in the logs
07:46.37TrentCreekI think you have the wrong info in the fields
07:46.45TrentCreektell me what you got
07:47.11carrarfitty dollars
07:47.13Gopher_77TrentCreek: display, user, and auth names are the same
07:47.21Gopher_77TrentCreek: password is defined
07:47.35TrentCreekwhat is in user?
07:47.36Gopher_77TrentCreek: domain is the * ip address
07:47.47Gopher_77TrentCreek: I just called it "softphone"
07:48.13TrentCreekdoes it tie into an extension?
07:48.34TrentCreek#?
07:48.39Gopher_77TrentCreek: I intend to dial it from an extension, yes
07:48.51TrentCreekwhat is the extension number?
07:49.08Gopher_77TrentCreek: I haven't set up the extension
07:49.27TrentCreekthen set one up
07:49.38Gopher_77TrentCreek: it should register without an extension set up right?
07:49.47drmessano.....
07:49.57TrentCreekjust trying to make it easy :-D
07:50.05TrentCreekI had problems too, but mine is working
07:50.42TrentCreekonce you got the extension setup, put the extension number in the user field
07:51.12Gopher_77TrentCreek: ok I have the extension dial SIP/softphone
07:51.22TrentCreekno
07:51.42TrentCreekthe extension number becomes the user name
07:51.54Gopher_77TrentCreek: softphone is also the heading in sip.conf
07:52.06Gopher_77TrentCreek: I'll try making it a number then
07:52.11carrar[softphone]
07:53.13Gopher_77carrar: I changed all the softphone entries to 5001
07:53.24carrarif your * box in on the internet, don't use extension numbers for your [XXXX]
07:53.27Gopher_77carrar: but yes, that's what I meant
07:53.28TrentCreekoh..it's a timeout
07:53.56TrentCreeki use extention numbers..on the net..works great
07:54.15carrarI get so many scans a day for XXXX in sip
07:54.27drmessanocarrar: But it works great
07:54.27carrarpeople doing password guessing
07:54.29Gopher_77carrar: I'll change it later
07:54.40carrarworks great, but not so SMRT
07:54.48Gopher_77carrar: thanks for the tip though
07:54.56TrentCreekI also have one extension that connects to another's * to join a conference
07:54.57drmessanocarrar: Stop using mouth sense
07:55.12TrentCreekcarrar: then use stronger passwords
07:55.17carraryes
07:55.18Gopher_77drmessano: at least he has some sense
07:55.29*** join/#asterisk unasi7 (n=unasi7@84-75-21-204.dclient.hispeed.ch)
07:55.34carrarwny not even use something completely different
07:55.38drmessanoGopher_77: Shut up.. You cant even config a basic SIP extension
07:55.45drmessanoGopher_77: FAIL
07:55.50carraraccountcode+entesnion
07:56.01carraror something crazier!
07:56.04Gopher_77drmessano: yeah, you can probably do it because you've done it before
07:56.07drmessanocarrar: MAC addresses
07:56.16carraryeah MAC works too
07:56.25drmessanoGopher_77: Dont troll me if you cant take it
07:56.27carraras long as the paswors is also not the mac
07:56.31carrarpassword
07:56.35Gopher_77drmessano: wah wah
07:56.55Gopher_77I'll just use 1234:1234
07:56.57carrarMAC, ROT13 MAC for password!
07:57.00carrarheh
07:57.02drmessanoROFL
07:57.04Gopher_77like an idiot's luggage combo
07:57.05drmessanoROT13 FTW
07:58.06carrar(accountcode+extension) encrypted
07:58.10TrentCreekI downloaded Bria tlast week
07:58.29carrarBria is nice
07:58.47TrentCreekwell...i am still out if i like it
07:59.19Gopher_77so the [] contains the username?
07:59.22TrentCreeki was hoping it would be more like X-lite with the extra features enables
07:59.24Gopher_77how about the password?
07:59.41carrar[XXXX] should match the username
07:59.59carrarpassword is whatever you want
08:00.15Gopher_77password=? secret=?
08:00.25carrarsecret is the passord
08:00.28carrar+w
08:00.32Gopher_77my machine is so slow
08:02.15drmessanocarrar: In Exchange Server 2007, they have two legacy settings for admin and routing groups to make 2007 backwards compatible with earlier versions, and the names they used were FYDIBOHF23SPDLT and DWBGZMFD01QNBJR.. Shift the letters one down on one, and one up on the other, and they both translate to EXCHANGE12ROCKS
08:02.43drmessanoSo you never know where something completely laughable will turn up
08:02.54Gopher_77~exchange
08:03.08Gopher_77~expertsexchange
08:03.21Gopher_77request timeout
08:03.56carrarxlite?
08:03.57carrarhttp://pastebin.ca/1327824
08:04.28carrarthats a pretty simple setup
08:06.04carrarI delete a ton of crap from that
08:06.17carrarbut should be ok for most people
08:07.22TrentCreekmaybe there is something else wrong..check your logs
08:07.36tamielhello, in asterisk1.4 (fresh svn update), when doing an atxtransfer on parking exten (set in features.conf), agent auto hangup after caller parking . But in 1.4.23.1 (last 1.4 release), atxtransfer don't hangup agent after caller parking. What is the normal way ? It is a regression on 1.4svn ?  (Note: I must use 1.4 svn because blind transfer is buggy with sip caller)
08:07.42carrarLOGS KICK ASS
08:07.49carrarwe can always agree on that
08:08.00TrentCreekyeah...when debug dont show..logs will
08:08.34carrarwow
08:08.38carrarbleeding edge
08:08.45carrargusty
08:08.54*** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it)
08:09.34carrarcall should hang up at a park
08:09.49carrarthen it goes into "park
08:09.50carrar"
08:11.13Gopher_77carrar: that setup isn't working either
08:11.19carrarok
08:11.21Gopher_77:(
08:11.24carrarlet me simplify
08:11.51TrentCreekGopher_77: check you rlogs
08:11.54Gopher_77carrar: of course these 2 are both behind the same firewall so I put in nat=no and nat=never just to cover both
08:12.19carrarhttp://pastebin.ca/1327827
08:12.28*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
08:12.37carrarthat will assume default I think
08:12.44carraradd context
08:13.05carrarotherwise you have a connectivity issue
08:13.17Gopher_77what's this nat=1?
08:13.26carrarnat is beautifull
08:13.30carrarenbrace it
08:13.45carraror RTFM
08:13.58Gopher_77so use it even though the path is shorter without?
08:14.20tamielso if called (agent) do an atxtransfer to park caller, he must be hangup ? But how to park caller and keep line up ? (like hold line)
08:14.22*** join/#asterisk krdian (i=krdian@killer.radom.net)
08:14.22carrarhas nothing to do with 'shoter paths'
08:15.09Gopher_77and of course canreinvite=no if I'm using nat
08:15.18*** join/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net)
08:15.34carrarto each his own
08:15.36*** part/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net)
08:15.40*** join/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net)
08:15.47*** part/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net)
08:15.59*** join/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net)
08:16.00carrarIf you don't completely understna EACH AND EVERYONE command, then why are you using it
08:16.06carrar-ONE
08:16.19carraras it will effect you
08:16.24*** join/#asterisk prxtien (n=proleone@ppp121-45-69-101.lns10.adl6.internode.on.net)
08:16.40prxtiendoesnt dahdi echo cancelling still work if im doing sip to sip calls?
08:16.40ScribbleJI love to understand everything, but there's usually not time.
08:16.51prxtienwell im actually doing sip to a sip gateway, the out to pstn
08:17.03prxtienbut people i call are reporting an echo of what they say
08:17.10carrarScribbleJ, whats why people pay consultants
08:17.31Gopher_77you know, I could kick myself
08:17.43Gopher_77I forgot about iptables
08:17.46ScribbleJNot unless you're opped, Gopher.
08:17.48carrarprxtien, read up where echo is introduced
08:18.31carrarnot gonna explain echo here
08:18.38carrarLOTS of echo docs out there
08:18.46prxtienvie read the sources of echo articile on voip-info.org
08:19.07*** join/#asterisk nima0102 (n=nima@91.98.219.231)
08:19.18nima0102hi every body
08:19.23carrarHI!!
08:19.36ScribbleJHi Doctor Nick!
08:19.51carrarnice
08:19.59carrarnima0102, are you in IRAN?
08:20.15nima0102carrar: yes
08:20.26carrarHow is life over there?
08:20.37nima0102carrar: good
08:20.48carrarno crazieness outside?
08:20.57nima0102carrar: news on bbc and cnn are not true
08:21.02carraryeah
08:21.06carrarit never is
08:21.08carrarheh
08:21.08nima0102carrar: Iran is secure
08:21.16carrarheh
08:21.27carrarNew is so biased
08:21.37yangcarrar: very friendly people live in Iran, if you aren't an american tourist, nothing can go wrong
08:21.38carrarnews
08:21.46*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
08:21.51carrarNOTHING CAN GO WRONG!! :)
08:22.30carrarnima0102, IRAN likes Obama more then bush?
08:22.35TrentCreeknima0102: And Al Jazeera is? ;-)
08:22.48Gopher_77carrar, TrentCreek, thanks for the help
08:22.54TrentCreeksure
08:23.01carrarcome back anytime
08:23.07yangand you can buy a ski-ticket for just 4 EUR a day
08:23.15carrarWHAT
08:23.33*** join/#asterisk Mr_BOnD_007 (i=bond0070@119.160.199.6)
08:23.38carrarSki tickets In Seattle (area) are like $30-$50
08:23.39TrentCreekWouldn't that be Anit-Islamic?...ooopps..that's Saudia Arabia
08:23.44yangIn Iran
08:23.45carrarPERDAY
08:23.55nima0102carrar: I like peace on the world, then any president want peace I like him [her]
08:24.10yangnima0102: are you trying to build an Asterisk server ?
08:24.17carrar:)
08:24.20carraryes
08:24.21nima0102yang: yes
08:24.24carrarHow can we help
08:24.47carrarI digress
08:25.02yangnima0102: good, you will probably need to buy some hardware which is probably hard to get in Iran (I assume)
08:25.38carrardid I miss something?
08:25.39TrentCreekcarrar: Obama is suppose to creat us 2 million jobs within 18 months
08:25.45nima0102I want to use asterisk on our office, what web interface do I select ? astersik GUI or Freepbx
08:25.46yangnima0102: but hey, only if you will do Analog/ISDN telephony, you can still use SIP-based link with the VOIP provider
08:26.02Mr_BOnD_007OBAMA :D Good news ;) but what about who lost the job right now :D
08:26.06TrentCreekcarrar: but what he did not tell you, that he will allow 2 million people immigrate during that same time peroid
08:26.10nima0102yang: where are you from ?
08:26.15yangnima0102: Europe
08:26.16carrarnima0102, this channel is mainly for people who compile Asterisk from source
08:26.22carrarand are CLI based
08:26.34carrar(Cammand Line Interfaced)
08:26.41op3ranyone knows any web based asterisk monitor?
08:26.46nima0102carrar: I have compiled astersik from source on debian
08:26.56carrarYou  might want #asterisk-gui
08:27.00TrentCreekop3r: FreePBX is one
08:27.03Mr_BOnD_007op3r i think VICIDIAL < GOOd option dont know much more about taht
08:27.12nima0102carrar: Do you use web interface ?
08:27.18carrarno
08:27.28carrarI am %100 command line
08:27.38carrarexcept for my Switchvox installs
08:27.41TrentCreekop3r: PBX ina Flash is a whole Asterisk package with FreePBX and other added goodies
08:27.56nima0102carrar: surely you are  professional about asterisk
08:28.00TrentCreekI like FreePBX
08:28.12carrarI like to think I am ok
08:28.12TrentCreekno more darned typing everything
08:28.39carrarAsterisk since 2001
08:28.46nima0102TrentCreek: do you suggest freePBX ?
08:28.57TrentCreekbut FreePBX is not so user friendly..meaning it has a learning curve
08:29.00yangnima0102: for GUI try #elastix (www.elastix.org) or FreePBX (www.frepbx.org)
08:29.22carrarmaybe it was 2002
08:29.25carrarwhatever
08:29.27TrentCreekor PBX ina Flash
08:29.37TrentCreekeven comes with Linux distro
08:29.56TrentCreekinstalls everything for you
08:29.57nima0102yang: I have installed Asterisk ,and I do not want reinstall asterisk
08:30.25TrentCreeknima0102: then FreePBX is one way to go
08:30.52yangnima0102: this channel is for command line driven asterisk as carrar said before
08:31.09Gopher_77hmmm... calls are failing: Service Unavailable
08:31.12nima0102yang: excuse me for this mistake
08:31.30TrentCreeknima0102: you have a connection issue
08:31.46TrentCreeknima0102: actually this channel is for Asterisk in genral
08:31.52carrarnima0102, install from source
08:31.57nima0102TrentCreek: why ?
08:31.57carrarthats really the best way
08:32.17yangcarrar: there is a good asterisk allready packaged into debian,which you can use
08:32.17carrarwe're just not GUI people here
08:32.19*** join/#asterisk scruz (n=scruz@41.220.73.170)
08:32.24scruzhello
08:32.28TrentCreekbecause there are other channels for addons
08:32.36TrentCreekyou can join #freepbx
08:32.36op3rTrentCreek: freepbx != web based monitor?
08:32.42carrarI download everything and install
08:32.43nima0102carrar: Does handle asterisk 200 users?
08:32.47carraryes
08:32.50carrareasy
08:32.51TrentCreekop3r: yes it is..I have FreePBX installed
08:32.53yangcarrar: depends on your machine
08:33.08op3rerrrr
08:33.17carrarI have several clints 200+ phoes on 1 box
08:33.21carrarphones
08:33.24TrentCreeknima0102: it depends on your hardware
08:33.30op3rI wanted something like monast without the lag :(
08:33.37carrardual proc boxes
08:33.56carrarto be safe to have lost of overhead
08:34.22TrentCreekA P4 with 1 Gig canhandle 200+ calls, if they are pure digital calls only
08:34.42nima0102Trentcreek: may suggest efficient hardware for 200 users?
08:34.45carrarHow many conncurent calls
08:34.49carrarthats key
08:35.10carrarP4 single
08:35.13carrarP4 dual?
08:35.21TrentCreeknima0102: Are you doing any analog to digital conversion for the voice>
08:35.26carrarwhat else is on that box
08:35.32nima0102carrar: 20 up to 40 concurrent
08:35.43carrarthats tight
08:35.56*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
08:35.57carrarmaybe
08:36.00yanghi mort_gib
08:36.04carrarif 100% sip passthrough
08:36.17nima0102Trentcreek: I hav to voip gateway for PSTN connection, tenor
08:36.22mort_gibHi Yang, how are things?
08:36.31yangmort_gib: quite wel, thanks
08:37.05yangnima0102: for PSTN switch you will need some additional hardware
08:37.12TrentCreekmort_gib: okay then a basic computer can handle 200 calls...else if you start connecting to land line, you need a LOT more computing power
08:38.02nima0102yang: I want to use tenor as FXO, is it ok ?
08:38.08yangnever heard of it
08:39.10mort_gibTrentCreek: Huh??
08:39.13*** join/#asterisk ChannelZ (i=channelz@burner.com)
08:39.27TrentCreeksorry...eyes are shutting down..clicked wrong name
08:39.47TrentCreektiem to sleep
08:39.55yangnima0102: did you allready purchase tenor equipment?
08:40.05ChannelZI'm on a TDM800 -- should ztmonitor be able to 'see' the dialtone when I make a call?
08:40.06*** join/#asterisk frogonwheels (n=michaelg@203.59.141.93)
08:40.57nima0102yang: tenor is product of Quintum company,
08:41.24nima0102yang: In Iran tenor gateway is good and many company are using that
08:41.44mort_gibTrentCreek: Don't quote me for that please it's wildly incorrect
08:42.01yangnima0102: I would give more trust to companies like Sangoma, Digium, xorcom which are all well integrated into asterisk, you might have driver issues with tenor, I never heard about it.
08:42.41yangso, i cannot say if its good or bad
08:43.09nima0102yang: http://www.quintum.com/products/products.html
08:44.07yangnima0102: that is why I suggested other brands, if you haven't purchased tenor yet...go with the others
08:44.36nima0102yang: please suggest other brands
08:44.46nima0102yang: sangoma
08:44.51nima0102yang: ?
08:44.54carrarWHAT DID I MISS
08:45.37carrarnima0102, go with Digium every time
08:45.44yangnima0102: Sangoma/Digium/Xorcom are among the popular
08:46.14carrarhttp://pics.osburn.com/photo/5522/medium (asterisk early days)
08:46.55nima0102yang: these are PC-based hardware, but tenor is separate hardware that can asterisk trunk to tenor gateway
08:47.14yangcarrar: your asterisk from 2001 ?
08:47.21carraryup
08:47.35yanginteresting
08:48.55carrarI got sucked into it
08:49.23carrarjust after FreeI when Chapter 11
08:49.26carrarwent
08:49.42carrar1.0 days
08:50.05carrarI sprang for the X100
08:50.28carraras caller ID didn't work on the cheapy cards
08:51.35nima0102yang: Is Sangoma more expensive than digum ?
08:51.47carrar8 ports VS 4 ports?
08:51.52carrarprobably
08:51.54yangnima0102: I am not familiar with the prices, you will need to find the appropriate seller
08:52.26carrarT1's are CPU intensive
08:52.29*** join/#asterisk matthew___i (n=matthew@adsl-163-35-202.hsv.bellsouth.net)
08:53.46carraryang, did you notice the date
08:53.55carrarI happen to take a pic
08:54.03carrarrandom chance
08:54.20yanghm, same day as today?
08:54.35carrarno
08:54.51carrarI was watching IP TV
08:54.57carrarvia MBP4
08:55.00carrarerr
08:55.03carrarMBGP4
08:55.09carrarmulitcast
08:55.47carrarscreen shot was live
08:55.56carrarhttp://pics.osburn.com/photo/5521/original
08:56.11*** join/#asterisk denon (i=denon@synapse.subneural.net)
08:56.11*** mode/#asterisk [+o denon] by ChanServ
08:56.44yangcarrar: oh my :)
08:57.32carrarhistory is crazy
08:57.55Gopher_77goodnight
08:57.58carrarso many things happen that day
08:58.05carrarwas weird
08:59.41yang~tell nima0102 about book
08:59.46tzafrir_laptopChannelZ, yes. Normally the dialtone is noticably different. Though with the one-dimetional display there it's sometimes not as easy to spot
08:59.54carrar~book
08:59.55jbotwell, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
09:00.20*** part/#asterisk Gopher_77 (n=Jim@cpe-71-72-19-206.neo.res.rr.com)
09:00.21ChannelZtzafrir_laptop: I was capturing the audio and actually listening to it - I think I got it working now though, needed to grab the 'pre-echo cancelled' stream
09:01.01ChannelZTrying to debug a dialout problem, it sounds like my telco is being (randomly) lazy about providing a dialtone when asterisk goes to make an outgoing call, and it just starts dialing before they're ready
09:02.12liriis it normal that when an extension enters a queue then it most of the time 'hears' silence instead of ringback? because thats what seems to happen
09:02.43carrarwtf
09:03.13carrarwhats the ring stratigy
09:03.29carrarwho's a member
09:03.37carrarwho is logged in
09:03.56tzafrir_laptopIf you have captured audio, any audio editor with a wave display (e.g. audacity) will clearly show it
09:04.01carrarwhats the "hold music"
09:04.09*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.198)
09:05.05tzafrir_laptopcore show application MusicOnHold
09:05.32ChannelZyes I'm getting it now, I wasn't before with ztmonitor -f because it apparently mutes it; ztmonitor -F works.
09:05.33tzafrir_laptopoh, this isn't the asterisk CLI?
09:06.08carrarthis is
09:06.11carrarand only is
09:06.22ChannelZthanks, and g'night
09:06.24*** part/#asterisk ChannelZ (i=channelz@burner.com)
09:06.25mort_gibhey guys, what does this mean:  chan_sip.c: We could NOT get the channel lock for SIP/
09:07.20carrarhttp://tinyurl.com/burjn6
09:07.49tzafrir_laptopIf you're having any issues with echo and such, looking at the different sides separately (-r/-R , -s/-S IIRC)
09:09.09liricarrar: was that directed to me?
09:19.43mort_gibAnyone get's :  chan_sip.c: We could NOT get the channel lock for SIP/
09:45.35nima0102what softphone do I use on linux that supports  alaw/ulaw codecs ??
09:45.39*** join/#asterisk HoverHell (n=hell@92.241.225.10)
09:47.15yangnima0102: www.ekiga.org
09:48.56nima0102yang: I am using EKIGA for now but it does not support these codecs
09:49.13nima0102there is no these codecs on list codec
09:49.22*** join/#asterisk HoverHell (n=hell@91.146.50.221)
09:52.12frogonwheelsnima0102: yeah - I think they're disguised.
09:52.35frogonwheelsnima0102: ONe of the numbered codecs is actually ulaw/alaw
09:52.55frogonwheelsnima0102: I only allow ulaw on my system, and have had ekiga working.
09:53.20nima0102frogonwheels: thanks for your reply, what is its number?
09:53.37frogonwheelsnima0102: google is your friend on this one.
09:55.44frogonwheelsnima0102: ahh. or it could be PCMU PCMA
09:55.59*** join/#asterisk contrabanda (n=contr@92.241.69.134)
09:56.09nima0102frogonwheels: thanks for your attention
09:56.26contrabandaI got errors on CLI "chan_sip.c:3751 sip_write: Asked to transmit frame type 4, while native formats is 0x400 (ilbc)(1024) read/write = 0x8 (alaw)(8)/0x4 (ulaw)(4)"
09:56.30contrabandahow can i fix it?
10:04.17*** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl)
10:11.57kerxhi everyone for some reason i am not receiving caller id's when using Asterisk AMI, but I do:    Callerid: Name <number>
10:12.01kerxIs there another method to make this work?
10:14.55LuisTorresMorning..., anyone know an application like faxdetect that can be used for outbound?
10:21.19*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
10:42.37*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
10:48.53*** join/#asterisk sergee (n=serg@voip1.west-call.com)
10:52.40*** join/#asterisk kerx (n=kerx@adsl-69-104-77-234.dsl.irvnca.pacbell.net)
10:57.11*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
11:10.09*** join/#asterisk tamseel (n=IceChat7@116.71.216.23)
11:10.17tamseelhi all
11:10.57tamseeli have problem with g729 installation with i7 system
11:11.48tamseel(h)
11:16.24*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
11:16.47*** part/#asterisk MrWGW (n=MrWGW@74.124.206.166)
11:31.39*** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar)
11:33.30*** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44)
11:35.07*** join/#asterisk Greek-Boy (n=greek@41.222.89.70)
11:39.45*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
11:48.22*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net)
11:52.36*** join/#asterisk Mr_BOnD_007 (i=bond0070@119.160.199.6)
11:59.50*** join/#asterisk scythe (n=scythe@pD9E100E0.dip0.t-ipconnect.de)
11:59.53scythehi
12:04.16*** join/#asterisk SkywaIker (n=pirch@58.147.17.166)
12:06.19*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-359f2d511e49a63e)
12:09.38*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
12:13.43*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
12:16.19*** join/#asterisk MrNaz (n=mrnaz@ppp118-208-209-224.lns10.mel6.internode.on.net)
12:21.18pifhi, anyone has a musiconhold phone number outside of france? I need to test a sip provider for two simultaneous calls
12:26.15bochdoes asterisk rotates its own logs ?
12:27.34sipyno it doesn't rotate by default. it has logrotate that you have to put in cron
12:29.26*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
12:29.58sipyand run 'logger rotate' to rename the old/start new one and reload logger.com
12:30.03sipylogger.conf
12:30.26*** join/#asterisk frogonwheels (n=michaelg@203.59.141.93)
12:31.15*** join/#asterisk scruz (n=scruz@41.220.73.170)
12:31.48scruzhi
12:32.10sipyhi
12:37.40*** join/#asterisk criten (n=criten@216.75.233.220.exetel.com.au)
12:37.41*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
12:48.24*** join/#asterisk path_ (n=path@227-124-21-190.adsl.terra.cl)
12:48.45*** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il)
12:53.03*** join/#asterisk DarkRift (n=dark@65.92.250.117)
12:54.53*** join/#asterisk anonymouz666 (n=anonymou@189.24.79.73)
12:55.51*** join/#asterisk frogonwheels (n=michaelg@203.59.141.93)
13:00.54*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net)
13:08.25*** join/#asterisk ingenius (n=alektro@69.90.72.173)
13:10.47*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
13:13.32*** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi)
13:13.57*** join/#asterisk Khratos (n=khratos@190.166.103.180)
13:14.14ZenBSDiHey guys.. file convert in asterisk
13:14.25Khratos'morning
13:14.30ZenBSDimorning..
13:14.33Lymag'day
13:14.34ZenBSDi:)
13:14.42Khratos:)
13:14.53ZenBSDiAnyone know about file convert in asterisk?
13:15.04ZenBSDi1.2 doesn't seem to have it.. did I miss something to enable it?
13:16.04ZenBSDidamn it .. i't s only 1.4 :p
13:16.06ZenBSDigrrrrr
13:16.12ZenBSDithanks guys
13:16.26*** join/#asterisk _gm (n=gmustafa@202.133.78.60)
13:17.43*** part/#asterisk squish102 (n=squish10@cpe-098-024-067-184.carolina.res.rr.com)
13:21.01*** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp)
13:21.45*** join/#asterisk inam (n=root@116.71.216.23)
13:21.45*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net)
13:22.05*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
13:23.43inami have some compilation problmes while installing g729 codecs asterisk-g72x-1.0-beta8.tar.bz2 on i7 processor's
13:28.06*** part/#asterisk scruz (n=scruz@41.220.73.170)
13:28.12*** join/#asterisk oej (n=olle@ns.webway.se)
13:28.45contrabandahello
13:30.11contrabandai need help with ooh323. i have h323 hard phone and would like to register on asterisk. i have created config in ooh323.conf
13:30.17contrabanda[dito]
13:30.17contrabandatype=peer
13:30.17contrabandaallow=all
13:30.17contrabandacontext=int
13:30.18contrabandahost=92.241.69.134
13:30.43contrabandabut when i run ooh323 show peers i dont see my peer registered
13:31.01contrabandaphone is configured in P2P mode
13:31.04contrabandaplease help me
13:31.29[TK]D-Fendercontrabanda: P2P mode sounds like "I don't register"
13:31.36[TK]D-Fendercontrabanda: Change your phone's mode
13:32.01[TK]D-Fendercontrabanda: And you should have host=dynamic if you want it to ABLE tor egister
13:32.34contrabandaok should i change phone mode to GK?
13:32.43[TK]D-Fendercontrabanda: Sounds better...
13:33.29*** join/#asterisk jeremey_g (n=root@195.134.229.2)
13:33.43jeremey_ganyone used asterisk Bridge app
13:35.01path_is there a way to make public what extensions are available ?
13:35.34*** join/#asterisk freckle (n=chatzill@195.74.96.118)
13:36.22*** join/#asterisk marsilainen (n=matt@87-194-208-236.bethere.co.uk)
13:37.14*** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim)
13:37.40[TK]D-Fenderpath_: copy your extensions.conf onto a web-server :)
13:38.09path_hahaha
13:38.11jeremey_gtwo sip phones registered with my * box are talking to each other. * receives a sip call that arrives on its extension 2000. Now how can i connect this call to the phone A and then hangup the phone B.
13:38.32jeremey_gany * guru outthere to answer this
13:38.36*** join/#asterisk tobias (n=tobias@cpe-071-070-218-098.nc.res.rr.com)
13:39.26*** join/#asterisk dlewis_ (i=c7340d66@about/security/staff/dlewis)
13:40.51[TK]D-Fenderjeremey_g: ...HUH?
13:44.39jeremey_g[TK]D-Fender :i am trying to use bridge app
13:45.14jeremey_g[TK]D-Fender :wanna see my dialplan
13:49.18*** join/#asterisk jeremey_g (n=root@195.134.229.2)
13:49.23jeremey_ghttp://www.pastebin.ca/1327981 <--sip.conf
13:50.13jeremey_g<PROTECTED>
13:50.13jeremey_g# #[34m                # #   ##(B      #[1m#(B#      ##[1m##[1C              #[1#[1C    #(B           #           #   # #[1m##(B##### # #         #[1m##   #(B##                 #           #   ##[1m###(B#[1m#(B####(B#(B                #[1m  #[1m##            ##       ##(B       ##(B#   #  # #####(B     ###   #   ###(B
13:50.13jeremey_g<PROTECTED>
13:50.57*** join/#asterisk jeremey_g (n=root@195.134.229.2)
13:50.59jeremey_gsorry
13:51.08contrabanda[TK]D-Fender: Still have problems with h323 hardware phone registration on asterisk. :( please suggest what i have to do
13:51.36[TK]D-Fenderjeremey_g: Your description was rather poor.
13:51.40jeremey_gw] [ kaldemar  ] [ hads      ]
13:51.41jeremey_g[ jeff      ] [ Pagautas  ] [ KingOfDos ] [ JT        ] [ stintel   ]
13:51.41jeremey_g[ twisted   ] [ DaPrivatee] [ inv_arp   ] [ carrar    ] [ MooingLemu]
13:51.41jeremey_g[ scooby2   ] [ hohum     ] [@Corydon76-] [ elguero   ] [ keulin    ]
13:51.41jeremey_g[ fnordus   ] [ NoxIn-    ] [ andrewy   ] [ adr3nalin3] [ hawk      ]
13:51.41jeremey_g[ rob0      ] [ jbot      ]
13:51.43jeremey_g#(B#[1m#(B [Users(#asterisk:3)]
13:51.45jeremey_g[ justdave  ] [ tAnkOSX   ] [ derrick_  ]
13:51.47jeremey_g#(B#[1m#(B Channel #asterisk was created at Sun Nov 26 07:42:43 2006
13:51.49jeremey_g<jeremey_g> sorry
13:51.51jeremey_g#(B#[1m#(B BitchX: Join to #asterisk was synched in 1.414 secs!!
13:51.52[TK]D-Fenderjeremey_g: PASTERBI
13:51.53jeremey_g<contrabanda> [TK]D-Fender: Still have problems with h323 hardware phone
13:51.55jeremey_g<PROTECTED>
13:51.57jeremey_g#(B#[1m#(B SignOff inam: #asterisk (Remote closed the connection)
13:51.59jeremey_g<[TK]D-Fender> jeremey_g: Your description was rather poor.
13:52.00stintelargh
13:52.01jeremey_g<PROTECTED>
13:52.01*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
13:52.03jeremey_g<PROTECTED>
13:52.07jeremey_gi am so sorry
13:52.09*** kick/#asterisk [jeremey_g!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender)
13:52.11*** join/#asterisk jeremey_g (n=root@195.134.229.2)
13:52.14stintellol
13:52.17KingOfDoshaha
13:52.17stintelchatting as root and all
13:52.19jeremey_gits just that i am on a system with no decennt window manager
13:52.25jeremey_g:)
13:52.27sipyanyone install the latest stable asteriskNOW iso? the install is broken.
13:52.27KingOfDosas root? that's bad
13:52.32[TK]D-FenderPEBKAC <-
13:52.37tzafrir_laptopjeremey_g, what system?
13:52.42KingOfDos+b *!root@*
13:52.46KingOfDos;)
13:53.07[TK]D-Fendersipy: and "latest stable" doesn't tell us which.  NUMBERS dammit!
13:53.30jeremey_gi meant to use paste bin but the x windows and wm is minimal and doesnt have a luxury such as a right click clipboard
13:53.41jeremey_gi use the middle mouse button for copy paste o
13:53.49*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
13:54.11[TK]D-Fenderjeremey_g: Ok, enough witht he story, just don't do it again
13:54.13tzafrir_laptopjeremey_g, I always use that . On xterm (e.g.: for irssi) you can also use shift-insert
13:54.22sipytk: did you not get your 8 minutes of sleep? 1.0.2
13:54.24jeremey_ghttp://www.pastebin.ca/1327984
13:54.29jeremey_gthats my extensions.conf
13:54.35[TK]D-Fendersipy: hardly stable :p
13:54.42[TK]D-Fendersipy: get 1.5  SERIOUSLY
13:54.50[TK]D-Fendersipy: 1.0.2 uses friggen rPath Linux
13:55.02[TK]D-Fendersipy: 1.5 uses Centos and includes FreePBX
13:55.08*** join/#asterisk write_erase (n=Olivier@LNeuilly-152-22-89-110.w193-251.abo.wanadoo.fr)
13:55.27jeremey_gso any suggestions to my problem
13:55.31*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
13:55.39stintelbrrr rPath
13:55.44jeremey_ghow to bridge an incoming sip call with an ongoing sip call
13:56.04[TK]D-Fenderjeremey_g: GOOD pastebin please....
13:56.10*** join/#asterisk jjshoe_ (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net)
13:56.28sipy1.5 beta stable??
13:56.30sipyhehe
13:56.34jeremey_gin my case, 2011 is dialing  2010 and a call is up. then a call arrives at 2000 that needs to be connected to 2011
13:56.38jayteejjshoe, you in Madison?
13:56.40sipystabeta
13:56.45jeremey_gi am on 1.6
13:56.46[TK]D-Fenderjeremey_g: PASTEBIN what you are doing NOW.
13:56.46jjshoejaytee no but close
13:56.51write_eraseHi ... How can I write an extend based on a callerid  ? I'd like to answer/hangup a spam callerID  thx
13:56.55jjshoejaytee what's up?
13:57.07jayteesaw the tds.net and the loco and just wondered.
13:57.17jayteeused to work for a subsidiary of tds
13:57.24jjshoejaytee ahh yeah no, switched from fonality to working with Tony and Ethan
13:57.40jeremey_gwhats wrong with my bridge app use? am i doing it wrong?
13:57.44[TK]D-Fenderwrite_erase: exten => _mypattern/2125551234,1,NoOp(Hangup on that  212 SOB)
13:57.56[TK]D-Fenderjeremey_g: Where is the pastebin?
13:58.22jjshoejaytee consider me un-impressed with tds, they do the re-directing of domains that don't exist
13:58.37*** join/#asterisk rpaulson (n=rpaulson@72.80-202-243.nextgentel.com)
13:58.45jayteejjshoe, and their subsidiary butt surfs it's employees
13:58.47marsilainenHi, I can make outbound calls via a SIP provider, sipgate.co.uk, but I can't get inbound calls working. When I dial my number I get a voice message "the number you have called is not compatible with this service"
13:58.52[TK]D-FenderHIS NAME IS ROBERT PAULSON
13:58.57jjshoejaytee surfs?
13:59.06path_How can I define voip lines to specific extensions ?
13:59.12marsilainenthe call does seem to get through to my asterisk box, I get the following sip debug info: http://pastebin.com/mc16cb80
13:59.16marsilainenany ideas?
13:59.17[TK]D-Fenderpath_: PARDON?
13:59.21jayteejjshoe, just an expression
13:59.31contrabandahere is my H323.conf file http://pastebin.ca/1327990. i have defined user there but can not register my hardware h323 phone there. please help me
13:59.31jjshoejaytee yeah, wondering what it means :D
13:59.41[TK]D-Fenderjjshoe: "rides their asses"
13:59.51jayteejjshoe, think of prison, showers and bending over for soap
14:00.14jjshoejaytee ahh, hahaha, yeah, I just got out of that, night and day here.
14:00.18jayteewho is Robert Paulson?
14:00.19path_[TK]D-Fender, when I'm setting up the user, I can see an option 'Line number'
14:00.45[TK]D-Fendermarsilainen: in your [sipgate] peer do "insecure=very"
14:01.00[TK]D-Fendermarsilainen: They probably don't like the auth challenge when sending you a call
14:01.03*** join/#asterisk docelmo (n=vircuser@pool-70-110-120-47.lyn.east.verizon.net)
14:01.11marsilainenok, thanks, will give it a try
14:01.57[TK]D-Fenderpath_: ......
14:02.01write_erase[TK]D-Fender, thx
14:02.31path_[TK]D-Fender, never mind I figured it out thanks anyway :)
14:04.09marsilainen[TK]D-Fender: many thanks, that did the trick
14:04.12*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:04.23[TK]D-Fendermarsilainen: You're welcome
14:05.04*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
14:10.31*** join/#asterisk boomboom99 (n=boomboom@c-71-229-40-177.hsd1.ga.comcast.net)
14:10.43contrabandahere is my H323.conf file http://pastebin.ca/1327990. i have defined user there but can not register my hardware h323 phone there. please help me
14:10.48*** join/#asterisk Tili (n=tili@58-27-163-244.wateen.net)
14:11.38jeremey_gDoes anyone remember that you can only bridge channels created with Local? Is that true?
14:11.46jeremey_gLike the bridge app works only for Local
14:12.19boomboom99I got iaxmodem and hylafax working on my main server and have sent and received several faxes just fine.  got another server on a fiber optic connection with the exact same setup and we can't send faxes between the 2 over a iax trunk.  any idea why?  ulaw used for all channels...
14:13.27boomboom9927 ms ping over the IAX trunk
14:18.07[TK]D-Fenderjeremey_g: No.
14:18.30[TK]D-Fenderjeremey_g: bridge is for CHANNELS.  the device on the end does not matter
14:21.19*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)
14:21.25jeremey_g[18~but what am I doing wrong then?
14:22.06jeremey_g[TK]D-Fender : could you please look at my .confs
14:22.13jeremey_gthere are fairly simple
14:22.45sipywaiting for PBND!
14:22.51[TK]D-Fenderjeremey_g: 5 lines of conf isn't going to tell us much of anything
14:23.08jeremey_gdidn't you get the full pastebin link
14:24.00[TK]D-Fenderjeremey_g: No, we didn't
14:24.27sipyhehe
14:25.55*** join/#asterisk cesau2 (n=cesau@66.94.94.66)
14:27.20jeremey_ghttp://www.pastebin.ca/1328012 <-- extensions.conf
14:28.59*** join/#asterisk intralanman (n=Raymond@va-67-76-163-209.sta.embarqhsd.net)
14:30.44jeremey_ghttp://www.pastebin.ca/1328015 <--sip.conf
14:30.48jeremey_g[TK]D-Fender :can you now?
14:31.23[TK]D-Fenderjeremey_g: 2 SIP peers?  You aren't using Bridge in there anywhere.. where is the ATTEMPT?
14:31.58*** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56)
14:32.08jeremey_gin extensions.conf, i receive a call at extenion 2000 from another party that wishes to connect to 2010 which is already talking to 2011
14:32.28*** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44)
14:32.42[TK]D-Fenderjeremey_g: Show us the ATTEMPT
14:32.56jeremey_gso there in extension 2000, you can see the Bridge
14:33.02jeremey_gis that wrong?
14:33.11[TK]D-Fenderjeremey_g: Show us the ATTEMPT <----------
14:34.50jeremey_gok
14:36.32contrabandaI need help with h323. here is my H323.conf file http://pastebin.ca/1327990. i have defined user there but can not register my hardware h323 phone there. please help me
14:37.06*** join/#asterisk contrabanda (n=contr@92.241.69.134)
14:37.14*** part/#asterisk contrabanda (n=contr@92.241.69.134)
14:37.15*** join/#asterisk contrabanda (n=contr@92.241.69.134)
14:37.54jeremey_ghttp://www.pastebin.ca/1328024 <--ATTEMPT
14:37.54*** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
14:38.48jeremey_gobservation:  note that the parameter that bridge app gets contains text Zombie, also,bridge returns non-zero
14:40.56contrabandaI need help with h323. here is my H323.conf file http://pastebin.ca/1327990. i have defined user there but can not register my hardware h323 phone there. please help me
14:41.38[TK]D-Fenderjeremey_g: Go look at BRIDGERESULT when it exits out, next the call you are going from is not authing and for all I know is incompatible.
14:42.04[TK]D-Fenderjeremey_g: Why is that call coming in unauthed?
14:42.35jeremey_g[TK]D-Fender :it is coming from another trusted proxy sitting next to my *
14:42.59jeremey_g[TK]D-Fender :should i set insecure=very
14:44.31[TK]D-Fenderjeremey_g: You should be looking at that call in detail to see what CODECS it offered, ports assigned, etc.
14:44.54*** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com)
14:46.02*** join/#asterisk n3hxs (n=HAMming@c-76-105-181-171.hsd1.or.comcast.net)
14:47.13kerxHi everyone, I am having some difficulties with caller id's on AMI and call files that originate a SIP channel through my provider.  The majority of the calls that originate do not have caller ID and show the name as unknown and the number also, some calls actually show the caller ID.  Any idea where I could begin to fix this?
14:47.24jeremey_g[TK]D-Fender :the call comes in and * sends an OK back. when * attempts to bridge, it sends a bye to the A-party
14:48.07[TK]D-Fenderjeremey_g: And you continue to not SHOW us your debug.  Maybe you're MISSING something, and I'm tired of running blind
14:48.27jeremey_gok let me do that
14:49.50jeremey_gwhat verbose/debug levels would you like?
14:50.12jeremey_gis it 8
14:51.00*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
14:51.27*** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca)
14:51.48[TK]D-Fenderjeremey_g: Set up a peer for them and include SIP DEBUG.
14:51.56*** join/#asterisk nima0102 (n=nima@91.98.219.231)
14:52.20nima0102hi
14:52.33kerxhi
14:52.52nima0102Does everybody know about price of sangoma A200 ???
14:53.04nima0102Does everybody know about price of sangoma A200 with 2 FXOs???
14:53.10coppiceyes. the resellers
14:54.50*** join/#asterisk adam000 (n=adam@c-76-97-76-93.hsd1.ga.comcast.net)
14:54.52*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
14:54.55jeremey_g[TK]D-Fender :how to sip debug on 1.6. I know only on 1.2?
14:55.08jeremey_gcore set sip debug?
14:55.40jeremey_gsip set debug on
14:55.41jeremey_ggot it
14:55.45kerxHi, anyone know why sometimes asterisk would set CallerID correctly and sometimes it will not ?
14:55.52adam000What's the correct pronunciation of "Aastra"??
14:55.57cesau2just sip set debug, eh?
14:56.11jameswfah-st-ruh
14:56.43kerxaaaaah-ssssssss-trrrrruck <-not
14:57.21jameswfass truck, what happens when you drive 300 miles in the desert with no AC anf cloth seats
14:57.36kerxheh
14:57.46*** join/#asterisk dlewis_ (i=c7340d66@about/security/staff/dlewis)
14:57.52kerxi'm willing to give it a shot if i can get my caller id's to work! :)
14:59.49*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
14:59.49*** mode/#asterisk [+o russellb] by ChanServ
15:00.18jjshoeI pronounce it "Ass Truh"
15:02.56*** join/#asterisk Faustov (i=user@gentoo/user/faustov)
15:03.32Faustovhi, regarding pots connected to fxo card in a server with * - how many parallel calls can a single line handle?
15:05.48coppiceadam00: you can't correctly pronounce aastra, but all the pronouncable dot coms had been taken
15:06.36jayteeFaustov, that would be one
15:07.46Faustovic
15:08.34Faustovthanks jaytee
15:09.07*** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca)
15:09.23Faustovbtw, http://halokwadrat.pl/212.html <--- you guys might wanna add this to the bot, this site has a very nice explanation how sangoma cards work
15:09.41Faustovenglish version is default
15:10.51[TK]D-FenderFaustov: WOW!
15:11.02[TK]D-FenderFaustov: Looks just like http://sangoma.com/product/demo/n/a200/w/572/h/508.html on the page : http://sangoma.com/products_and_solutions/hardware/analog_telephony/a200.html
15:11.09[TK]D-FenderFaustov: AWESOME.
15:11.23[TK]D-FenderFaustov: Just like the manufacturers own site..... only poorly translated!
15:11.34[TK]D-Fender....
15:11.37[TK]D-Fenderpwned
15:11.39[TK]D-Fender:p
15:11.52[TK]D-FenderNEXT!@@!@!! (c) BKW
15:12.56jayteethank god, because my polish is really rusty
15:15.25*** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod)
15:15.29RModhello
15:16.05RModim having an issue with agi scripts, after a while they all get stuck and say defunct
15:16.08RModany insight?
15:16.42[TK]D-FenderIf you don't like... what you see here... get the funk out!!!!
15:16.52[TK]D-Fenderspins up some more Extreme
15:16.57Faustov[TK]D-Fender: heh use any opportunity to pwn, right? :> well i don't think that was available when i was researching on it
15:17.06Faustovwell
15:17.07Faustovnevermind
15:17.37[TK]D-FenderFaustov: It was hidden in the last place anyone would think to look.... on the manufacturer's page for the product.... I understand COMPLETELY!
15:17.38kaldemarlooks like someone's found a bigger mug for coffee.
15:17.39*** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net)
15:17.40kerxhttp://kenno.wordpress.com/2007/04/04/how-to-kill-defunct-process/
15:18.02kerxfirst line explains what a defunct means
15:18.10Faustovkill their mother!
15:18.13kerxi have no idea why your AGI's would do that though
15:18.16Faustovthen destroy the eggs!
15:19.04*** join/#asterisk kmiller_ (n=kmiller@rrcs-96-10-27-206.se.biz.rr.com)
15:19.08kerxRMod:  It seems like this guy in the mailing list has had the same issue:  http://www.archivum.info/asterisk-users@lists.digium.com/2008-08/msg00838.html
15:19.19boomboom99well, I found the fix to my faxing issue myself.  for those of you interested: transfer=no in the iax.conf for my iaxmodem entries made it work.  I guess Asterisk was trying to link my iaxmodems directly instead of handling the channel itself.
15:19.21kerxI think it contains the reason, that might be what is happening to you
15:20.07*** join/#asterisk adnc (n=adnc@unaffiliated/adnc)
15:20.48*** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net)
15:20.58*** join/#asterisk saftsack (n=oliver@f051103116.adsl.alicedsl.de)
15:21.03rbd_yo kmiller_
15:21.32*** join/#asterisk Makaveli4life (n=chatzill@196.206.236.248)
15:21.40adnci've setup a SVR record on my dns in order to point to my dynamic hooked up asterisk, none of my friends do use voip, is there a service online i could ask to initiate a call to my address in order to test this?
15:21.43kmiller_Is it possible to use stream file as a URL like so: "STREAM FILE voiceglue/url/127.0.0.1:9070/accessCastOrPlaylist?returnType=.mp3 "
15:22.04kmiller_I'm getting back the following error:  " WARNING[19582]: file.c:602 ast_openstream_full: File voiceglue/url/127.0.0.1:9070/accessCastOrPlaylist?returnType=.mp3 does not exist in any format"
15:22.58*** join/#asterisk watchy2 (n=watchy@76.196.98.139)
15:23.10watchy2hmm none of my polycoms at this sites buddywatch is working
15:23.15*** join/#asterisk docelmo (n=vircuser@pool-141-152-218-208.lyn.east.verizon.net)
15:23.17*** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4)
15:23.40*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:23.50[TK]D-Fenderkmiller_: No, it isn't
15:24.16[TK]D-Fenderkmiller_: It says "STREAM FILE" not "STREAM insertwhateverfreeformstuffyouwanthereandexpectittowork" :)
15:24.57kmiller_can I url quote those characters out?
15:25.09*** join/#asterisk Slashman (n=Slash@ariane.fimasys.com)
15:26.35rbd_I'm with kmiller_ ...using voiceglue (asterisk VXML library) ...basically we're trying to use a query URL in an <audio> tag (instead of a static file)...everything works up to this point, and STREAM FILE (which voiceglue calls) doesn't like that file name...so if there is no way to "friendly up" the query string, it looks like a patch to voiceglue is in order....
15:27.04jameswfAric Caley formaly of trixbox needs a job, http://trixbox.org/forums/trixbox-forums/open-discussion/so-long-and-thanks-all
15:32.33SparFuxWill the standard extension be used when there is no other extension suitable? and will the callerID be preserverd?
15:33.00*** join/#asterisk DonX (n=don@222.sub-70-197-173.myvzw.com)
15:33.33SparFuxCan I place a phone call via ISDN and use an arbitary CallerID?
15:34.40DonXIs there any way that I can make a queue mimic follow-me functionality....IE: When ringing my members, I want it to prompt for member interaction in order to accept the call...Otherwise it will just ring to the next person
15:35.14[TK]D-FenderDonX: "core show application dial" <- M()
15:35.23DonXSpar, depends on the Carrier and type of ISDN ckt
15:35.36*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
15:35.52SparFuxI use Hansenet in Germany and type of isdn ckt? I don't know. EuroISDN.
15:36.06SparFuxWhat is ckt?
15:36.18DonXcircuit
15:36.22SparFuxAh.
15:36.32SparFuxI think it's DSS1, Euro ISDN.
15:36.34DonXIE: PRI, BRI, etc
15:36.41*** join/#asterisk awk_r (n=awk_r@nat/digium/x-49b6bbd5fd5e1f23)
15:36.49*** part/#asterisk awk_r (n=awk_r@nat/digium/x-49b6bbd5fd5e1f23)
15:38.02DonXI'm not familiar with that signaling, but if it's not working then caller id is being set by your provider
15:38.17DonXthis will override anything that you send
15:38.25SparFuxDonX: I got BRI, I think. Standard 2-port HFC-S chip.
15:38.51DonXIf you don't setup any caller ID at all, does it show up as unavailable or your number/name?
15:39.07SparFuxas unavailable.
15:39.10DonXhrmm
15:39.17DonXthen yes, you should be able to
15:39.27watchy2hmm it seems none of my polycoms are watching the hints
15:39.30SparFuxhm....
15:39.43DonXhow are you setting the caller id?
15:39.43jplankwatchy
15:39.57jplankare you getting a 481 when the asterisk sends a notify?
15:40.11SparFuxDonX: but I do set Callerid() and it doesn't work.
15:40.40*** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman)
15:41.42watchy2jplank: i'm not seeing any errors, how can i check that
15:41.43DonXSet(CALLERID(num)=XXXXXXXXXX)
15:41.52DonXis that what you're doing?
15:42.31*** join/#asterisk Mattchis (n=Miranda@c-98-199-191-45.hsd1.tx.comcast.net)
15:42.40jplankwith a wireshark trace, I only ask because that just started happening to me with firmware 3.0 and 3.1.1
15:42.42SparFuxYes, that's exactly it. I have 10 MSNs (phone numbers) and I can set CALLERID() to all of them, but when I use an arbitrary number, then the number actually transmitted is the first of my MSNs.
15:42.57watchy2jplank: did you fix it?
15:43.03jplankno
15:43.21watchy2jplank: i have 1 site thats working fine and one thats not
15:43.27jplankI just noticed it the other day, haven't had too much time to work on it
15:43.29watchy2one is like 3.1 or something one is maybe 2.8
15:43.46jplankis the non-working one 3.1?
15:43.52SparFuxDonX: perhaps, then it isn't possible :-(
15:43.53watchy2ones 2.2.2.0084
15:43.54watchy2yea
15:44.15watchy2i have the polycom's directorys set with <bw>1
15:44.22watchy2but none are looking at them
15:44.44watchy2i wonder if its a bug
15:44.45DonXsounds like it's being set by your provider
15:44.48DonXcall and ask
15:45.11jplankthey all show offline?
15:45.20watchy2well
15:45.31watchy2<PROTECTED>
15:45.31watchy2<PROTECTED>
15:45.35watchy2show hints show that
15:45.40watchy2no one is watching anyone
15:46.05cesau2where are engines typically registered? such as odbc
15:46.21The_Boy_Wonderi need a timing interface for my iax trunk.  does dahdi have any tools for this?
15:46.24watchy2i wonder if i should drop the highest 2.x firmware and let them use that
15:46.31*** part/#asterisk Mattchis (n=Miranda@c-98-199-191-45.hsd1.tx.comcast.net)
15:46.48SparFuxDonX: I think it might be my provider who sets the CallerID  to the first one when I try to use a CallerID which does not belong to me.
15:46.59jplankwhat does the phone show?
15:47.15watchy2no idea i'm not onsite
15:47.19watchy2its to far away to drive their
15:47.23DonXcould be...
15:47.26watchy2but its apparently never worked
15:47.40DonXthey would certainly be able to tell you
15:48.06DonXI have PRI's that I send caller ID along without any issues
15:48.15jameswf~pb
15:48.15jbot[~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, or , http://bin.cakephp.org/
15:48.20jameswfoy newbs
15:49.18jplankwatch2: if you have wireshark installed on the box, type tshark -i any port 5060 at the linux prompt, then call one of the extensions being watched, see if you see the notify being sent to the phone thats supposed to be watching it, and if so, what does that phone reply back
15:49.53watchy2ok i'll install wireshark
15:50.02*** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-25786515e1a41c6e)
15:50.16Gido-Etcpdump is also capable of showing the sip headers
15:50.33[TK]D-FenderASTERISK is capable of telling you what it sees & sends
15:51.01jplanksorry, I use a lot more equipment then asterisk, so I'm more comfortable using wireshark, to each his own
15:51.09[TK]D-Fenderwatchy2: What firmware & models are you using now?
15:51.32SparFuxAh, now there is a command Gosub() I can use to not have to hangup and recall every time I do something on my homeline like listening to mailbox.
15:51.38watchy2hmm
15:51.46watchy2670s, 3.1 i think
15:51.55watchy2i think google just answered my issue
15:52.19jplankwhat did you find?
15:52.38watchy2<PROTECTED>
15:52.40watchy2what the fuck
15:52.45watchy2its turned off by defauilt?
15:52.49watchy2now?
15:52.50jplankever since 3.0
15:53.02watchy2i bet thats my issue
15:53.09jplankvery well could be
15:53.17jplankthen that doesn't explain my issue :P
15:53.43watchy2i wish i knew of a 670 i could reboot
15:53.49*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-68f661bb89ac3a3f)
15:53.49*** mode/#asterisk [+o putnopvut] by ChanServ
15:54.00jplankdo you have the reboot option turned on inside the phone?
15:54.19watchy2i can reboot it manually using a sipnotify command
15:54.29watchy2don't wanna wig out the secretaries though
15:54.30jplankthen yes ;P
15:54.32*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
15:54.40watchy2you know how stupid people are
15:54.41[TK]D-Fenderwatchy2: It has been defaulted to OFF since long before I started with * <-
15:54.43jplankthats also turned off be default ever since 3.0
15:54.46watchy2"my phone went blank!!"
15:55.02jplankfender, is that true? I thought that was new to 3.0?
15:55.14watchy2i've never had to enable it in sip.cfg before
15:55.44jplankwatch2 I usually use a dialplan app to reboot the phones, and then just email the secretary to do it
15:56.18watchy2i never thought of dialing a # to reboot it
15:56.53jplankwatchy2: let me know if that works, because then that means I have a bigger problem
15:57.02watchy2yea i'm about to try it
15:57.13AJFisherHi.  I'm having issues with the CDR.  When I Originate calls using the manager API, the CDR field 'answer' is blank, billsec is 0 and disposition is 'NO ANSWER'
15:57.14jplankI haven't been able to get it to work on 3.0 and up
15:58.14[TK]D-Fenderjplank: I've used down to 1.4... so no, its pretty much always been "off" by default
15:58.24jplankinteresting
15:58.42jplankwhat about the reboot on cfg change?
15:59.35jplankfinder, any idea why the polycom would send a 481 in reply to a SIP notify?
15:59.41jplankfender*
15:59.54watchy2jp: a 330 is rebooting now with a populated directory
15:59.57jplankI know I'm giving you see much info to work off of
16:00.00watchy2i'll see if it works on bootup
16:00.09AJFisherI'm using asterisk 1.6.0.5.  When calls are made normally, the cdr fields are correctly populated.
16:00.13[TK]D-Fenderjplank: doesn't really matter  IMO
16:00.22jplankwhat do you mean?
16:00.33jplankseems to matter, the phone thinks all the other phones are offline
16:00.37rue_mohrok question, how can I use a 1mw from the pots to determine if the incomming levels are right?
16:00.49[TK]D-Fenderjplank: the response.  OPTIONS is jsut used for Qualify NAT keep-alive anyways, and * doesn't care about the response
16:00.54watchy2<PROTECTED>
16:00.57watchy2fixed jplank
16:01.42rue_mohrit seems to me that if there is excessive far end echo, the singnal may be going out too hot
16:01.49*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-45c17a16b9793d6d)
16:01.49*** mode/#asterisk [+o Deeewayne] by ChanServ
16:02.17jplankso then maybe I have my config screwed up, * is sending a Notify to the phone watching all the other phones when a call is made from the other phones, and the phone replies back a 481, and never shows the right hints
16:02.59*** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it)
16:03.00watchy2mine seems fixed jplank, but untill i rebooting the 670s i won't know
16:03.11watchy2the secretaries will have to let me know
16:03.19jplankyou don't know if the phone is actually seeiing the other ones right
16:03.35watchy2well i wasn't getting any watchers on any now, now this 330 is watching
16:03.51watchy2so i'm know its paying attention now.
16:04.29jplankoh I should of mentioned, my phone shows as a watcher :P
16:04.54jplankI thought you said yours WAS showing as a watcher
16:05.32talirk81When using a php agi script how can you "dial a number" and get get the result of that dialed commands. I dont recall want to use a manager if i can help it , i would rather just send something to STDOUT and  let asterisk return something to STDIN is this possible?
16:06.00jplank<PROTECTED>
16:06.28xrmx__speaking of polypcom phones, anybody knows how can i make the 330 stop pausing first call when i have another call on second line?
16:06.35cesau2I'm getting an engine not available error when trying to load from odbc -- i debugged the problem to the config_engine_list not having any entries in config.c -- any ideas why i wouldnt get odbc in my config_engine_list? http://pastebin.com/m7ead5b90
16:06.38*** join/#asterisk sack (n=sack@69.Red-79-148-190.dynamicIP.rima-tde.net)
16:08.50*** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56)
16:11.00Corydon76-digcesau2: did you preload res_odbc and res_config_odbc from modules.conf?
16:11.14*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
16:12.00cesau2! zomg
16:12.41cesau2just res_odbc
16:12.55Corydon76-digWell, you need both
16:15.08SwK_anyone seen slepp around lately?
16:15.21cesau2uncommenting preload => res_config_odbc.so didnt change the result at all =/
16:15.58*** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis)
16:16.09*** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)
16:16.43jjshoehrm, isn't there a config option to adjust voicemail to email volume?
16:17.05cesau2Corydon76-dig: I love you!
16:17.06Corydon76-digjjshoe: yes
16:17.11watchy2hey tk, you got a list of feature codes you use?
16:17.19cesau2Thank you so much!
16:17.20Corydon76-digcesau2: calm down; my boyfriend will get jealous
16:17.39watchy2haha
16:17.50watchy2i wish i had a boyfriend
16:17.56cesau2such a lucky guy!
16:17.58*** join/#asterisk seanmh (n=johndoe@abq-216-31-109-157.dsl.zianet.com)
16:17.59jjshoeCorydon76-dig lol, do you know the name of it?
16:18.04denonjjshoe: yes, g(#)
16:18.07denonjjshoe: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail
16:18.09jjshoeCorydon76-dig looking at voicemail.conf on voip-info and nothing is poppig out at me
16:18.17jjshoedenon thanks.
16:18.37denonnp
16:19.00jjshoedenon ahh, not quite the right thing, but mentions "volgain"
16:19.29jjshoeof course nothing mentions acceptable input for volgain
16:19.58jjshoeand not in my asterisk book, more digging, at least I know the option now
16:20.38coppiceif you need to tamper with gains at the point of recording your system is kinda broken
16:20.55denonjjshoe: volgain is passing to sox, so likely the valid range is sox's range
16:21.37denoncoppice is probably right, but sometimes reality dictates needs for stuff like this
16:21.47denonbut might be worth checking to see if you can fix the real issue instead of just gaining things
16:22.33Corydon76-digI dunno, sometimes the volume is adequate in the PBX, but isn't adequate on Windows
16:22.42*** join/#asterisk CunningPike (n=arodgers@204.239.10.119)
16:23.02Corydon76-digSo clearly, Windows is broken?
16:23.08coppicea *lot* of people have bizarre gain settings, such that voice is almost permanently clipping. you should see some of the funky audio files that have been sent to me for FAX or EC problem analysis :-\
16:23.24denonor user is using poor audio levels anyway
16:23.27*** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk)
16:23.52jjshoeCorydon76-dig :) let them soap box!
16:24.51*** join/#asterisk SlicerDicer (n=kvirc@69-92-107-4.cpe.cableone.net)
16:25.07*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
16:25.36*** join/#asterisk yoda1410 (n=yoda@aether.hipocoon.be)
16:27.17*** join/#asterisk cesar_CR (n=cesar@200.91.75.67)
16:28.11*** join/#asterisk The_Boy_Wonder (n=davidvos@nat/digium/x-ac10a94735f5f89e)
16:29.02rue_mohrhaha, I'm reading articles written by my buddy kb1 http://lists.digium.com/pipermail/asterisk-users/2005-March/089437.html  figures I'd know the only guy who knows the most
16:30.45rue_mohrwhat I need though is a way to determine outdoing levels
16:32.25jplanki dont get this, why is it when I play VM it sooo low? but if I download the file and listen to it, it sounds perfect?
16:33.11rue_mohrsounds like you system dosn't have the power to muster playing it
16:33.36rue_mohror your using hands free without the mic muted and the phone is picking up room noise
16:33.38jplankregular recording work find
16:34.05jplankauto attendant, the pre recorded VM sounds (allison) everything else sounds fine
16:35.11[TK]D-Fenderjplank: because you get to play with your soundcard for playback
16:35.27[TK]D-Fenderjplank: it isn't distorted, just low.
16:35.47[TK]D-Fenderjplank: So the base gain on the channel you recorded from is too far from baseline
16:36.33jplankwhen you say the channel, do you mean the SIP channel, or something else?
16:36.40[TK]D-Fenderrue_mohr: You know... in my experience Sangoma's seem to do AGC where I get to leave everything at 0.0 and never get echo or anything ;)
16:36.48[TK]D-Fenderjplank: the one that recorded the VM
16:37.16rue_mohrwell, I'm working with stick shift here
16:37.45coppiceI doubt they AGC. it would be disasterous
16:38.12rue_mohrwell, I'm sure the digium card is sending too strong a signal
16:38.20rue_mohror the impedence is wrong
16:38.43rue_mohrI'm gonna do some stuff at home with my channelbank and my scope this weekend and find out what 'normal' is
16:39.54*** join/#asterisk killown (n=Yamato@unaffiliated/killown)
16:40.28[TK]D-Fendercoppice: Well... I'll leave it that I've never had to touch it.
16:40.51[TK]D-Fendercoppice: perhaps I've flubbed as well and it was impedence matching.
16:41.09coppicealmost all touching of telephony gains is wrong. people just go crazy
16:41.30[TK]D-Fenderrue_mohr: You are using yourself as a baseline comparison of "normal"?  Your logic is tragically flawed :p
16:41.39rue_mohryou know why excessive far end echo occurs?
16:41.43[TK]D-Fendercoppice: Shorter trip for some than others ;)
16:41.57coppiceis a telephony card did any AGC, no modems would work, EC wouldn't work, lots of things demand a stable channel
16:42.02rue_mohrI'll use a normal pots phone hooked right up, get numbers and make things match
16:43.24[TK]D-Fendercoppice: I believe what I was thinking of was indeed impedence matching...
16:43.42rue_mohrhow do I turn the hwec off?
16:43.51[TK]D-Fenderrue_mohr: "echocancel=no"
16:44.01rue_mohrand reload dahdi?
16:44.10[TK]D-Fenderrue_mohr: sure
16:44.11rue_mohrcause I aleady did that and restarted asterisk
16:44.46coppicethe modern analogue interface devices have tunable impedance matching. that's what fxotune tweaks. they won't fine tune automatically, though. the country code you set will select coarse tuning
16:46.01rue_mohrthe echo canceling is still ON I can tell when I dial
16:46.10rue_mohrI get echo on the first digit
16:47.23rue_mohrwhat vpb.conf?
16:47.36rue_mohrchan_dahdi.conf:;echocancel=yes
16:47.42rue_mohroh wait, hehe
16:48.00rue_mohrthis sounds like a case of missing defaults
16:48.16jplankfender, I'm sorry, but I still dont follow, call comes in from SIP/foo to SIP/user user is unavailable so SIP/foo leaves a message, SIP/user listens to the message, and message is very low, but then SIP/foo calls back, SIP/user answers, and everything sounds perfect
16:48.24path_Im having this weird issue. I have multiple voip accounts on my asterisk, I want some of the incoming calls go to specific extensions though there are a sort of overlapping behaviour.
16:48.26jplankwhere are you saying I need to increase the gain, on SIP/foo?
16:49.22SparFuxin a special context can I have a way to always get back DISA(context) no matter where I did Goto() or whatever happens at the time? Like no matter wether in mailbox or whatever?
16:49.38path_If I dial to one extension or another it gives the same output.. though they have different incoming call rules.
16:49.51*** join/#asterisk bbryant (n=brett@68.208.65.50)
16:51.40[TK]D-Fenderrue_mohr: nice ASSumption
16:52.59*** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk)
16:53.23[TK]D-FenderSparFux: make sure your call does a dial to a Local channel with a dynamic applicationmap feature that sends them back to a similar local exten (so this can recurse) and call's DISA
16:54.01path_this when I call to my first voip account: http://pastebin.com/d116cd8a3 .Now this is when I call to the other voip account http://pastebin.com/m5c50ebd6
16:54.17SparFuxFender: I once tried features for this. I think I'll have to try again. Should work.
16:55.47*** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44)
16:55.58jplankfender, I'm sorry I don't understand. Are you telling me I need to increase the volume level on my soundcard for the playback of VM to be louder?
16:56.08[TK]D-Fenderpath_: You are hitting the same exten (s) in the same CONTEXT for both calls
16:56.48[TK]D-Fenderjplank: No, I'm saying that because you are playing it on your soundcard you ahve control over the speaker knob and it just happens to sound fine.  your recording IS low however.
16:56.59[TK]D-Fenderjplank: Fix the gain on the channel that LEFT the message
16:57.52*** part/#asterisk Makaveli4life (n=chatzill@196.206.236.248)
16:58.26path_exten = s,1,Goto(default,10001,1)
16:58.34path_default would be the context, right?
16:58.45jplankfender this maybe where I'm losing you, if SIP/foo calls SIP/user the volume is fine, if SIP/foo calls SIP/user and leaves a VM, the sound is low. Your saying I need to adjust the gain on SIP/foo?
16:58.53[TK]D-Fenderpath_: Go look at the SIP debug of the 2 incoming calls to see if you SHOULD be able to differentiate between them
16:59.14[TK]D-Fenderpath_: - Executing [s@DID_8668:1] Goto("SIP/8668-b6327a58", "queues|46000|1") in new stack
16:59.29[TK]D-Fenderpath_: s@DID_8668:1 <--------
16:59.47[TK]D-Fenderjplank: bOTH OF YOUR PHONES ARE ODD THEN
16:59.54[TK]D-Fenderjplank: What are you using?
17:00.15path_[TK]D-Fender, but they're different in the extensions.conf
17:00.22jplankwell, SIP/foo is my SBC, SIP/user is a polycom
17:00.30*** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net)
17:01.26[TK]D-Fenderpath_: Look at whats getting CALLED
17:01.38[TK]D-Fenderpath_: the call is being dumped in the same damn place
17:01.53jplankagain, its only playback of the voicemail, even if I download the wav file (I know, speakers etc, but even if I turn the volume down to its lowest setting) it still sounds better
17:01.54path_yes, I know.. thats what freaking me out
17:01.55[TK]D-Fenderjplank: SBC = says nothing
17:02.00[TK]D-Fenderlunch time, BBIAB
17:02.00jplankits a nextone MSx
17:02.19jplankbut I doubt thats the issue, every other call that flows sounds find, its just the VM message
17:02.33jplankeven if I use the same channel to record a greeting for the IVR, it sounds fine
17:02.38path_but http://pastebin.com/d5b8aa856
17:03.07path_is the same If I dial to 7499 than If I dial to 8668
17:03.26path_If I change the incoming rule for 8668 automatically changes the other one
17:03.49jplankI wouldn't be against upping the the volume on the SIP endpoint, but I can't use rxgain on a endpoint in sip.conf, can i?
17:05.00*** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net)
17:12.52*** join/#asterisk rnovotny22 (n=david@209.150.201.166)
17:13.51*** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net)
17:14.00*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176206367.dsl.bell.ca)
17:16.30*** join/#asterisk codefreeze-lap (n=murf@72.21.67.40)
17:16.42talirk81When using a php agi script how can you "dial a number" and get get the result of that dialed commands. I dont recall want to use a manager if i can help it , i would rather just send something to STDOUT and  let asterisk return something to STDIN is this possible?
17:16.53talirk81recall=really
17:18.23*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
17:19.34SparFuxFender: can't I just reset the whole call and make asterisk pretend that the connection had just been established?
17:21.45*** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net)
17:25.51SparFuxThis is all really complicated. It's not possible to return the call to the beginning in a general way.
17:28.17*** join/#asterisk tamseel (n=root@116.71.216.23)
17:28.45*** part/#asterisk tamseel (n=root@116.71.216.23)
17:29.22theharbye rusellb! have fun in brusells!
17:30.03*** join/#asterisk tamseel (n=root@116.71.216.23)
17:30.45tamseeljoin /freepbx
17:31.04*** part/#asterisk tamseel (n=root@116.71.216.23)
17:32.42carrarw00t
17:33.33[TK]D-FenderSparFux: The trick is enforcing that you always have rights to activate that DTMF.. you can only do that through Dial
17:34.51[TK]D-Fenderpath_: -- Executing [s@DID_8668:1] Goto("SIP/8668-b6327a58", "queues|46000|1") in new stack
17:35.06[TK]D-Fenderpath_: Your channel is the same as its matching the same peer for both calls.  Go look at the SIP DEBUG like I told you
17:38.19jplankfender, could of just told me about the VMGAIN option in the voicemail application :P
17:38.39*** join/#asterisk grantm (n=grant@68.142.138.4)
17:38.46[TK]D-Fenderjplank: No, that is PATCHING the problem... any other recording, otherdevices etc will all still be off
17:38.50[TK]D-Fenderjplank: wrong approach
17:44.11*** join/#asterisk michaely (n=Mike@207.114.199.107)
17:44.15*** part/#asterisk michaely (n=Mike@207.114.199.107)
17:44.21*** join/#asterisk jets (n=brian@72.22.244.222)
17:44.32jetsis there any easy way to know which modules are present on a TDM410
17:44.43jetsdmesg tells me four, but I do not know which type of module they are
17:44.51*** join/#asterisk oej (n=olle@ns.webway.se)
17:45.13jets(and I do not have physical access to see the color :(
17:45.46*** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net)
17:47.43jplankfender, thats the weird part, other recordings and devices sound perfect, it was JUST VM recordings
17:48.12[TK]D-Fenderjplank: becasue all other devices are EQUALLY screwed.
17:48.28[TK]D-Fenderjplank: VM records at the level you send.  if its too low... then everything is too low
17:48.41jplankthats so weird
17:49.07jplanklike I said, all other recordings sound perfect, it was JUST VM recordings (not even the greetings)
17:49.58jplankwe are planning on getting a new server for *, and we were going to rebuild from scratch, so when I have that in the lab, I'll start testing on their with this same config (minus the VM gain) and see if that has the same problem
17:50.21[TK]D-Fenderjplank: Like I sad... Vm PROMPTS are NORMALIZED.  what you record is your SEND GAIN.  THAT is bad.
17:50.27jplankthis is a production server, so I had to at least get a band aid to get it working
17:50.29[TK]D-Fenderjplank: Is this so hard to comprehend?
17:50.35jplankI don't mean VM prompts
17:50.41jplankthe recorded VM greeting
17:50.51jplankor the recorded IVR
17:50.54jplankthey all sound perfect
17:51.43jplankis it possible to boost the RXgain on a sip endpoint?
17:52.25jplankthen I could remove the VMgain, boost the SIP channels rxgain, and retest, if its at a normal level, I know the issue is with the endpoint, if its still low, I know the issue is with the *
17:52.28*** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178)
17:53.17ruben23hi
17:53.52*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
17:57.15[TK]D-Fenderjplank: It isn't * unles you set the VMgain ata ll
17:57.19[TK]D-Fenderat all*
17:57.27WHYSI want email-to-fax - looking for the path of least resistance - do I start with Hylafax or AsterFax?
17:58.51[TK]D-FenderWHYS: RxFax + 1-line shell script
17:59.14WHYStThanks, I'll check it out.
17:59.23*** join/#asterisk nima0102 (n=nima@91.98.219.231)
18:00.12nima0102Hi
18:00.43*** join/#asterisk jupeterson (n=John@c-24-126-160-141.hsd1.ga.comcast.net)
18:00.45nima0102in order to connect my asterisk server to PSTN, do you suggest solution ??
18:00.56*** part/#asterisk scythe (n=scythe@pD9E100E0.dip0.t-ipconnect.de)
18:01.56WHYSyou should but an ATA
18:01.59jupetersonhey all!  can someone help me.. I'm trying to set up Asterisk 1.6 to store voice messages in MySQL and I'm getting this error messag
18:02.05jupeterson<PROTECTED>
18:02.05jupeterson[Feb  5 17:21:03] WARNING[27374]: res_odbc.c:281 ast_odbc_direct_execute: SQL Exec Direct failed.  Attempting a reconnect...
18:02.05jupeterson[Feb  5 17:21:03] NOTICE[27374]: res_odbc.c:815 odbc_obj_connect: Connecting asterisk
18:02.05jupeterson[Feb  5 17:21:03] NOTICE[27374]: res_odbc.c:843 odbc_obj_connect: res_odbc: Connected to asterisk [MySQL-asterisk]
18:02.06jupeterson[Feb  5 17:21:03] WARNING[27374]: app_voicemail.c:3149 insert_data_cb: SQL Direct Execute failed!
18:02.08jupeterson[Feb  5 17:21:03] WARNING[27374]: res_odbc.c:281 ast_odbc_direct_execute: SQL Exec Direct failed.  Attempting a reconnect...
18:02.11jupeterson[Feb  5 17:21:08] NOTICE[27374]: res_odbc.c:815 odbc_obj_connect: Connecting asterisk
18:02.15jupeterson[Feb  5 17:21:08] NOTICE[27374]: res_odbc.c:843 odbc_obj_connect: res_odbc: Connected to asterisk [MySQL-asterisk]
18:02.18jupeterson[Feb  5 17:21:08] WARNING[27374]: app_voicemail.c:3258 store_file: SQL Execute error!
18:02.20jupeterson[INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)]
18:02.26kaldemar/kick jupeterson
18:02.40path_[TK]D-Fender, is there a way to grab the whole debug ? its too much
18:03.32jupetersonsorry..didnt mean to paste the whole thing
18:03.43kaldemar~pb
18:03.43jbot[~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, or , http://bin.cakephp.org/
18:03.47jupetersondidnt kow it was all in my buffer
18:03.58kaldemarjupeterson: ^use that in the future
18:03.59jupetersonanyone seen this error before?
18:04.23KhratosDoes Asterisk AMI sends some kind of signal or EOF notification after you send the logoff command ?
18:05.24KhratosOr just the 'Response: success" string?
18:06.42*** join/#asterisk queuetue (n=scott@MTRLPQ02-1279391519.sdsl.bell.ca)
18:07.09jupetersonanyone know how to setup asterisk to store voice messages in mysql ?
18:09.16queuetueI'm about to install a blank machine as a new * pbx.  What's a recommended platform?  (In the past, I started with Debian and installed from scratch, but I was hoping there might turnkey installs that are worth using, maybe with one of the UIs?)
18:09.36kaldemarKhratos: why don't you telnet to the port and try it?
18:09.38theharI use Ubuntu but I hear CentOS is the way to go?
18:09.51theharor use asterisknow
18:10.07*** part/#asterisk catpants (i=catpants@c-71-228-179-232.hsd1.al.comcast.net)
18:10.26queuetueAsterisknow is in the middle of abandoning the current tech and adopting freepbx, right?
18:10.29KhratosTelnet will not acctually show if some sort of \0 \r \n or EOF information is sent, it will only process it
18:11.53kaldemarKhratos: well if you tried it, you would know that it sends "Response: Goodbye" paired with a message.
18:14.42jayteewell, the soldering iron is heating up and the motherboard is in position.
18:16.29*** join/#asterisk MrTelephone (n=test@h697179-171.picriverisp.net)
18:16.30*** join/#asterisk grantm (n=grant@68.142.138.4)
18:16.57saftsackqueuetue, why using a gui? i would prefer debian with a blank asterisk
18:17.49MrTelephoneWhen someone turns on CALLERID blocking asterisk doesn't know which user to authenticate because the from header says From: Anonymous <anonymous@10.1.3.50>. The problem is I have two lines at 10.1.3.50 so it uses the wrong user from sip.conf. Does anyone know a solution or work around for this?
18:17.57queuetuesaftsack, Like I said, that's what I'm used to, but getting lazy in old age, looking for a trustworthy UI to do heavy lifting. :)  If there isn't one, then I can just do it the way I usually have.
18:19.28rnovotny22Does anyone know how to change vmail.cgi to display locale time instead of GMT?
18:20.17saftsackthen try freepbx
18:24.59carrarMrTelephone, do the blocking on *, not the phone then
18:26.00*** join/#asterisk nicoAMG (i=asgalt@201.203.96.42)
18:26.00[TK]D-Fenderrnovotny22: vi vmail.cgi
18:27.01MrTelephonetoo bad we couldn't get asterisk to authenticate using the username from the Remote-Party-ID header
18:27.23MrTelephoneif you set trustrpid=yes will asterisk authenicate with the remote-party-id?
18:27.30rnovotny22Thanks Fender, I just don't know where in the file the time stamp needs to be changed.
18:29.22QwellMrTelephone: RPID != authentication
18:29.57*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
18:30.58MrTelephoneI'm trying to think of ways I can get the client to hint to asterisk who the user is besides looking at the DEF CONTACT info
18:31.28davevg-btwtechrnovotny22, it appears it is just outputting the value from origdate in the msg...txt file, you could either change your vm settings to specify the timezone the extensions are in, or look for anything in vmail.cgi with $fields->{'origdate'} and convert them
18:32.48jaytee~grandstream
18:32.49jbot[grandstream] the Yugo of VoIP hardware.  Run.  Run away now.
18:33.40*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
18:34.05[TK]D-FenderMrTelephone: RPID is for using auth and passing the other CID info on the backside
18:34.27[TK]D-FenderMrTelephone: Make sure to NOT use "insecure" in your peer
18:36.00MrTelephoneI'm going to try and do nat mapping on the first 2 sipura lines so that the Lines each have a different return port.. hopefully that is enough to distinguish who the user is..
18:36.29[TK]D-FenderMrTelephone: have them use their AUTHID
18:37.03MrTelephoneauthid is turned on
18:38.12[TK]D-FenderMrTelephone: So why is it you're not authing these guys full like normal?
18:38.26[TK]D-FenderMrTelephone: You want an end user to configure their CID via the enpoint?
18:38.31MrTelephonethe problem stems from the sipura using source port 5060 for line 1 and line 2
18:38.49MrTelephoneits an 8 port device locked up in a room, they can't change their caller info
18:39.22*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
18:39.22*** mode/#asterisk [+o russellb] by ChanServ
18:39.23MrTelephonelines 3-8 all use different source ports by default so asterisk can compare the user to their def. contact info
18:39.42MrTelephonei beleive thats what is happening
18:39.53MrTelephonebecause client side caller blocking works fine on the cable modems
18:40.29jupetersonanyone storing voice messags in MySQL via ODBC here?
18:40.36rue_mohrhaha, the hwec wont turn off
18:40.48MrTelephoneAddr->IP     : 10.1.3.15 Port 50000
18:40.53MrTelephoneAddr->IP     : 10.1.3.15 Port 500234
18:41.05MrTelephonefor example there are two clients from the same device but differnet source addresses
18:42.09*** join/#asterisk grantm (n=grant@68.142.138.4)
18:42.53[TK]D-FenderMrTelephone: 6-digit Port #'s WHEE!
18:42.55rue_mohrok, if I have echocancel=no in chan_dahdi.conf, it should turn off the hwec right?
18:43.00*** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net)
18:43.05[TK]D-Fenderrue_mohr: Depending
18:43.09rue_mohron what?
18:43.18[TK]D-Fenderrue_mohr: Where the line appears...
18:43.52rue_mohrafter echocancel=no
18:44.43kaldemarthat solves it.
18:44.46[TK]D-Fenderrue_mohr: the line appears after ITSELF?
18:44.57[TK]D-Fendergrabs his ClueBat (tm)
18:45.30kaldemarthe way things have been this week, i'm running out of pop corn soon.
18:45.36rue_mohrhow would I do that,cmon, use context dude, the echocancel=no comes before the channel definitions
18:46.33rue_mohrhttp://eds.dyndns.org/~ircjunk/not_public_dont_open/phonesys/asterisk/chan_dahdi.conf
18:46.47rue_mohrI set all 3 echo settings to  no
18:47.17rue_mohrthats the version where their set to yes
18:47.27rue_mohrIset them (the echo settings) to no
18:48.08MrTelephonelinksys needs to be bombed
18:48.50*** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)
18:49.08rue_mohrso the hwec cant be turned off
18:49.10[TK]D-Fenderechotraining=yes <- bad
18:49.40rue_mohri told you that the file I gave the link for was the "yes" version and that I had set them all locally to "no"
18:49.53[TK]D-Fenderrue_mohr: know what... I'm not wading through hundreds off lines of junk... clean up your mess
18:50.05rue_mohrthats the default mess, not mine
18:50.11[TK]D-Fenderrue_mohr: and I'm not taking your word for it that you did it right
18:50.26[TK]D-Fenderrue_mohr: You left it that way, so its YOUR mess now.
18:50.27kaldemarposting irrelevant configurations doesn't help much either.
18:51.21rue_mohrthe only difference is that I set echocancel to no, echocancelbridged to no and echotraining to no
18:52.31rue_mohrso I'll repost the file so you can see they are no and not help me turn the hwec off, in an hour or so I'll just give up, publish the fact somewhere that asterisk CANNOT turn off the hwec and pull the card while I work on setting my levels
18:52.42rue_mohrso lets hurry up and get this overwith
18:54.59MrTelephonei manually changed the source sip port for each line in the 8-port sipura and asterisk is authenticating properly now.
18:55.53Qwellrue_mohr: Why would you want to turn off the hwec?
18:55.58Qwelland yes, you can turn it off
18:56.46[TK]D-FenderQwell: Some people just never listen...
18:57.53rue_mohrQwell, there seems to be a horrid problem with the outgoing levels that is causing a problem with echo that not even the hwec is dealing with well
18:58.06rue_mohrto adjust the levels I need to turn the hwec off
18:58.17*** join/#asterisk voxter (n=voxter@76.77.95.2)
18:58.37Qwellwhy are you mucking with the gain?
18:59.07rue_mohrbecause the signal is going out so loud that I'm getting excessive far end reflection
18:59.17rue_mohrbottom line, to make the system work
19:00.17Qwellrue_mohr: and why is it loud?
19:00.47rue_mohrbecause the design of the card and the default settings are incorrect to match my pots provider properlyt
19:00.56rue_mohrhttp://www.pastebin.ca/1328240
19:01.09rue_mohrok, using that config file, the hwec is still on
19:01.16rue_mohrthat config file is http://www.pastebin.ca/1328240
19:02.00*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)
19:02.14rue_mohrwe have two options here, you can tell me the hwec shoudl be off, or you can say there is something I missed
19:02.41rue_mohrsorry if I'm being a little gruff, tk is driving me nuts
19:02.47Qwellrue_mohr: and what did support say when you called?
19:03.18rue_mohrI can do that if you dont want to answer...
19:04.06Qwellyou haven't shown anything that says ec is still enabled
19:04.25Qwellare you simply assuming that's the case?
19:04.46rue_mohrhow would I do that, you have to go on my word that I hear echo in the first dialed digit (quite clearly) then its gone, its done that since the echo can was installed
19:05.05rue_mohrwould you like a recording of the card in and another of the card out?
19:05.11Qwellso you've never actually checked that it's enabled?
19:05.30rue_mohrsee then there would be no point, cause I'm prolly gonna have to pull the card to disable it
19:05.51rue_mohrthe config says its disabled, is there a means to query it from the asterisk console?
19:05.56Qwellyes
19:06.14[TK]D-Fenderrue_mohr: Screw echotraining, you should ahve removed that and is why you get that moment of screwup
19:06.17rue_mohrah, how might that be done, there are no dahdi commands I'm aware of
19:06.30QwellThere are dahdi commands.
19:06.47rue_mohrah, :) well
19:06.58rue_mohrthere weren't when the drivers didn't work, sorry
19:07.03Qwellclearly
19:07.58rue_mohrEcho Cancellation: 0 taps unless TDM bridged, currently OFF
19:08.00rue_mohr!?!
19:08.25rue_mohrmakes a call on it
19:08.26Qwellso it's...disabled.
19:08.37QwellSO, like I asked earlier.  What makes you thiink it's the EC?
19:09.01*** join/#asterisk emiller (n=ed@static-98-117-7-38.hrbgpa.fios.verizon.net)
19:09.04rue_mohrits doing echo canceling
19:09.20Qwellclearly it isn't.
19:09.21rue_mohrI just cant prove it other than an audio recording
19:09.44Qwellyou can't prove it because it isn't on.
19:10.09Qwellso moving on.  it isn't the EC causing your problems.  What are the symptoms?
19:10.40rue_mohrok, let me play for a bit
19:12.09rue_mohrok sorry, at some point it did turn off, I'v dialed a dead air number and can hear the echo
19:13.49[TK]D-Fendershakes the loose bits out of rue_mohr's head
19:14.32*** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net)
19:14.40rue_mohrhttp://www.pastebin.ca/1328253
19:14.50rue_mohrnow I have a broken graph
19:15.26*** join/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info)
19:15.34rue_mohraccording to this, the signal is comming in at 200% :)
19:17.40*** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net)
19:20.03edibracI used to get bad HDLC frame log messages with my Digium TE121, but my Sangoma A101 works like a champ. The HDLC errors would then red alarm, drop all calls, and recover in a few seconds. Today I got a Red Alarm but w/o the HDLC -- how often should I see this? I've looked at: PCI timings, interrupts and BIOS settings.
19:21.33rue_mohrhttp://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html  the howtos I'm reading keep ending with my buddies name
19:23.56emillerhello all. I am having issues getting a few of my phones registered. could use a little direction. they were working at one point. they are able to receive calls, but not make them
19:25.00emillerhttp://pastebin.com/d495edfe0
19:25.56emillerhere is my sip show peer ... http://pastebin.com/d41b42194
19:27.37*** join/#asterisk jtodd (n=jtodd@96.8.38.118)
19:27.37*** mode/#asterisk [+o jtodd] by ChanServ
19:30.35*** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
19:31.29[TK]D-Fenderemiller: None of them are registered
19:32.35edibraci've read about people setting up a cron job to restart asterisk each night -- specifically what problems does this fix?
19:35.14pfneh, why does asterisk try to insert my voicemail into sql?  I don't have it configured for it in extconfig
19:35.23pfn[Feb  4 14:02:28] WARNING[27202]: res_odbc.c:149 ast_odbc_smart_execute: SQL Execute returned an error -1: HY000: Couldnt create (in-line) large object.;
19:35.27pfnUnrecognized key passed to SQLGetInfo30. (80)
19:36.27[TK]D-Fenderedibrac: Ancient idea when * was far more unstable
19:37.28florz[TK]D-Fender: 1.6 seriously is somewhere near anything one could call "stable"?
19:37.49davevg-btwtechpfn, did you enable ODBC voicemail in menuselect before compiling?
19:37.59edibrachmm ok another random question -- the Sangoma wiki talks about how to bind an IRQ to a certain CPU, what problem does that fix?
19:38.09[TK]D-Fenderflorz: It is far more stable for its age than 1.4 was.  Might be production-worthy... no real show-stoppers on previous functionailty.
19:38.23[TK]D-Fenderflorz: a few kinks, but thats par for the course
19:38.28rue_mohrk, well, see how -3db does for it, I can detect any difference for a standard call
19:40.01florz[TK]D-Fender: you mean, like, far more stable than 1.4.0.5 (or what ever the corresponding version war), or actually more stable than 1.4.23?
19:40.12florz*was
19:42.06adncis rtp also transfered with udp or is it tcp?
19:43.56carrarudp
19:44.58adncthanks
19:45.17*** join/#asterisk quaqo (n=quaqo@83-103-40-166.ip.fastwebnet.it)
19:46.36pfndavevg-btwtech, hmm, maybe
19:46.40pfndavevg-btwtech, I don't recall
19:46.46pfnI so rarely use voicemail
19:47.15[TK]D-Fenderflorz: At least a dovez versions ahead :)
19:47.44[TK]D-Fenderdozen*
19:48.14pfnhmm, the line numbers in ast_log don't correspond in app_voicemail
19:48.33Qwellrue_mohr: where are you getting 200% from?
19:49.34*** join/#asterisk Blackvel (n=blackvel@dslb-088-065-093-147.pools.arcor-ip.net)
19:49.35Blackvelhi all
19:51.13Blackvelwas there any way to whitelist a callerid (database put whitelist ...) while setting up day time dynamically?
19:52.18[TK]D-FenderBlackvel: PARDON?
19:52.36pfn[Feb  4 14:02:28] WARNING[27202]: app_voicemail.c:1392 store_file: SQL Execute error!
19:52.42pfnugh, why is the line number so horribly off
19:52.50Kobazhmmmmm
19:52.58*** join/#asterisk n3hxs (n=HAMming@71.39.159.200)
19:53.33Kobazman i just got tons of echo on a polycom phone to a sip trunk
19:53.51Kobazeverything i said came back to me 500ms later
19:53.58BlackvelI can manually whitelist a callerid (e.g when office is closed) but I want to whitelist and specify at what times (e.g 6-7 o' clock) the caller gets thru the system. whitelisting with AstDB does not care about the times.
19:54.40pfnrebuilds
19:54.41Blackvelbut i don't really want to program times in a fixed way. times per whitelisted callers would be a great thing.
19:54.58Blackveli am already using GotoIfTime / IfTime
19:55.06Blackvelbut how to do that dynamically?
19:57.23[TK]D-FenderBlackvel: Its your dialplan... use variables, do AstBD lookups, etc...
19:57.57[TK]D-FenderBlackvel: You are responsible for coding absolutely everything
19:58.27*** join/#asterisk Vooloo (n=sadasd@unaffiliated/vooloo)
19:58.57*** join/#asterisk cesar_CR (n=cesar@200.91.75.67)
19:59.23VoolooI need a call forward solution where I can track each calls duration. If it exceeds for example 2 minutes it will then log this so I can debit a client. Is this possible?
20:01.03carrarCDR logs all calls
20:01.17Blackvelhow would you combine astdb lookups with gotoiftime? mapping entered digits in astDB (1-20) to fromTime/toTime variables?
20:01.33codefreeze-lapcarrar:  you hope!
20:01.51[TK]D-FenderBlackvel: Thats one way.  Or do it all in an AGI with a more normal programming syntax and not use dialplan logic for that part
20:01.52Vooloocarrar: what is that?
20:01.58[TK]D-Fender~cdr
20:01.59jboti guess cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw
20:02.06[TK]D-FenderVooloo: Go read... THE BOOK
20:02.08[TK]D-Fender~book
20:02.09jboti guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook
20:02.09carrarVooloo> I need a call forward solution where I can track each calls duration. If it exceeds for example 2 minutes it
20:02.11carrar<PROTECTED>
20:02.13*** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis)
20:02.23carraryeah what TK said :)
20:02.28carrarREAD UP!
20:02.45carrarCDR = Call Detail Records
20:02.47Vooloo[TK]D-Fender: I just need to know one question. Do I need to read an entire book to know if I should dig deeper?
20:03.07Vooloook I see
20:03.07carrar1 download book
20:03.11carrar2 look at index
20:03.18carrar3 find call detail records secotion
20:03.21[TK]D-FenderVooloo: there is a CHAPTER in the book on this.
20:03.27[TK]D-FenderVooloo: And you can do what you want.
20:03.45harry_vbooks are always good
20:03.58Blackvel[TK]D-Fender] will have to look into that AGI thing then. was hoping I can abuse astdb commands for that ;)
20:04.04harry_vvooloo, you have he asterisk bookÉ
20:04.28Blackvelhow is asterisk business these days?
20:04.34Voolooharry_v: no I don't buy stuff until I will use it
20:04.35kaldemarBlackvel: you can use the function to get a value in the gotoif.
20:05.21harry_vsome how my ascii map is messed up on my question mark key.
20:05.32harry_vÉé
20:05.50Blackvelfriend with company was asking me about a small voip solution. probably he is looking for dlink horstbox professional solution or voip system with web gui. is freepbx/asterisknow stable enough to configure external sip peers with media gateways (e.g patton isdn bri gw)?
20:06.40*** join/#asterisk slima (i=slima@unaffiliated/slima)
20:06.56Blackveli was suggesting to go the direct asterisk way....not sure what his final requirements written down will be
20:07.02*** join/#asterisk seanmh (i=seanmh@216.31.101.68)
20:07.16*** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus)
20:08.48eppigyhello
20:08.48[TK]D-FenderYOU ARE DAVE
20:10.42*** join/#asterisk cryptnix (n=andrew@corp-nat.kaslnet.net)
20:10.42*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
20:10.52*** join/#asterisk workdraft (n=workdraf@124.107.139.132)
20:12.10workdrafti have now a running asteriskNOW box and would like to link it to other SIP providers. is it possible? like id like to have a capability to dial local numbers outside my country?
20:14.19sipyworkdraft, find a provider where you want, connect a trunk to em and dial away.
20:16.54sipyBlackvel, http://astbill.com/wiki/index.php/Asterisk_and_Patton_How_to
20:19.30*** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net)
20:20.03workdraftare they called this "sip trunk providers"?
20:20.48*** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca)
20:20.53kaldemarworkdraft: http://en.wikipedia.org/wiki/ITSP
20:21.06workdraftthnx
20:21.26kaldemarworkdraft: http://www.voip-info.org/wiki/view/VOIP+Service+Providers
20:21.42Nate187is there some trick to get "ringinuse=no" to work in queues.conf?
20:21.50*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
20:22.20grandpapadotHi all, what's the best way to check and see if a peer exists and if it's available? I've tried ChanIsAvail but it seems to always return 0 no matter what.
20:24.35Blackvelwipy: ahh...great link!!! I have btw a running asterisk patton configuration for myself. but if that would work with asterisk + gui (asterisknow, freepbx, etc.) or horstbox pro, I wonder if sip peers can be manually configured in a gui
20:26.48kaldemargrandpapadot: core show function DEVICE_STATE
20:27.05*** join/#asterisk bbhenry (n=hhuang@wsip-68-105-250-226.sd.sd.cox.net)
20:27.19grandpapadotIn 1.4, no function found.  Is that a 1.6 function?
20:27.22grandpapadot(and thanks)
20:27.27ruben23hi....does rtp uses only udp or it also use tcp protocol.../
20:27.57Nate187rtp only used udp
20:29.16sipyits DEVSTATE
20:30.01grandpapadotThanks, must be in 1.6, it's not in 1.4.x
20:30.29bbhenrynot many people chatting here
20:30.35sipyI have it in 1.4
20:30.55grandpapadotReally? It shows up with a "core show function DEVSTATE"?
20:31.37sipyyes
20:31.40ruben23Nate187:thanks....
20:31.44kaldemargrandpapadot: yes, DEVICE_STATE is a 1.6 function
20:31.58cesau2http://www.elastix.org/index.php?option=com_fireboard&Itemid=55&id=13448&catid=26&func=fb_pdf
20:32.23sipyhowever DEVSTATE is a 1.4 function
20:32.27grandpapadotAhh, it's backported.
20:32.37[TK]D-FenderBIG PRINT
20:33.16*** join/#asterisk docelmo (n=vircuser@pool-141-152-198-116.lyn.east.verizon.net)
20:33.52cesau2(caps angers D, use sparingly!)
20:36.15[TK]D-Fendercesau2: Not at all...
20:36.44emillerwould i run into any known problems if i renamed my asterisk production server?
20:37.15Qwellemiller: renamed?
20:37.43*** join/#asterisk bmoraca (n=bmoraca@209.60.253.58)
20:38.04scooby2any reason the cdr would show the queue number instead of the number called for dst?
20:38.20emillerQwell: the host name
20:38.29Qwellemiller: Are you using the hostname in any configs?
20:38.44emillernot that i know of.
20:38.48emillerbut you know how that goes.
20:39.03Qwellthen we can't possibly know the answer to your question
20:39.05cvnethi
20:39.07[TK]D-Fenderemiller: Not applicable
20:39.14cvnethow can i just show ringing?
20:39.19emillerThanks Qwell and D-Fender
20:39.54[TK]D-Fendercvnet: To who?  How long?
20:40.23cvnetI mean exten => _X.,1,Ring(1 Sec) <-- can i do something like that?
20:40.49cvnetfake ringing i mean
20:41.01[TK]D-Fendercvnet: Ringing()
20:41.15cvnetbut can u put how many sec as a paramter?
20:41.28[TK]D-Fendercvnet: Go read the instructions.
20:45.25cvnetexten => s,2,wait(2) <-- the 2 is is sec, or m-Sec ?
20:46.13*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
20:46.23*** part/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
20:47.06[TK]D-Fendercvnet: go try and SEE
20:47.31rue_mohrhmm "In testing various analog pstn I/O cards, I've found the sangoma A200D
20:47.31rue_mohrcard (with hardware echo canceler) to be the best pstn analog interface " <- you know, it occurs to me that the channelbank I have at home has a 'loop length' paramiter for each channel
20:48.39rue_mohrI have a telco lineman school book, If I can get some numbers I may try an impedence matcher on the lines between the pots and the dahdi card
20:48.43deadpigeonive had no problems with te110p
20:48.49rue_mohrif needed
20:48.53rue_mohrhmm
20:49.45*** join/#asterisk jeffgus (n=jeffgus@green.zimage.com)
20:51.11*** join/#asterisk jtrimmer (n=jtrimmer@75.151.66.133)
20:56.22*** join/#asterisk jsolis (n=jimmy@190.41.153.85)
21:00.45*** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com)
21:01.51*** join/#asterisk LND (n=LND@89.193.253.240)
21:01.53Qwellrue_mohr: what card is this?
21:04.47*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)
21:05.03marv[work]does asterisk have api docs up at some url?
21:05.33Qwellhttp://www.asterisk.org/doxygen/
21:06.53marv[work]thanks. i was going to complain that google didn't find that, but after repeating my search I see that I actually missed it. oops.
21:09.14jsolisto 1.6 exist any doxygen?
21:10.08*** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
21:10.08*** mode/#asterisk [+o russellb] by ChanServ
21:15.47pfnhow do I make LOG_DEBUG messages show up?
21:15.52pfncore set debug 2 doesn't seem to show it
21:17.19marv[work]/etc/asterisk/logging.conf
21:17.35*** part/#asterisk marv[work] (n=timr@router.asteriasgi.com)
21:17.55*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
21:18.40pfnwhat's core set debug for then?
21:18.41*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
21:24.44*** part/#asterisk Vooloo (n=sadasd@unaffiliated/vooloo)
21:24.44pfnwhere's the ast_log macro defined?
21:28.16rue_mohrQwell, I have a tdm800P with 4 fxo and 2 fxs channels
21:28.25rue_mohrhave to go install tv's back later
21:29.42cvnetI have a client who uses voipswtich, and when i just fake rings for 30 sec, my system doesnt charge him for calls but his system charges him for call, he is telling me that i should turn off "connected clients immidiately"
21:30.36Corydon76-digpfn: logger.c
21:30.50Corydon76-digpfn: or the macro is in logger.h
21:32.19pfnoh, the LOG_ are the macros
21:32.21pfnnot ast_log
21:32.52pfnno wonder I was confuzzled
21:33.00pfnnow to figure out how I make LOG_DEBUG print crap out to the console
21:33.18Corydon76-digpfn: enable it in logger.conf
21:33.29pfncore set debug 1 app_voicemail.c
21:33.36pfncore set verbose 0
21:34.27pfner, don't need the latter
21:35.48RModhey guys
21:35.53pfnhmm, I guess that didn't work either
21:36.24RModi have an agi script that dials a number, everything works exept when the number it is calling is busy, it just hangs up instead of hearing busy tone
21:36.29Corydon76-digpfn: did you enable it in logger.conf?
21:36.32pfnno
21:37.41*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
21:37.43pfnit has to be explicitly enabled in logger.conf first?  hrm
21:38.47Corydon76-digpfn: the console line has to specify debug output
21:41.27pfn[Feb  5 13:41:22] DEBUG[30299]: app_voicemail.c:2146 generic_prepare: SQL statement: SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/100/INBOX'
21:41.32pfnwhy is this happening 2x a second
21:49.29*** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
21:49.29*** mode/#asterisk [+o russellb] by ChanServ
21:50.08*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com)
21:50.12Kattyweeeeeeeeeeeeee!!
21:50.53*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
21:51.39jayteewheeeeeeeeeeeee!!!!
21:52.03Kattywiiiiiiiiiiiiiiii!
21:52.10jayteeeven more fun!
21:52.15Kattyso you claim.
21:52.16jayteeI love Wii bowling
21:52.17KattyBUT IS IT REALLY
21:52.46jayteeso what were you weeeeee'ing about?
21:53.05timeshellWhy is it that "stop now" doesn't work in 1.6.1?
21:53.09sipyher knee
21:55.00russellbtimeshell: core stop now
21:55.39timeshellUm.
21:55.43timeshellThat doesn't work either
21:55.49Kattyi really wasn't weeeing about anything imparticular.
21:55.58Kattyhowever.
21:56.01Kattyi am highly caffinated.
21:56.04Kattyperhaps that has something to do with it
21:56.06jayteeyay!
21:56.23jayteeI heartily approve!
21:56.23*** join/#asterisk frogonwheels (n=michaelg@203.59.141.93)
21:56.30jayteeI roast my own coffee
21:56.49eppigyhello
21:56.50timeshellrussellb , core stop now doesn't do anything in 1.6.1-rc1
21:56.53toadonsurfboardyou are dave
21:57.03eppigyyes
21:57.13toadonsurfboardtimeshell, type help core or help stop
21:57.20russellbtimeshell: Hm, then I don't remember ...
21:57.58timeshelltoadonsurfboard neither helps
21:58.07toadonsurfboardneither shows anything?
21:58.17timeshellhelp stop shows nothing
21:58.21eppigySTOP NOW
21:58.26timeshellhelp core doesn't seem to show me anything useful
21:58.32toadonsurfboardTRABAJO
21:58.51eppigyNO NECISITO
21:58.52toadonsurfboardI don't have 1.6 running here so I can't check it.
21:58.59eppigyDONDE ESTA
21:59.00sipyamportal stop?
21:59.00timeshellIt works in 1.6.0
21:59.04timeshellIt doesn't in 1.6.1
21:59.21timeshellEu sou aqui agora.
21:59.22toadonsurfboardtimeshell, are  you running asterisk as a service?
21:59.32timeshellOui oui oui
21:59.33toadonsurfboardor just from the command line?
21:59.36timeshellservice
21:59.39toadonsurfboardtype exit
21:59.46Kattytoadonsurfboard: roaster.
21:59.54eppigyservice asterisk stop
22:00.01toadonsurfboardand on the command line type /etc/init.d/asterisk stop
22:00.04eppigyps -A | grep asterisk
22:00.07eppigykill -9
22:00.08toadonsurfboardor service asterisk stop
22:00.09timeshelleppigy.  Tried that.
22:00.12timeshelldoens't kill it
22:00.17Kattyreally
22:00.19eppigywhich one
22:00.20timeshell1.6.1 seems a little buggy still
22:00.23Kattyi find it hard to believe that kill doesn't kill.
22:00.28toadonsurfboardKatty, I have an iRoast2
22:00.34eppigy4rl
22:00.36sipystop gracefully
22:00.37Kattytoadonsurfboard: schnazzy.
22:00.50Kattytimeshell: hold the power button in for 5 seconds.
22:00.57eppigyKILL YOUR FAMILY
22:00.59toadonsurfboardhttp://www.sweetmarias.com/prod.hearthwareiRoast2.shtml
22:01.00timeshellyah, thanks katty
22:01.08Kattyglad to be of service.
22:01.23timeshellSo, it do /etc/init.d/asterisk stop
22:01.36Kattyjaytee: oh
22:01.46Kattyjaytee: that's lovely.
22:01.46timeshellbut ps -ef | grep asterisk still shows safe_asterisk running and asterisk -f -vvvg -c
22:01.55timeshellEven though it says the service stopped ok
22:01.58sipylovely?
22:02.05Kattyyes. lovely.
22:02.06timeshellI haven't been able to kill with the kill command either.
22:02.08timeshell:(
22:02.19jayteeso basically a hung process you can't kill with kill -9? that's odd
22:02.20outtolunckillall safe_asterisk
22:02.29jayteeyeah, I'd try that
22:02.44Kattysipy: appealing to the emotions as well as the eye.
22:02.45eppigyKILL EVERYTHYING
22:02.56Kattyexample: jaytee
22:03.00jayteeor do a kill -9 on the safe_asterisk script specifically by PID
22:03.15timeshellI can kill safe_asterisk butI can't kill asterisk -f -vvvg -c
22:03.30eppigyis your HD full?
22:03.35Kattyeppigy: that will happen soon enough.
22:03.35jayteeblushes bright red and hugs Katty
22:03.43eppigyKatty: nice
22:03.43timeshellkill -9 doesn't kill it
22:03.49timeshellNo, hdd not full
22:03.53eppigydoes it have a new pid each time?
22:04.02Kattyneeds to catch up on reddit today.
22:04.13timeshelloh, wait.  finally it dies
22:04.17eppigyLIES
22:04.17jayteetimeshell, what distro are  you running?
22:04.22KattyHORAY!
22:04.22timeshellNo, was the same pid
22:04.24KattyYOU MADE IT!
22:04.31eppigyRAGE FILLING EVERY CAPILLARY
22:04.39timeshellSo, how to you really stop 1.6.1 from CLI?
22:04.58jayteestop gracefully? stop when convenient
22:05.00Kattystop now?
22:05.16Qwelltimeshell: Did you ever stop Asterisk in 1.4?
22:05.18Kattyctrl c
22:05.25jayteehe didn't get that option or they changed it so that calls in progress wouldn't get terminated maybe?
22:05.36timeshellqwell:  Um.  Yah
22:05.38*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
22:05.47Qwelltimeshell: Then you ignored the deprecation notices, and it's your fault. ;)
22:05.53jayteeI'll have to look at 1.6.0.1 at home to see if it's changed
22:05.59Qwellcore stop now
22:06.02jayteebe back later, quittin time
22:06.06Kattybuhbye
22:06.09outtolunc+pretty please
22:06.12jayteebye Katty
22:06.12timeshellqwell:  I mentioned earlier stop now works in 1.6.0.5
22:06.23Qwelland you ignored those deprecation notices too :)
22:06.31timeshellqwell:  I was just troubleshooting something else with awk_r and was using it quite a bit.
22:06.35Kattythat's what they're there for.
22:06.37Kattyto ignore.
22:06.44Kattyright? ;)
22:06.45sipymajor major changes in 1.6?
22:06.45timeshellqwell:  And incidentally, core stop now I already tried too and it didn't work either.
22:06.51Kattysipy: yes.
22:06.53Qwell"didn't work"?
22:06.58Kattysipy: agi stuff.
22:07.02timeshell"DIDN'T WORK"
22:07.03Kattysipy: amongst other things.
22:07.15timeshellUnknown command
22:07.34timeshellI'd say that's probably a bug for a RC?
22:07.50timeshellDoesn't show as an available core command AT ALL
22:09.34eppigyi am swamped
22:10.30Kattyoh hey
22:10.33Kattythey discovered another mammoth
22:11.52Carlos_PHXholds back witty penis joke.
22:11.59theharhaha
22:12.34Katty:<
22:14.02*** join/#asterisk nix8n82 (n=nate@63.162.27.243)
22:14.25Carlos_PHXApparently most of West Virginia is without VoIP.  Anyone know a carrier with strong rural coverage?
22:14.41*** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it)
22:15.06Kattyrats.
22:15.33Carlos_PHXRats have VoIP?
22:15.43Kattythey're a carrier.
22:15.46Kattywith strong rural coverage.
22:15.50Carlos_PHXROFL
22:15.56*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
22:16.00Kattyhi fender.
22:16.01Carlos_PHXDamn, I'm slow today.
22:16.11[TK]D-FenderKatty: Mew.
22:16.20*** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it)
22:16.28Carlos_PHXCame in just in time to miss your awesome wit
22:19.39*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
22:21.22sipyHave you successfully completed your 8 minutes of sleep?
22:23.22[TK]D-Fendersipy: That was LAST night!
22:23.37[TK]D-Fendersipy: I'm working towards tonights 8 minutes of sleep
22:23.39*** join/#asterisk asheron (n=stephan@95-36-36-255.dsl.alice.nl)
22:26.15*** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-bcb5e11d82e3b523)
22:28.20*** join/#asterisk bgmarete (n=marebri_@196.201.217.35)
22:28.26talirk81Is anyone around familar with  AGI ... I am planning to do 'EXEC DIAL "SIP/1XXXXXXX,NumberofRings,gmL[Duration]"' my question is if im using  fgets(STDIN) to get the results will that also pull in  ${DIALSTATUS} so i can tell if the call connected
22:29.20talirk81I ask because a Dial can be a success even if the call didnt connect right? aka congestion or no answer in the alloted Number of Ring Seconds
22:30.37[TK]D-Fendertalirk81: If you're in AGI you need to use the "GET VARIABLE" AGI command
22:31.00timeshellIs provisioning automatically enabled for polycom phones or does something need to be done to enable provisioning?
22:31.33[TK]D-Fendertimeshell: OOB they look for DHCP opt 66 for the FTP server using the cred the admin guide told you
22:33.09pfnjust switches back to filesystem based voicemail
22:35.34*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
22:35.43*** join/#asterisk abatista (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
22:39.10*** join/#asterisk speedwagon (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net)
22:39.21timeshellTKD, I kind of meant on the asterisk server side.
22:40.20timeshellDoes asterisk have built in ftp service or is it necessary to configure a separate service
22:41.09eppigyTRIPPIN
22:41.42pfnbuilt in ftp service?  what for...
22:41.46pfnrun tftp separately
22:41.50pfnit's called tftp for a reaso
22:41.51pfnn
22:41.52[TK]D-Fendertimeshell: No.  Asterisk is not an FTP server, a mail server, a home theater system, a lunch-box cleaner, OR a vacuum cleaner
22:42.14timeshellTKD:  But with asterisk-gui it IS an http server.
22:42.21eppigyi use mine to get out those dried in stains
22:42.21timeshellTKD: Hence the question
22:42.22putnopvut[TK]D-Fender: actually, Asterisk is a lunch-box cleaner
22:42.31putnopvut(okay, i lied)
22:42.36timeshelllol
22:42.37timeshellnm
22:43.08[TK]D-Fendertimeshell: 1 tiny piece of crap HTTP daemon.  Big deal.. that jsut because we assume that 90% of users are also too dumb to configure Apache for themselves.  And we're RIGHT
22:43.35timeshelllol
22:44.00[TK]D-Fendertimeshell: Configure XYZ phone for you magically?  No, thats YOUR job
22:44.09timeshellNonetheless, it's there and I want to know if any other service for provisioning exists before I spend my time configuring one.
22:44.34[TK]D-Fendertimeshell: Sure, Trixbox has their craptastic "endpoint manager"
22:44.35timeshellI'm not an asterisk expert.  That's why I'm HERE asking questions.  :D
22:45.22[TK]D-Fendertimeshell: Go steal^H^H^H^H^Hborrow it from their setup
22:46.21pfnwriting a tftp server for asterisk would be pretty handy, though...
22:46.35Qwellpfn: why?
22:46.52pfnQwell, one less external service to run, why else?
22:47.12timeshellQwell:  Well asterisk IS a PBX.  You WANT it to provision your phones don't you?
22:47.13QwellSo why not ask for exchange backend support?
22:47.19eppigytrue
22:47.32eppigycan I do calendar sharing
22:47.34pfnQwell, because tftp and asterisk are related, exchange and asterisk are not
22:47.35eppigywith asterisk?
22:47.39Qwellpfn: says you.
22:47.43[TK]D-Fenderpfn: And while you're at it, just think how cool it'd be if you included a kernel with it!
22:47.48eppigyi have so many emails abotu asterisk
22:47.50[TK]D-Fenderpfn: And an ftp server
22:47.50eppigyits crazy
22:47.57timeshellpfn:  Exchange and asterisk could be VERY closely related.
22:48.00pfntftp is pretty essential to asterisk phone provisioning
22:48.06[TK]D-Fenderpfn: And maybe a web server & something like MySQL!
22:48.09thedonvaughnasterisk can also use a RDBMS db.  should it include one too? :)
22:48.14eppigypfn: yeah among a million other things
22:48.23pfnsqlite is embeddable in any case  :p
22:48.27[TK]D-Fenderpfn: All at once... heck, why not even distribute it in .iso format!
22:48.28Corydon76-digpfn: not all phones need tftp
22:48.41pfn[TK]D-Fender, that's trivial and already done  :p
22:48.43Corydon76-digpfn: http is supported more and more
22:48.44eppigyi provision with http
22:48.48[TK]D-Fenderpfn: But don't worry, your wheel will be ROUNDER!
22:49.03pfnCorydon76-dig, maybe that's true now... I've only been using old tftp phones
22:49.12[TK]D-Fenderpfn: Which?
22:49.46[TK]D-FenderI provision everything I can via FTP instead of TFTP.
22:50.05drmessanoAsterisk comes with an httpd
22:50.09drmessanoProblem solved
22:50.10Kattyhmm.
22:50.13[TK]D-Fenderdrmessano: See above...
22:50.17*** join/#asterisk aae (n=andrew@user-0c8hlc7.cable.mindspring.com)
22:50.57Kattytime to go home
22:50.58Kattybuhbye
22:51.13[TK]D-FenderKatty: BAI BAI
22:51.30eppigyi wish i could go home
22:51.35aaei'm new to asterisk, i'd like to play around a bit with it, but i dont' want to invest too much money into it, anybody have any suggestions on some very reasonably priced interface cards?
22:51.42eppigyBROKEN HOME
22:52.19eppigyaae: you dont need interface cards to play around with it
22:52.57Qwellaae: What do you want to do?
22:53.06aaei would like to set it up with some "time warner digital phone" lines though....
22:53.21Qwellaae: so, just one fxo port?
22:53.28drmessanoeppigy: ROOF CAVE IN?
22:53.33aaethere are two, but just one would do for playing around
22:53.48Qwellaae: well, the digium tdm410 is modular.  you can get the base card + 1 fxo module
22:54.19aaeis that different from the AEX410?
22:54.29eppigydrmessano: YES
22:54.30Qwellaex is pci-e
22:54.37aaeok
22:54.42aaei see the link, lemme check it out
22:55.12aaeok, so digium doesn't sell the base card?
22:55.28Corydon76-digNot by itself, no
22:55.35Qwellit isn't exactly useful by itself :)
22:55.52aae:D neither is a CPU, but plenty of people sell those :P
22:56.02Qwellyou typically buy it in a bundle
22:56.05aaeok
22:56.14aaethe cheapest bundle i see here is the TDM411E then?
22:56.16[TK]D-Fenderaae: Here : www.telephonydepot.com <- some of the best pricing around for * related gear
22:56.19Corydon76-digTechnically, Digium does, but only to distributors and resellers
22:56.38Corydon76-digEnd users only get packages in useful configurations
22:56.49aaethanks
22:57.18aaeso, would you guys suggest i start with loading it up myself in linux, or using asterisk now!
22:57.19QwellCorydon76-dig: btw, see #-dev :D
22:57.19aae?
22:57.26infinity1whats wrong with this ?
22:57.27infinity1Set(CALLERID(number)=1${CALLERID(num));
22:57.28Corydon76-digHeh
22:57.46Corydon76-diginfinity1: for one, you're missing a closing brace
22:57.48aaei've loaded asterisk now! on VM and played around with softphones, but i could never get the VOIP to work...
22:57.54infinity1oh shit
22:58.25infinity1Corydon76-dig: thanks :)
22:59.26pfn[TK]D-Fender, cisco 7960
23:01.21aaealso, is there a VOIP provider out there with the same or better uptime than companies like vonage?
23:01.34aaeand good service
23:02.27*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
23:02.55talirk81i like aretta
23:03.05talirk81ive had nothing but good service with them
23:03.37[TK]D-Fenderpfn: ah.....
23:03.56[TK]D-Fender~itsplist-us
23:03.56jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
23:04.02[TK]D-Fenderaae: ^^^
23:04.24*** join/#asterisk keith4 (n=keith@lust.cc.lehigh.edu)
23:04.31aaethanks
23:04.45eppigystarving
23:05.00pfnvoicepulse... are they still horribly expensive?
23:05.11pfnhmm, need to check those others on the list
23:05.19pfnbeen using voipjet mostly for my intl termination
23:05.54[TK]D-Fenderoff to martial arts, BBL
23:06.03rob0hi YA!
23:06.23[TK]D-Fender~tsksr
23:06.24rob0tried to sound like Miss Piggy doing a karate chop
23:06.46*** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net)
23:06.49rob0Like Yoda instead, I sounded.
23:07.06[TK]D-Fenderrob0: http://en.wikipedia.org/wiki/Tenshin_Sh%C5%8Dden_Katori_Shint%C5%8D-ry%C5%AB
23:07.13[TK]D-Fenderrob0: My art.  Read up
23:07.23*** join/#asterisk docelm0 (n=vircuser@pool-151-199-177-35.lyn.east.verizon.net)
23:08.24pfnjnctn.com kinda sucks, can't find any info without signing up?
23:10.08*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
23:10.33pfnhmm, I guess voipjet still has the lowest rates for where I want to call
23:10.45SkramXive used iax.cc
23:10.55SkramXwhich is now vitelity or something
23:10.59SkramXidk how they've held up though
23:11.06harry_vshy of voipjet going out of business :)
23:11.21pfnshrugs
23:11.35pfnI just need outbound termination, if they go out of business, I'll shift my traffic elsewhere
23:11.50cesau2i fail at teh telephony =(
23:11.54aaeif one of these places goes out of business, is it that hard to get your number switched?
23:12.02*** join/#asterisk bgmarete (n=marebri_@196.201.217.53)
23:12.33harry_vpfn, thats because people dont give a crap about loyalty and that is why at times thay do go out of business.
23:12.53pfnshrugs
23:13.05pfnbeen using voipjet for near 4 years now
23:13.06pfnI think
23:13.35harry_vpfn whats your call volume with voipjet and does call quality suffer at times?
23:13.47pfncall quality does suffer, and not a whole lot of volume
23:13.51pfnjust a few thousand minutes a month
23:14.51aaei'm here looking at teliax, but i just want a SIP trunk, i don't want to have to pay for mailboxes and stuff
23:15.02aaejust want like.... 5 channels or so, one phone number...
23:15.17aaeand a reasonable incoming/outgoing
23:16.14Carlos_PHXMy company provides SIP trunks, 1.9 cpm, minimum $50/mo though.
23:16.35aaewhat's 1.9 cpm?
23:16.37Carlos_PHXWe're not going out of business, but then, we all claim that.
23:16.42Carlos_PHXcents per minute
23:16.45rob0~siptrunk
23:16.46jbotNo such thing, my friend.. Like too much salty plum soda.
23:16.49aaein or out?
23:16.58Carlos_PHXBoth
23:17.17Carlos_PHXrob0:  That's nice.
23:17.19aaequick question on that
23:17.31aaeif i make two calls at 30 seconds each, is that 2 minutes or 1 minute?
23:17.38Carlos_PHX1
23:17.41aaeoooo
23:17.42Carlos_PHX6-second billing
23:17.46aaergr
23:17.57Carlos_PHXNo minimum per call, so we do take call centers.
23:18.02*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
23:18.08talirk81aretta is 1.39 cpm  for me
23:18.14talirk81unless im using a toll free number
23:18.19Carlos_PHXI recommend using them then.
23:18.59talirk81toll free is a bit more
23:19.13talirk81plus they also have a  flat rate plan  for personal/dev usage :)
23:19.25aaeis there an application for iphone that can either 1, spoof your phone number, or 2, allow you to dial through PBX?
23:19.33aaedunno if 1 is possible
23:19.34Carlos_PHXYes
23:20.07aaeok, yes on #2?
23:20.12Carlos_PHXOn both.
23:20.18Carlos_PHXIt's in the App store.
23:20.26aaek, let me look
23:20.31Carlos_PHXI don't recall the name.
23:20.48Carlos_PHXBasically it's just dialing a VoIP service probably using Asterisk and changing CID.  No magic.
23:21.13aaeright, thought so... so they'll want you to pay usage...
23:21.21Carlos_PHXYes, something like 10 cpm
23:21.23aae#2 on the other hand... seems a lot more plausible....
23:21.33aaefor a lot less money
23:21.36aaesomething seemless
23:21.39Carlos_PHXThat product is called SIAX
23:21.43aaeseamless*
23:21.47Carlos_PHXThere's another one I tried and sucked.
23:21.53Carlos_PHXEither way, VoIP on a PDA is crap.
23:22.02Carlos_PHXVery limited use.
23:22.03QwellWhy not dial into your own PBX and dial out from there?
23:22.03aaei dont' want to access via data
23:22.17aaeqwell, exactly, but i'd like to use the phonebook on my phone
23:22.30Qwellyou still can...sort of
23:22.38Qwellunless the iPhone phonebook is garbage
23:22.43aaeprobably is
23:22.53QwellJust prefix all the entries with your PBX # and a hard pause
23:23.00Carlos_PHXQwell: Under your standards, it is garbage.
23:23.03aaeright, thats what i thought you were gonna say
23:23.11*** join/#asterisk SparFux (n=raoul@e182020105.adsl.alicedsl.de)
23:23.16aaedunno why there wouldn't be an APP that just does that for you
23:23.17Qwellmy $0 t-mobile phone could do that
23:23.26Carlos_PHXYeah, there is, but tied to THEIR service.
23:23.32Carlos_PHXIf someone wanted to write an app....
23:24.01Carlos_PHXVerizon sucks donkey balls.  47 minutes on hold so far.
23:24.03aaeif i knew anything about that, i would... just flip a switch in the APP to turn it ON, and every call is prefixed by your PBX# and a hard pause or code whatever, you need...
23:24.15Carlos_PHXYup, most phones used to do that.
23:24.22*** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec)
23:24.23Carlos_PHXWould be a simple feature.
23:24.31Carlos_PHXIt was called "calling card prefix" or similar.
23:24.38aaehmmm
23:25.02SparFuxPhone ATAs are really a great idea :-)
23:25.02Carlos_PHXMy freakin' Nokia from 1998 does it.
23:25.10*** join/#asterisk eric2 (n=nobody@69.60.247.142)
23:25.44aaei could just turn the "show my caller ID" to off on my iphone
23:26.02Carlos_PHXThen nobody would answer.  I wouldn't, and most people I know wouldn't.
23:26.23Carlos_PHXI face the same annoyance, I want people to call my VoIP number and not the cell directly.
23:27.20carraralways speed dial to your * box and then dialout from there
23:27.34aaebut i want to dial using the phonebook
23:27.48carrarcreate speed dials on the *
23:28.06aaei could....
23:28.13aaethis would be easier, and it's simple...
23:28.25aaei'd rather go through my phonebook and edit them all in on the PC
23:28.28aaethen sync
23:28.28carrarget a cell phone that has a SIP client in it
23:28.31*** part/#asterisk SparFux (n=raoul@e182020105.adsl.alicedsl.de)
23:28.40aaeSIP clients are easy to come by for iphone i think....
23:28.45aaejust don't have a good data plan
23:28.50aaei'm running a hacked phone on tmobile
23:29.00aaeunplan... so unlimited minutes...
23:29.01pfnhuh, iphone doesn't have a calling card feature?
23:29.02Carlos_PHXI have unlimited.  I have used SIP clients on many PDAs.  THey all suck.
23:29.13Carlos_PHXpfn: No
23:29.14aaepfn, you'd think it would... i'm looking now...
23:29.40carrarJust buyout AT&T
23:29.49Carlos_PHXLooks like there's an add-on for that.
23:30.10pfnindeed, sounds like it needs 3rd party integration, suck
23:30.26aaethere is an app called granddialer...
23:30.41Carlos_PHXAlso found an app called... ta-da
23:30.45Carlos_PHXCalling Card
23:30.46aaegoes over the data to your grand central account, and has it call your cell phone, and then connect you to the number you dialed
23:31.02aaei just did a search of the app store for those exact two words, lol
23:31.16Carlos_PHXhttp://homepage.mac.com/aamann/CallingCard.html
23:32.01Carlos_PHXSo I wonder if you can have GC send any CID, or just the one they assign to you?
23:32.32aaethe one they assign, which isn't bad at all, i mean they gave me that number free...
23:32.38aaei almost feel like just using it.....
23:32.45aaeis there a way to interface GC to PBX?
23:32.49Carlos_PHXThe app is $1, I think I can swing that.
23:32.55harry_vcarlos, looking for a pstn phone with some amazing range?
23:33.03Carlos_PHXHuh?
23:33.15aaeCarlos_PHX, there is also an app called Card Caller that might also work for free
23:33.18Carlos_PHXPSTN phone....range...as long as the cable?
23:33.19*** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk)
23:33.33Carlos_PHX$1 ... free hmm
23:33.45Carlos_PHXOh, there's that fax guy, watch it now.
23:34.14harry_vthere is a phone that can be used up to 1/4 mile or further or 12 floors of concrete. Very expensive phone though. It is a wireless to analog phone like a typical home phone but on staraoids
23:34.34aaenot just sterioids, STARoids
23:34.36harry_vsteriods
23:34.37Carlos_PHXNothing that I need.
23:34.53Carlos_PHXSteroids even.
23:35.02harry_vyea
23:35.04harry_vokay
23:35.30harry_vthis phone is good for farms or construction sites.
23:35.31Carlos_PHXThere are a bunch of DECT phones with huge range.  But all way expensive.
23:35.40Carlos_PHXI haven't found a use for them yet, as cool as they'd be to play with.
23:35.41harry_vmyfunky keyboard
23:36.02harry_vIsee
23:36.28*** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net)
23:36.50carrarThere are a lot of illegal (In the USA) phones
23:37.01carrarthat use the HAM Radio spectrum
23:37.04*** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis)
23:41.46Carlos_PHXSo I've been on hold with Verizon for an hour and six minutes, and my sales rep calls me to see if there's anything else he can sell me....
23:43.32Carlos_PHXaae: You going to try any of those iPhone apps?
23:43.33aaecould you illegally set up your own cell tower?
23:43.42Carlos_PHX"Could" you?  Sure.
23:43.42aaeeventually, i dont have a PBX
23:44.02Carlos_PHXSorry to hear that.  Everyone should have at least one.
23:44.06*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
23:44.11aaethat's why i'm here
23:44.30Carlos_PHXI have 14, but I don't share well.
23:44.31carrarWhy am I here
23:44.39Carlos_PHXWhy are any of us here?
23:44.43Carlos_PHXAre any of us really here?
23:44.54rob0Well, before you were born, your mommy and your daddy ...
23:44.59carrarVirtuaally!
23:48.20*** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com)
23:54.10*** join/#asterisk SiXcAps (n=nukli@pool-173-69-180-239.bltmmd.fios.verizon.net)
23:56.50*** join/#asterisk killown (n=Yamato@unaffiliated/killown)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.